| Entries |
| Document | Title | Date |
|---|
| 20080201152 | Apparatus for Encoding and Decoding Audio Signal and Method Thereof - A method and/or apparatus for encoding and/or decoding an audio signal is disclosed, in which a downmix gain is applied to a downmix signal in an encoding apparatus which, in turn, transmits, to a decoding apparatus, a bitstream containing information as to the applied downmix gain. The decoding apparatus recovers the downmix signal, using the downmix gain information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the encoding apparatus can apply an arbitrary downmix gain (ADG) to the downmix signal, and can transmit a bitstream containing information as to the applied ADG to the decoding apparatus. The decoding apparatus recovers the downmix signal, using the ADG information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the method and/or apparatus can also vary the energy level of a specific channel, and can recover the varied energy level. | 08-21-2008 |
| 20080201153 | Generation of Multi-Channel Audio Signals - A decoder ( | 08-21-2008 |
| 20080208600 | Apparatus for Encoding and Decoding Audio Signal and Method Thereof - A method and/or apparatus for encoding and/or decoding an audio signal is disclosed, in which a downmix gain is applied to a downmix signal in an encoding apparatus which, in turn, transmits, to a decoding apparatus, a bit stream containing information as to the applied downmix gain. The decoding apparatus recovers the downmix signal, using the downmix gain information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the encoding apparatus can apply an arbitrary downmix gain (ADG) to the downmix signal, and can transmit a bit stream containing information as to the applied ADG to the decoding apparatus. The decoding apparatus recovers the downmix signal, using the ADG information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the method and/or apparatus can also vary the energy level of a specific channel, and can recover the varied energy level. | 08-28-2008 |
| 20080208601 | UNIVERSAL CONTAINER FOR AUDIO DATA - Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent. | 08-28-2008 |
| 20080215340 | Compressing Method for Digital Audio Files - A compressing method for digital audio files mainly utilizes a harmonic structure quad tree (HSQT) to re-arrange the frequency coefficient in each frame, and applies concurrent encoding in hierarchical trees (CEIHT) algorithm to increase and simplify the processing speed; the coefficient of the CEIHT is symbolized according to an arithmetic coding; the record of the probability of the symbol is used to determine the number of bits to be stored; the probability is in inverse order of the number of bits requiring storage, and thus increasing the occurrence probability of the symbol may greatly reduce the number of bits to be stored. As a result, the overall compressing method is done in simplified processing procedures and outputting an audio compressed file with a high compression ratio. | 09-04-2008 |
| 20080215341 | SIGNAL COMPRESSING APPARATUS - An input signal is quantized into a quantization-resultant signal. The quantization-resultant signal is compressed into a compression-resultant signal. The compression-resultant signal is formatted into a formatting-resultant signal corresponding to a predetermined format for a digital recording disc. The formatting-resultant signal includes segments corresponding to user data areas prescribed in the predetermined format. The compression-resultant signal is placed in the segments of the formatting-resultant signal. The formatting-resultant signal is encoded into an encoding-resultant signal of a CD format. The encoding-resultant signal is recorded on a recording medium. | 09-04-2008 |
| 20080215342 | SYSTEM AND METHOD FOR ENHANCING PERCEPTUAL QUALITY OF LOW BIT RATE COMPRESSED AUDIO DATA - A system and method for converting an audio data is described. The method includes separating the audio data into a first set of data and a second set of data. The method further includes converting the first set of data into a track of the audio data. The method also includes converting the second set of data into an at least one created sound and a reference to each created sound. The method includes mapping the at least one reference to the created sound to an at least one position in the track where the created sound is to be played when the track is played. | 09-04-2008 |
| 20080215343 | AUDIO DECODING APPARATUS AND AUDIO DECODING SYSTEM - A work memory for storing a compressed audio stream to be decoded, a plurality of decoding process sections, and a plurality of output process sections are provided. A control section switches the start and stop of a decoding process between each decoding process section in predetermined switching cycles so that only any one of the decoding process sections uses the work memory to perform a decoding process at any single point of time, and saves, into an outside of an audio decoding apparatus, a compressed audio stream stored in the work memory which has been referenced by a decoding process section to be caused to stop a decoding process. When a stopped decoding process section is caused to resume a decoding process, the control section reads a compressed audio stream saved for the decoding process section to be caused to resume the decoding process, from the outside. | 09-04-2008 |
| 20080215344 | METHOD AND APPARATUS FOR EXPANDING BANDWIDTH OF VOICE SIGNAL - A method and apparatus for expanding a bandwidth of an input narrowband voice signal is provided. The narrowband voice signal is analyzed separately for each frame, and a Degree of Voicing (DV) and a Degree of Stationary (DS) are calculated depending on the analysis. A Degree of Difficulty of Bandwidth Expansion (DDBWE) of the narrowband voice signal is calculated based on DV and DS. Bandwidth expansion is controlled according to DDBWE. | 09-04-2008 |
| 20080221905 | Encoding an Information Signal - The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames. | 09-11-2008 |
| 20080221906 | Speech coding system and method - A system for enhancing a signal regenerated from an encoded audio signal. The system comprises a decoder arranged to receive the encoded audio signal and produce a decoded audio signal, a feature extraction means arranged to receive at least one of the decoded and encoded audio signal and extract at least one feature from at least one of the decoded and encoded audio signal, a mapping means arranged to map the at least one feature to an enhancement signal and operable to generate and output the enhancement signal, whereby the enhancement signal has a frequency band that is within the decoded audio signal frequency band, and a mixing means arranged to receive the decoded audio signal and the enhancement signal and mix the enhancement signal with the decoded audio signal. | 09-11-2008 |
| 20080221907 | 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. | 09-11-2008 |
| 20080221908 | MULTI-CHANNEL AUDIO ENCODING AND DECODING - An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes. | 09-11-2008 |
| 20080228499 | Method For Generating Encoded Audio Signal and Method For Processing Audio Signal - A method for generating an encoded audio signal, and a method for processing the same during the multi-channel C audio coding are disclosed. The present invention provides the method for generating an encoded audio signal comprising: including fixed channel configuration information acting as configuration information of a predetermined output channel; and including arbitrary channel configuration information. | 09-18-2008 |
| 20080228500 | METHOD AND APPARATUS FOR ENCODING/DECODING AUDIO SIGNAL CONTAINING NOISE AT LOW BIT RATE - A method and apparatus for encoding/decoding audio signals at a low bit rate are provided. The encoding apparatus selectively encodes one or more reference samples having the highest amplitudes among frequency samples of an audio signal, determines amplitudes of the remaining samples other than the reference sample according to a predetermined pattern, using a predetermined random function, and then encodes the remaining samples using random function input information for causing the same pattern to be generated using the same random function, thus maximizing an encoding rate for an audio signal having a large amount of noise. | 09-18-2008 |
| 20080228501 | 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. | 09-18-2008 |
| 20080228502 | 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 group reference value corresponding to a plurality of data included in one group through first grouping and external grouping for the first grouping and a difference value corresponding to the group reference value and obtaining the data using the group reference value and the difference value. | 09-18-2008 |
| 20080235033 | METHOD AND APPARATUS FOR ENCODING AUDIO SIGNAL, AND METHOD AND APPARATUS FOR DECODING AUDIO SIGNAL - Methods and apparatuses for encoding and decoding of an audio signal using a mixture of a time-frequency method and a parametric method according to the audio band are provided. An encoding method of an audio signal includes: dividing input audio signals into a plurality of audio bands; selecting a coding method for each audio band; encoding each audio band according to the selected coding method for each band; and generating a bit stream including all the data encoded for each audio band, wherein selecting a coding method for each band comprises selecting smaller encoded data either from a parametric coding method or a time-frequency coding method. | 09-25-2008 |
| 20080235034 | METHOD AND APPARATUS FOR ENCODING AUDIO SIGNAL AND METHOD AND APPARATUS FOR DECODING AUDIO SIGNAL - Provided are a method and apparatus for encoding an audio signal to efficiently encode a harmonic envelope and a method and apparatus for decoding an audio signal to decode the harmonic envelope. The method of encoding an audio signal includes performing harmonic analysis with respect to an input signal to determine harmonic parameters with respect to harmonic signals; regarding amplitudes of the harmonic signals included in the harmonic parameters as signals in the time domain so as to perform time-frequency transformation; and encoding the time-frequency transformed values. When expressing a harmonic envelope, the amplitudes of the harmonic signals are regarded as signals in the time domain so as to perform a time-frequency transformation and only a part from among the transformed values is selected to be encoded. Therefore, sound quality is not affected and coding efficiency greatly improves. | 09-25-2008 |
| 20080235035 | Method For Decoding An Audio Signal - The invention relates to a method for decoding an audio signal, to allow an audio signal to be compressed and transferred more efficiently. The inventive method comprises steps of receiving an audio signal with spatial information signal, obtaining location information using the number of time slot and parameter of audio signal, establishing a multi-channel audio signal by applying spatial information signal to down-mix signal, and performing a multi-channel array for a multi-channel audio signal in response to the output channel. | 09-25-2008 |
| 20080235036 | Method For Decoding An Audio Signal - The invention relates to a method for decoding an audio signal, to allow an audio signal to be compressed and transferred more efficiently. The inventive method comprises steps of receiving an audio signal with spatial information signal, obtaining location information using the number of time slot and parameter of audio signal, establishing a multi-channel audio signal by applying spatial information signal to down-mix signal, and performing a multi-channel array for a multi-channel audio signal in response to the output channel. | 09-25-2008 |
| 20080243518 | System And Method For Compressing And Reconstructing Audio Files - A system and method for the improved compression of audio signals and the restoration and enhancement of audio recordings missing high frequency content. In the preferred embodiment the different context models are applied to increase the compression ratio of spectral information, quantization coefficients and other information. Context models and arithmetic compression are used for final compression. The time-frequency amplitude envelope and degree of tonality parameters are extracted from the low frequency component. An estimate of the high frequency component is performed by applying a multiband distortion effect, waveshaping, to the low frequency content. Control of tonality is achieved by varying the number of bands within the multiband framework. A filterbank is used that roughly shapes the reconstructed high frequency component according to an estimation of the most probable shape. | 10-02-2008 |
| 20080243519 | Method For Decoding An Audio Signal - The invention relates to a method for decoding an audio signal, to allow an audio signal to be compressed and transferred more efficiently. The inventive method comprises steps of receiving an audio signal with spatial information signal, obtaining location information using the number of time slot and parameter of audio signal, establishing a multi-channel audio signal by applying spatial information signal to down-mix signal, and performing a multi-channel array for a multi-channel audio signal in response to the output channel. | 10-02-2008 |
| 20080243520 | AUDIO CODING - A method of encoding a multi-channel audio signal including at least a first signal component (LF), a second signal component (LR) and a third signal component (RF). The method comprises the steps of encoding the first and second signal components by a first parametric encoder ( | 10-02-2008 |
| 20080249783 | Layered Code-Excited Linear Prediction Speech Encoder and Decoder Having Plural Codebook Contributions in Enhancement Layers Thereof and Methods of Layered CELP Encoding and Decoding - A layered code-excited linear prediction (CELP) encoder, an Adaptive Multirate Wideband (AMR-WB) encoder and methods of CELP encoding and decoding. In one embodiment, the encoder includes: (1) a core layer subencoder and (2) at least one enhancement layer subencoder having an adaptive-gain multiplier configured to apply a gain for an adaptive contribution to excitation and a fixed-gain multiplier configured to apply a gain for a fixed contribution to the excitation that is separate from the gain for the adaptive contribution. | 10-09-2008 |
| 20080249784 | Layered Code-Excited Linear Prediction Speech Encoder and Decoder in Which Closed-Loop Pitch Estimation is Performed with Linear Prediction Excitation Corresponding to Optimal Gains and Methods of Layered CELP Encoding and Decoding - A layered code-excited linear prediction (CELP) encoder, an Adaptive Multirate Wideband (AMR-WB) encoder and methods of CELP encoding and decoding. In one embodiment, the encoder includes: (1) a core layer subencoder and (2) at least one enhancement layer subencoder, at least one of the core layer subencoder and the enhancement layer subencoder having first and second adaptive codebooks and configured to retrieve a pitch lag estimate from the second adaptive codebook and perform a closed-loop search of the first adaptive codebook based on the pitch lag estimate. | 10-09-2008 |
| 20080255855 | Method and apparatus for coding and decoding amplitude of partial - Provided are a method and apparatus for coding and decoding an amplitude of partials, in which a step phenomenon can be prevented in the result of coding the amplitude of continuation partial partials in a parametric codec, thereby improving reproduced sound quality. The method of coding the amplitude of partials includes obtaining an inversely quantized amplitude of partials of a previous frame, determining a quantization level based on a function for the inversely quantized amplitude of the partials of the previous frame, and quantizing an amplitude of partials of a current frame based on the determined quantization level. | 10-16-2008 |
| 20080255856 | Audio Encoding and Decoding - An audio encoder ( | 10-16-2008 |
| 20080255857 | 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. | 10-16-2008 |
| 20080255858 | 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 reference value corresponding to a plurality of data and a difference value corresponding to the reference value and entropy-decoding the difference value and obtaining the data using the reference value and the entropy-decoded difference value. | 10-16-2008 |
| 20080255859 | Method for Encoding and Decoding Multi-Channel Audio Signal and Apparatus Thereof - Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the decoding method, a down-mix signal is generated based on an input signal, and spatial information is generated based on the down-mix signal through estimation. Then, a multi-channel audio signal is generated based on the down-mix signal and the spatial information. Therefore, it is possible to compensate for a down-mix signal or generate additional spatial information by using additional information. | 10-16-2008 |
| 20080262850 | Adaptive Bit Allocation for Multi-Channel Audio Encoding - The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder ( | 10-23-2008 |
| 20080262851 | 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 number of data corresponding to a pilot reference value and if a number of data bands meets a predetermined condition, obtaining the pilot reference value 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. The number of the data is obtained using the number of the data bands in which the data are included. The present invention includes the steps of deciding one of a plurality of data coding schemes using a number of data and decoding the data according to the decided data coding scheme, wherein a plurality of the data coding schemes include a pilot coding scheme at least. | 10-23-2008 |
| 20080262852 | 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 difference value and index information and entropy-decoding the index information and identifying an entropy table corresponding to the entropy-decoded index information and entropy-decoding the difference value using the identified entropy table and obtaining data using a reference value corresponding to a plurality of data and the decoded difference value. | 10-23-2008 |
| 20080262853 | Method for Encoding and Decoding Multi-Channel Audio Signal and Apparatus Thereof - Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal. | 10-23-2008 |
| 20080262854 | Method for Encoding and Decoding Multi-Channel Audio Signal and Apparatus Thereof - Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information that is calculated based on a multi-channel audio signal and a downmix signal is encoded, and additional configuration information is generated based on information that is selected from the encoded spatial information. The downmix signal is encoded, and then, a bitstream is generated by combining the encoded downmix signal with the encoded spatial information. Thereafter, the additional configuration information is inserted into the bitstream. Therefore, it is possible to configure an optimum bitstream according to the circumstances by retransmitting all or part of information included in a header. | 10-23-2008 |
| 20080262855 | ENTROPY CODING BY ADAPTING CODING BETWEEN LEVEL AND RUN LENGTH/LEVEL MODES - An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding. | 10-23-2008 |
| 20080270143 | Method and Apparatus for Processing Encoded Audio Data - To locate an encoded audio frame boundary and begin decoding audio at a point corresponding to that frame boundary, an audio decoder generates a matching pattern containing a syncword and additional bits related to a header of an encoded audio frame, detects an audio frame boundary by searching a data stream of encoded audio frame for instances of the matching pattern, and begins decoding audio frames at a point in the data stream corresponding to the detected frame boundary. | 10-30-2008 |
| 20080270144 | 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 mode information and obtaining a pilot reference value corresponding to a plurality of data and a pilot difference value corresponding to the pilot reference value according to data attribute indicated by the mode information and obtaining the data using the pilot reference value and the pilot difference value. | 10-30-2008 |
| 20080270145 | 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 group reference value corresponding to a plurality of data included in one group through grouping and a difference value corresponding to the group reference value and obtaining the data using the group reference value and the difference value. | 10-30-2008 |
| 20080270146 | 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 group reference value corresponding to a plurality of data included in one group and a difference value corresponding to the group reference value through first grouping and internal grouping for the first grouping and obtaining the data using the group reference value and the difference value. | 10-30-2008 |
| 20080270147 | 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 group reference value corresponding to a plurality of data included in one group through data grouping and internal grouping for the data grouping and a difference value corresponding to the group reference value and obtaining the data using the group reference value and the difference value. | 10-30-2008 |
| 20080275709 | Audio Encoding and Decoding - A method of encoding a digital audio signal, wherein for each time segment the signal is spectrally flattened to obtain a spectrally flattened signal (r) and possibly spectral flattening parameters (LPP). The spectrally flattened signal is modelled by an excitation signal comprising a first partial excitation signal (p | 11-06-2008 |
| 20080275710 | Scale Searching for Watermark Detection - The present invention relates to a method, device ( | 11-06-2008 |
| 20080275711 | Method and Apparatus for Decoding an Audio Signal - Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes receiving filter information, applying spatial information to the filter information to generate surround converting information, and outputting the surround converting information. The apparatus for decoding an audio signal includes a filter information receiving part receiving filter information; an information converting part applying spatial information to the filter information to generate surround converting information, and a surround converting information output part outputting the surround converting information. | 11-06-2008 |
| 20080275712 | 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 gains and a pilot difference value corresponding to the pilot reference value; and obtaining the gain using the pilot reference value and the pilot difference value. | 11-06-2008 |
| 20080281602 | Coding Reverberant Sound Signals - The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In the encoder an audio signal is split into an anechoic signal part and information regarding a reverberant field associated with the audio signal, preferably by a representation using only few parameters such as reverberation time and reverberation amplitude. The anechoic signal is then encoded using an audio codec. At the decoder the anechoic signal part is restored using the audio codec, and the restored anechoic signal part is transformed into the substantially original audio signal by applying reverberance according to the information regarding the reverberant field, preferably by convolution with a room impulse response generated on the basis of the reverberant field information. According to the invention the audio codec involved needs only be capable of encoding anechoic audio signals, thus solving the problem of parametric audio codecs providing poor performance on reverberant audio signals. | 11-13-2008 |
| 20080281603 | Sound encoder and sound decoder - A sound encoder multiplexes a plurality of codes into a sound code in an order determined by a multiplexing order determination unit ( | 11-13-2008 |
| 20080281604 | METHOD AND APPARATUS TO ENCODE AND DECODE AN AUDIO SIGNAL - A method and apparatus to encode and decode an audio signal. In the encoding method and apparatus, one or more important frequency components may be detected from an audio signal, the frequency components may be encoded, and then an envelope of the audio signal may be encoded. In the decoding method and apparatus, an audio signal may be decoded by adjusting envelopes at one or more bands containing one or more important frequency components in consideration of the energy values of the frequency components. Accordingly, it is possible to maximize the coding efficiency without degrading the sound quality of the audio signal even if the audio signal is encoded or decoded using a small amount of bits. | 11-13-2008 |
| 20080288261 | METHOD FOR DYNAMICALLY ADJUSTING AUDIO DECODING PROCESS - A method for dynamically arranging DSP tasks. The method comprises receiving an audio bit stream, checking a remaining execution time as the DSP transforms the audio information into spectral information, simplifying the step of transforming the audio information when the DSP detects that the remaining execution time is shorter then a predetermined interval, and skipping one section of the audio information and decoding the remaining section when the execution time is less than a predetermined interval. | 11-20-2008 |
| 20080288262 | Decoding apparatus and decoding method - A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data. | 11-20-2008 |
| 20080288263 | Method and Apparatus for Encoding/Decoding - An encoding method and apparatus and a decoding method and apparatus are provided. The encoding method includes encoding data, generating a data bit array comprising the encoded data and encoding information, and generating an align bit array comprising one or more Is. According to the encoding method and apparatus and the decoding method and apparatus, it is possible to enhance the quality of decoded data by inserting various information into an align bit array that is included in a bitstream for aligning a plurality of encoded data bit arrays with one another. In addition, according to the encoding method and apparatus and the decoding method and apparatus, it is possible to efficiently utilize data bandwidths that are allocated for the encoding/decoding of data at low bitrates by enhancing the efficiency of data encoding. | 11-20-2008 |
| 20080294444 | Method and Apparatus for Decoding an Audio Signal - Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes extracting a downmix signal and spatial information from a received audio signal and generating a pseudo-surround signal using the downmix signal and the spatial information. The apparatus for decoding an audio signal includes a demultiplexing part extracting a downmix signal and spatial information from a received audio signal and a pseudo-surround decoding part generating a pseudo-surround signal from the downmix signal, using the spatial information. | 11-27-2008 |
| 20080294445 | METHOD AND APAPRATUS FOR SINUSOIDAL AUDIO CODING - Provided are a method and apparatus for sinusoidal audio coding, which employs a tracking method for further effective coding of sinusoids extracted in the process of a sinusoidal analysis of parametric coding. The sinusoidal audio coding method includes: extracting sinusoids of a current frame by performing a sinusoidal analysis on an input audio signal; with respect to each of the extracted sinusoids, setting a mode selected from a birth mode in which a sinusoid is newly generated irrespective of sinusoids of a previous frame, a continuation mode in which the sinusoid is only one sinusoid continued from one of the sinusoids of the previous frame, and a branch mode in which the sinusoid is one of a plurality of sinusoids continued from one of the sinusoids of the previous frame; and coding the extracted sinusoids according to the selected mode. Accordingly, a plurality of sinusoids that can be continued from one previous track component are set to the continuation mode or the branch mode. Therefore, the number of bits of coded data is significantly reduced, compared with the case of the birth mode. | 11-27-2008 |
| 20080306744 | Device and a Method of Playing Audio Clips - A device for playing audio clips, a method of playing audio clips and a data storage medium having stored thereon computer code means for instructing a computer system to execute a method of playing audio clips. The device comprising a processor scalable in voltage, frequency, or both; a switch for selecting one of a plurality of output modes of the device; and a controller for controlling the processor to decode input joint/MS stereo mode encoded audio data representing said audio clip based on the selected output mode; wherein each output mode defines a number, m, of subbands of an M channel and a number, s, of subbands of an S channel of the joint/MS stereo mode encoded data; and the controller controls the processor to decode and store only data from the m and s subbands for playback. | 12-11-2008 |
| 20080306745 | Distributed audio coding for wireless hearing aids - The aim of the invention is to provide inter-channel level differences ICLD related to audio signals for hearing aids. This aim is achieved by a method for computing ICLD from a first and second audio source signals, the first source signal being wired with a first processing module and the second source signal being wired with a second processing module, the second processing module receiving wirelessly information from the first processing module, this method comprising the steps of: acquiring first samples of the first sound signal by the first processing module, defining a first time frame, converting the first time frame into first frequency bands and grouping them into two first frequency sub-bands, calculating a first power estimate of each first frequency sub-bands, encoding and transmitting same to the second processing module, acquiring second samples of the second sound signal by the second processing module,
| 12-11-2008 |
| 20080312936 | APPARATUS AND METHOD FOR TRANSMITTING/RECEIVING VOICE DATA TO ESTIMATE VOICE DATA VALUE CORRESPONDING TO RESYNCHRONIZATION PERIOD - Provided are an apparatus and method for estimating a voice data value corresponding to a silent period produced in a key resynchronization process using the sine waveform characteristic of voice when encrypted digital voice data is transmitted in one-way wireless communication environment. The apparatus includes a transmitter that generates a key resynchronization frame containing key resynchronization information and vector information on voice data inserted thereinto and transmits the key resynchronization frame, and a receiver that receives the key resynchronization frame from the transmitter, extracts the vector information inserted in the key resynchronization frame, and estimates a voice data value corresponding to the key resynchronization period. Based on change ratio between slopes calculated using received voice data, it is possible to estimate the voice data corresponding to a silent period, which improves communication quality. | 12-18-2008 |
| 20080319764 | Method for Determining an Audio Data Spatial Encoding Mode - A method of determining a spatial coding mode for audio data sent by a sender entity ( | 12-25-2008 |
| 20080319765 | Method and Apparatus for Decoding a Signal - An apparatus for decoding a signal and method thereof are disclosed, by which the audio signal can be controlled in a manner of changing/giving spatial characteristics (e.g., listener's virtual position, virtual position of a specific source) of the audio signal. The present invention includes receiving an object parameter including level information corresponding to at least one object signal, converting the level information corresponding to the object signal to the level information corresponding to an output channel by applying a control parameter to the object parameter, and generating a rendering parameter including the level information corresponding to the output channel to control an object downmix signal resulting from downmixing the object signal. | 12-25-2008 |
| 20090006102 | Effective Audio Segmentation and Classification | 01-01-2009 |
| 20090006103 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique. | 01-01-2009 |
| 20090006104 | METHOD OF CONFIGURING CODEC AND CODEC USING THE SAME - Provided is a codec that may be configured to adaptively optimize to various service environments by obtaining information on a current operating environment of the codec and configuring the codec in view of quality of service (QoS) parameters based on the obtained information. | 01-01-2009 |
| 20090006105 | Method for Generating Encoded Audio Signal and Method for Processing Audio Signal - A method for generating an encoded audio signal, and a method for processing the same during the multi-channel audio coding are disclosed. The present invention provides the method for generating an encoded audio signal comprising: generating basic spatial information including basic configuration information requisite for a multi-channel audio coding process and basic data corresponding to the basic configuration information; and generating extension spatial information including extension configuration information selectively required for the multi-channel audio coding process and extension data corresponding to the extension configuration information. | 01-01-2009 |
| 20090006106 | Method and Apparatus for Decoding a Signal - An apparatus for decoding a signal and method thereof are disclosed, by which the audio signal can be controlled in a manner of changing/giving spatial characteristics (e.g., listener's virtual position, virtual position of a specific source) of the audio signal. The present invention includes receiving an object parameter; extracting object information by parsing the received object parameter; generating a control parameter using the extracted object information and control information including at least one of user control information, default control information, device control information, and device information; and, generating a rendering parameter determining a position and level of an object in an output signal using the object parameter and the control parameter. | 01-01-2009 |
| 20090012796 | Apparatus and Method for Encoding/Decoding Signal - An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a compatible down-mix signal optimized for a first multi-channel decoder from the input bitstream, converting the compatible down-mix signal to be optimized for a second multi-channel signal by performing a compatibility processing operation on the compatible down-mix signal, and generating a three-dimensional (3D) down-mix signal by performing a 3D rendering operation on the converted down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment. | 01-08-2009 |
| 20090024395 | AUDIO SIGNAL ENCODING METHOD, AUDIO SIGNAL DECODING METHOD, TRANSMITTER, RECEIVER, AND WIRELESS MICROPHONE SYSTEM - It is an object of the present invention to provide an audio signal encoding method, an audio signal decoding method, a transmitter, a receiver, and a wireless microphone system which can compress an audio signal at a relatively high compression ratio at a relatively high quality with a relatively low delay. The compression encoder | 01-22-2009 |
| 20090024396 | AUDIO SIGNAL ENCODING METHOD AND APPARATUS - An audio signal encoding method and apparatus for efficiently encoding an audio signal in an interval having many birth sinusoids and enabling tracking of sinusoidal signals in the next interval, and a computer readable recording medium having embodied thereon a computer program for executing the audio signal encoding method are provided. According to the method and apparatus, by applying transform coding instead of parametric coding to a frame having many birth sinusoids, the sinusoids are encoded, thereby reducing the number of bits required for the encoding and enabling efficient coding. Also, when transform coding is applied to a frame of a predetermined interval, an inverse transform of the transform coding is applied to the encoded data in order to decode the data, and then sinusoids are extracted from the decoded data, thereby enabling tracking of sinusoids of the next frame. | 01-22-2009 |
| 20090024397 | UNIFIED FILTER BANK FOR PERFORMING SIGNAL CONVERSIONS - A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules. | 01-22-2009 |
| 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 |
| 20090024399 | Method and Arrangements for Audio Signal Encoding - To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal. | 01-22-2009 |
| 20090024400 | MULTIPLEXED AUDIO DATA DECODING APPARATUS AND RECEIVER APPARATUS - A multiplexed audio data decoder apparatus is provided in which integration of an audio decoder is easy, and has a high flexibility when the number of the formats to be processed is increased or when the specification is changed. In an external ROM | 01-22-2009 |
| 20090030699 | PROVIDING A CODEBOOK FOR BANDWIDTH EXTENSION OF AN ACOUSTIC SIGNAL - A codebook spectral envelope may be used to extend the bandwidth of a bandwidth limited signal. A system includes codebooks that list codebook spectral envelopes. A codebook spectral envelope may be selected based on a characteristic of the spectral envelope of the bandwidth limited signal. Modifications of selected codebook spectral envelopes may generate a bandwidth extension signal that may be added to the bandwidth limited signal to improve the quality of the signal. | 01-29-2009 |
| 20090030700 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels are synchronously subdivided into blocks such that the lengths of the blocks into which the second channel is subdivided correspond to the lengths of the blocks into which the first channel is subdivided if the first and second channels are correlated with each other and difference coding is used. The first and second channels are decoded and the subdivided blocks of the first and second channels are interleaved if the first and second channels are synchronously subdivided. | 01-29-2009 |
| 20090030701 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels are independently subdivided into blocks if the first and second channels are not correlated with each other. The first and second channels are decoded, and the subdivided blocks of the first and second channels are not interleaved if the first and second channels are independently subdivided. | 01-29-2009 |
| 20090030702 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving an audio data frame having at least one channel. The channel is subdivided into a plurality of blocks, and at least two of the blocks are capable of having different lengths. The embodiment further includes obtaining indicator information indicating whether determining of a prediction order for each block is allowed, and determining the prediction order from the audio signal indicating the prediction order for each block if the indicator information indicating that determining of the prediction order for each block is allowed. The channel is decoded using the prediction order. | 01-29-2009 |
| 20090030703 | 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 first and second channel data is provided jointly if the first and second channel data are paired with each other. Block information indicating the block type is obtained. 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 block information. | 01-29-2009 |
| 20090037180 | TRANSCODING METHOD AND APPARATUS - Provided is a method and apparatus for encoding a bitstream, which was encoded by a predetermined method, in another method. By adaptively encoding a bitstream encoded by a predetermined method by selecting a domain in which the encoding is performed for each predetermined band, the bitstream can be efficiently encoded and transmitted and received, and compatibility can be provided. | 02-05-2009 |
| 20090037181 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes reading random access information from the audio signal. The random access information indicates whether or not random access operation is allowed in the audio signal. If the random access operation is allowed, the audio signal has a plurality of random access units. A random access unit includes one or more of frames and at least one of the frames is a random access frame. The random access frame is a frame encoded such that previous frames are not necessary to decode the random access frame, and the random access information further indicates a distance between random access frames in frames. The random access frame is decoded based on the random access information. | 02-05-2009 |
| 20090037182 | Apparatus and method of processing an audio signal - In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded-coefficient values; and obtaining table index information from the digital audio data. The table index information indicates whether entropy coding of coded-coefficient values is performed, and the table index information further identifies a table from a plurality of tables to select if the table index information indicating that entropy coding of coded-coefficient values is performed. A set of prediction coefficient values is reconstructed from the coded-coefficient values if the table index information indicates that entropy coding of coded-coefficient values is performed. The reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values. | 02-05-2009 |
| 20090037183 | 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. Block information indicating the block type is obtained. 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 is lossless decoded based on the block information. | 02-05-2009 |
| 20090037184 | 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 first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, obtaining block information indicating a block type, and lossless decoding the first and second channel data based on the frame length information and the block information. | 02-05-2009 |
| 20090037185 | 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 is provided jointly if the first and second channel data are paired with each other. The method 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 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 |
| 20090037186 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving the audio signal having configuration information and multi-channels, and reading a first indicator from the configuration information. The first indicator indicates whether or not channel mapping information is included in the configuration information. The channel mapping information is read from the configuration information if the first indicator indicates that the channel mapping information is included in the configuration information. The channel mapping information indicates to which speaker in a reproduction device to map each channel in the audio signal. A second indicator is also read from the configuration information. The second indicator indicates whether or not channel rearrangement information is included in the configuration information. The channel rearrangement information is read from the configuration information if the second indicator indicates that the channel rearrangement information is included in the configuration information. The channel rearrangement information indicates a rearrangement of the multi-channels. The multi-channels are decoded based on the channel mapping information and channel rearrangement information. | 02-05-2009 |
| 20090037187 | Apparatus and method of encoding and decoding audio signals - In one embodiment, the method includes receiving an audio signal having configuration information and at least one channel, and reading an indicator from the configuration information. The indicator indicates whether or not channel mapping information is included in the configuration information. The channel mapping information is read from the configuration information if the indicator indicates that the channel mapping information is included in the configuration information. The channel mapping information indicates to which speaker in a reproduction device to map each channel in the audio signal. The channel is decoded based on the channel mapping information. | 02-05-2009 |
| 20090037188 | Apparatus and method of encoding and decoding audio signals - In one embodiment, the method includes receiving an audio signal having configuration information and multi-channels, and reading an indicator from the configuration information. The indicator indicates whether or not channel mapping information is included in the configuration information. The method further includes reading the channel mapping information from the configuration information if the indicator indicates that the channel mapping information is included in the configuration information. The channel mapping information indicates to which speaker in a reproduction device to map each channel in the audio signal. Channel rearrangement information is read from the configuration information. The channel rearrangement information indicates the rearrangement of the multi-channels. The multi-channels are decoded based on the channel mapping information and channel rearrangement information. | 02-05-2009 |
| 20090037189 | Apparatus and Method for Encoding/Decoding Signal - An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a three-dimensional (3D) down-mix signal and spatial information from an input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using the spatial information and a down-mix signal obtained by the removal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of a reproduction environment. | 02-05-2009 |
| 20090043589 | RECORDING AND SELECTING AN AUDIO REGION - A program product, a graphical user interface, a computer system including such a graphical user interface and method for recording and selecting an audio region are described. Such a method comprises storing multiple takes of the same audio region and storing a data structure representing a take container to store multiple takes. | 02-12-2009 |
| 20090043590 | Noise Detection for Audio Encoding by Mean and Variance Energy Ratio - The techniques described are utilized for detection of noise and noise-like segments in audio coding. The techniques can include performing a prediction gain calculation, an energy compaction calculation, and a mean and variation energy calculation. Signal adaptive noise decisions can be made both in time and frequency dimensions. The techniques can be embodied as part of an AAC (advanced audio coding) encoder to detect noise and noise-like spectral bands. This detected information is transmitted in a bitstream using a signaling method defined for a perceptual noise substitution (PNS) encoding tool of the AAC encoder | 02-12-2009 |
| 20090043591 | AUDIO ENCODING AND DECODING - An audio encoder comprises a multi-channel receiver ( | 02-12-2009 |
| 20090048846 | Method for Expanding Audio Signal Bandwidth - A method expands a bandwidth of an audio signal by determining a magnitude time-frequency representation |G(ω,t) for example audio signals g(t). A set of frequency marginal probabilities P | 02-19-2009 |
| 20090048847 | Method and Apparatus for Encoding/Decoding Multi-Channel Audio Signal - Methods of encoding and decoding a multi-channel audio signal and apparatuses for encoding and decoding a multi-channel audio signal are provided. The method of decoding a multi-channel audio signal includes an unpacking unit which extracts a quantized CLD between a pair of channels of a plurality of channels from a bitstream, and an inverse quantization unit which inverse-quantizes the quantized CLD using a quantization table that considers the location properties of the pair of channels. The methods of encoding and decoding a multi-channel audio signal and the apparatuses for encoding and decoding a multi-channel audio signal can enable an efficient encoding/decoding by reducing the number of quantization bits required. | 02-19-2009 |
| 20090048848 | Method And System For Media Processing Extensions (MPX) For Audio And Video Setting Preferences - Device independent Media Properties Extension (MPX) data, corresponding to media data, may be decoded by a media rendering device and may be utilized to determine and/or execute processing steps and/or processing parameters for processing the media data. During the processing and/or rendering, processing steps and/or parameters may be dynamically determined and/or adjusted. A user preference profile, media rendering device profile and/or media rendering environment profile may be utilized to generate, store and/or restore MPX data. Furthermore, MPX data that may be input by a user, manufacturer or a vendor, may be stored in a plurality of ways, for example, within a media data file, an external file and/or within an MTP or PTP object property associated the media data. The media data may comprise one or more of video data, still image data and audio data, for example. | 02-19-2009 |
| 20090048849 | AUDIO ENCODING METHOD AND APPARATUS, AND AUDIO DECODING METHOD AND APPARATUS, FOR PROCESSING DEATH SINUSOID AND GENERAL CONTINUATION SINUSOID - An audio encoding method and apparatus, and an audio decoding method and apparatus, for processing a death sinusoid and a general continuation sinusoid. Using the unique characteristic of a death sinusoid, in that the death sinusoid has a tendency such that an amplitude component of the death sinusoid is less than that of a previous sinusoid, a method of adding an encoding syntax by distinguishing a general continuation sinusoid from a death sinusoid is provided. That is, when difference coding of the amplitude component of a death sinusoid is performed, the number of bits used when a negative number is coded is less than the number of bits used when a positive number is coded, in a Huffman table. By using this method, a bit rate in an entire coding decreases. | 02-19-2009 |
| 20090048850 | Apparatus and method of processing an audio signal - In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient value are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values. | 02-19-2009 |
| 20090048851 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving an audio data frame having at least one channel. The channel is subdivided into a plurality of blocks, and at least two of the blocks are capable of different lengths. The embodiment further includes obtaining information from the audio signal indicating the subdivision of the channel into the blocks, and decoding the channel based on the obtained information. In one embodiment, the channel is subdivided into the plurality of blocks according to a subdivision hierarchy, the subdivision hierarchy has more than one level, and each level is associated with a different block length. | 02-19-2009 |
| 20090055194 | ENCODING AND DECODING OF MULTI-CHANNEL AUDIO SIGNALS - An encoding device ( | 02-26-2009 |
| 20090055195 | INTERNET RADIO PLAYER - The present invention provides an apparatus for listening to music on the internet. The apparatus includes a memory unit configured to store a plurality of internet protocol addresses of internet radio stations, a communication unit configured to communicate with one of the internet radio stations using a corresponding IP address stored in the memory, a first logic configured to request a data from one of the internet radio stations, a second logic configured to receive a digital data stream from the internet station and store the received digital data stream in a buffer, a third logic configured to decode the received digital data stream, a fourth logic configured to convert the decoded digital data stream into an analog data stream, and a service module configured to extract information regarding a song from the digital data stream and forward the extracted information to an external device. | 02-26-2009 |
| 20090055196 | Method of Encoding and Decoding an Audio Signal - An apparatus for encoding and decoding an audio signal and method thereof are disclosed, by which compatibility with a player of a general mono or stereo audio signal can be provided in coding an audio signal and by which spatial information for a multi-channel audio signal can be stored or transmitted without a presence of an auxiliary data area. The present invention includes extracting side information embedded in non-recognizable component of audio signal components and decoding the audio signal using the extracted side information. | 02-26-2009 |
| 20090055197 | METHOD AND APPARATUS FOR ENCODING CONTINUATION SINUSOID SIGNAL INFORMATION OF AUDIO SIGNAL AND METHOD AND APPARATUS FOR DECODING SAME - Provided are a method and apparatus for encoding an audio signal and a method and apparatus for decoding an audio signal. The method includes performing sinusoidal analysis on an audio signal in order to extract a sinusoidal signal of a current frame, determining continuation sinusoidal signal information indicating a number of continuation sinusoidal signals of next frames, which continue from the sinusoidal signal of the current frame, by performing sinusoidal tracking on the extracted sinusoidal signal of the current frame, and encoding the determined continuation sinusoidal signal information by using different Huffman tables according to index information of the current frame, thereby allowing efficient encoding with a low bitrate. | 02-26-2009 |
| 20090055198 | Apparatus and method of processing an audio signal - In one embodiment, the method includes storing a plurality of entropy parameter tables, selecting one of the stored tables for entropy decoding based on a received audio signal, and entropy decoding at least one prediction coefficient value in the received audio signal based on the selected table. | 02-26-2009 |
| 20090063158 | EFFICIENT AUDIO CODING USING SIGNAL PROPERTIES - An audio encoder comprising optimizing means ET OPT adapted to generate an optimized encoding template OET based on properties PV of an input audio signal IN, such as in form of a property vector. The optimized encoding template OET is being optimized with respect to a predetermined encoding efficiency criterion. Encoding means ENC then generates an encoded audio signal OUT in accordance with the optimized encoding template OET. The audio encoder may comprise analyzing means AN adapted to generate the set of input signal properties PV based of the input signal IN. In a preferred embodiment the optimizing means ET OPT is adapted to estimate a resulting distortion associated with an encoding template. The optimizing means ET OPT may further be able to estimate bit rate associated with an encoding template. In one embodiment the optimizing means ET OPT is adapted to optimize a bit rate distribution to a number of sub-encoders based on the input signal properties (PV). In another embodiment, the optimizing means ET OPT is adapted to up-front decide on an adaptive segmentation based on the input signal properties (PV). The encoders according to the invention are advantageous in that complex processes of a plurality of encodings prior to deciding upon an optimized encoding template OET can be avoided since the optimal encoding template OET is found based on input signal properties (PV). | 03-05-2009 |
| 20090063159 | Audio Metadata Verification - A digital bitstream, comprising data bits representing audio, metadata intended to be correct for the audio, and metadata verification information, wherein all or part of the metadata may not be correct for the audio. The metadata verification information is usable to detect whether or not metadata is correct for the audio and, if not correct, to change it so that it is correct. The metadata verification information usable to detect and change metadata may include a copy, or a data-compressed copy, of a correct version of the metadata. | 03-05-2009 |
| 20090063160 | Configurable common filterbank processor applicable for various audio standards and processing method thereof - A configurable common filterbank processor applicable for various audio standards and its processing method. Inverse modified discrete cosine transform (IMDCT) and window and overlap-add (WOA) decoding operations required by AC-3 and AAC, and IMDC, WOA, and matrixing decoding operations required by MP3 are divided into several different modes, and a quick algorithm is provided for expediting the operation of these modes, and a hardware architecture is designed universally for these modes, so that the hardware architecture can be applicable for the decoding operations of three different audio standards, respectively AC-3, AAC and MP3, to expand the scope of applicability of a decoder. | 03-05-2009 |
| 20090063161 | METHOD AND APPARATUS FOR ENCODING AND DECODING CONTINUATION SINUSOIDAL SIGNAL OF AUDIO SIGNAL - Provided are an audio signal encoding method and apparatus that encode a continuation sinusoidal signal of a current frame in different ways according to information on a sinusoidal signal of a previous frame by using the characteristics of the continuation sinusoidal signal, and an audio signal decoding method and apparatus. The audio signal encoding method includes extracting a sinusoidal signal of a current frame by performing sinusoidal analysis on an input audio signal; extracting a continuation sinusoidal signal of the current frame, which is connected to a sinusoidal signal of a previous frame, by performing sinusoidal tracking of the extracted sinusoidal signal of the current frame; and encoding the continuation sinusoidal signal in different ways by using information on the sinusoidal signal of the previous frame, which is connected to the continuation sinusoidal signal. | 03-05-2009 |
| 20090063162 | PARAMETRIC AUDIO ENCODING AND DECODING APPARATUS AND METHOD THEREOF - Provided are parametric audio encoding and decoding apparatuses and methods thereof. In the parametric audio encoding method, an audio signal is segmented into a plurality of segments. At least one sine wave is extracted from each of the segments, and the extracted sine waves are connected. It is determined whether an extracted sine wave is a birth sine wave. If the extracted sine wave is a birth sine wave, a bit stream is generated by encoding the phase of the birth sine wave on the basis of the frequency of the birth sine wave, wherein the number of bits allocated to encode the phase of the birth sine wave is adjusted according to the frequency of the birth sine wave. | 03-05-2009 |
| 20090063163 | METHOD AND APPARATUS FOR ENCODING/DECODING MEDIA SIGNAL - Provided are a method and apparatus for encoding/decoding a media signal. The method of encoding a media signal includes: when harmonics exist in a sinusoid of a previous frame section, predicting a harmonic frequency of a current frame section that is to be encoded by using a harmonic frequency of the previous frame section, and generating a residual signal by using a difference between the predicted frequency and an actual harmonic frequency of the current frame section. | 03-05-2009 |
| 20090063164 | METHOD AND SYSTEM FOR REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES AND GENERAL PURPOSE AUDIO CODEC - A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms. | 03-05-2009 |
| 20090063165 | SYSTEM AND METHOD FOR PROVIDING AMR-WB DTX SYNCHRONIZATION - A system and method for providing improved adaptive multi-rate wideband (AMR-WB) discontinuous transmission (DTX) synchronization. According to various embodiments, an indication on the start of the inactive speech period is signalled to the decoder via a voice activity detection (VAD) flag a predetermined number of frames before the DTX period will start, i.e., before the SID_FIRST frame is received. When the VAD flag indicates active speech, or when the VAD flag has been set to zero less than the predetermined number of frames ago, the received NO_DATA frame can be classified with a high degree of reliability as active speech, i.e., considered as transmitter, network or terminal-initiated signalling, and can be substituted by a SPEECH_LOST frame. When the VAD flag was set to zero eight frames ago or earlier, the NO_DATA frame is classified as DTX. | 03-05-2009 |
| 20090070118 | AUDIO CODING AND DECODING - An audio encoding device ( | 03-12-2009 |
| 20090070119 | POWER EFFICIENT BATCH-FRAME AUDIO DECODING APPARATUS, SYSTEM AND METHOD - Power savings in a mobile device is accomplished by generating audio samples by decoding a bitstream with a decoding system within the mobile device. The generated audio samples are transferred into at least one memory bank in a set of memory banks in a power saver block within the mobile device. Parts of the decoding system not involved in the storing of the generated audio samples are switched off after batch decoding a bitstream associated with multiple audio frames. The bitstream includes bits less than that found in one audio file. At least one of the memory banks in the set of memory banks is power collapsible. The fetching of the decoded by the decoding system can be synchronized with a paging channel of a modem in the mobile device. The transferred audio samples is a lossless compression and may occur after a re-encoding. | 03-12-2009 |
| 20090070120 | Audio regeneration method - According to an aspect of an embodiment, a method for regenerating an audio signal including a low frequency component and a high frequency component by decoding a coded data including a first coded data and a second coded data, the method comprising the steps of: generating the low frequency component; generating the high frequency component; determining whether the low frequency component has transient characteristics or not; generating a low frequency correction component by removing a stationary component when the audio signal has the transient characteristics; generating a corrected high frequency component by correcting the high-frequency component on the basis of the duration of the low frequency correction component when the audio signal has the transient characteristics; and regenerating the audio signal by synthesizing the low frequency component with the corrected high-frequency component. | 03-12-2009 |
| 20090076828 | SYSTEM AND METHOD OF DATA ENCODING - A data encoding device is provided, comprising: a data frame encoder configured to receive an incoming frame of data and to generate an encoded frame of data using a frame encoding scheme; a data frame decoder configured to receive the encoded frame of data and to generate a decoded frame of data using the frame encoding scheme, and frame decoding parameters; a subtractor configured to subtract the decoded frame of data from the incoming frame of data to generate base quantization noise information; a quantization noise encoder configured to receive the base quantization noise information and the frame decoding parameters, and to generate encoded quantization noise information using a noise encoding scheme; and a transmitting circuit configured to transmit the encoded frame of data and the encoded quantization noise information. | 03-19-2009 |
| 20090076829 | Device for Perceptual Weighting in Audio Encoding/Decoding - A hierarchical audio coder for use in a frequency band divided into adjacent first and second sub-bands, said coder comprising: a core coder ( | 03-19-2009 |
| 20090076830 | Methods and Arrangements for Audio Coding and Decoding - A method for audio coding and decoding comprises primary encoding of a present audio signal sample into an encoded representation (T(n)), and non-causal encoding of a first previous audio signal sample into an encoded enhancement representation (ET(n−N+)). The method further comprises providing of the encoded representations to an end user. At the end user, the method comprises primary decoding of the encoded representation (T*(n)) into a present received audio signal sample, and non-causal decoding of the encoded enhancement representation (ET*(n−N+)) into an enhancement first previous received audio signal sample. The method further comprises improving of a first previous received audio signal sample, corresponding to the first previous audio signal sample, based on the enhancement first previous received audio signal sample. Devices and systems for audio coding and decoding are also presented. | 03-19-2009 |
| 20090083040 | ENCODING AND DECODING A SET OF SIGNALS - An encoding device ( | 03-26-2009 |
| 20090083041 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD - There is provided an audio encoding device capable of effectively encoding a stereo audio even when a correlation between channels of the stereo audio is small. In the device, a monaural signal generation unit ( | 03-26-2009 |
| 20090083042 | Encoding Method and Encoding Apparatus | 03-26-2009 |
| 20090083043 | METHOD OF CODING A SOURCE AUDIO SIGNAL, CORRESPONDING CODING DEVICE, DECODING METHOD AND DEVICE, SIGNAL, COMPUTER PROGRAM PRODUCTS - A method is provided for coding a source audio signal. The method includes the following steps: coding a quantization profile of coefficients representative of at least one transform of the source audio signal, according to at least to distinct coding techniques, delivering at least two sets of data representative of a quantization profile; selecting one of the sets of data representative of a quantization profile, as a function of a predetermined selection criterion; transmitting and/or storing the set of data representative of a selected quantization profile and an indicator representative of the corresponding coding technique. | 03-26-2009 |
| 20090083044 | Device and Method for Encoding by Principal Component Analysis a Multichannel Audio Signal - A system and a method for coding by principal component analysis (PCA) of a multi-channel audio signal comprising the following steps: decomposing at least two channels (L, R) of said audio signal into a plurality of frequency sub-bands ( | 03-26-2009 |
| 20090083045 | Device and Method for Graduated Encoding of a Multichannel Audio Signal Based on a Principal Component Analysis - A system and a method for the scalable coding of a multi-channel audio signal comprising a principal component analysis (PCA) transformation of at least two channels (L, R) of the audio signal into a principal component (CP) and at least one residual sub-component (r) by rotation defined by a transformation parameter (θ), comprising the following steps: formation of a frequency subband-based residual structure (Sf | 03-26-2009 |
| 20090083046 | EFFICIENT CODING OF DIGITAL MEDIA SPECTRAL DATA USING WIDE-SENSE PERCEPTUAL SIMILARITY - Traditional audio encoders may conserve coding bit-rate by encoding fewer than all spectral coefficients, which can produce a blurry low-pass sound in the reconstruction. An audio encoder using wide-sense perceptual similarity improves the quality by encoding a perceptually similar version of the omitted spectral coefficients, represented as a scaled version of already coded spectrum. The omitted spectral coefficients are divided into a number of sub-bands. The sub-bands are encoded as two parameters: a scale factor, which may represent the energy in the band; and a shape parameter, which may represent a shape of the band. The shape parameter may be in the form of a motion vector pointing to a portion of the already coded spectrum, an index to a spectral shape in a fixed code-book, or a random noise vector. The encoding thus efficiently represents a scaled version of a similarly shaped portion of spectrum to be copied at decoding. | 03-26-2009 |
| 20090094037 | Adaptive Approach to Improve G.711 Perceptual Quality - In order to achieve the best improvement of ITU G.711 related codec perceptual quality, perceptual weighting controlling parameter(s) should be at least adaptive to relative quantization error statistics or adaptive to signal level. When the relative quantization error statistics are larger or the signal level is lower, the perceptual weighting should be “stronger”, which means α in (5) is smaller; when the relative quantization error statistics are smaller or the signal level is larger, the perceptual weighting should be “weaker”, which means α in (5) is larger. | 04-09-2009 |
| 20090094038 | EFFICIENT DESIGN OF MDCT / IMDCT FILTERBANKS FOR SPEECH AND AUDIO CODING APPLICATIONS - A more efficient encoder/decoder is provided in which an N-point MDCT transform is mapped into smaller sized N/2-point DCT-IV and/or DCT-II transforms with isolated pre-multiplications which can be moved to a prior or subsequent windowing stage. That is, the windowing operations may be merged with first/last stage multiplications in the core MDCT/IMDCT functions, respectively, thus reducing the total number of multiplications. Additionally, the MDCT may be systematically decimated by factor of 2 by utilizing a uniformly scaled 5-point DCT-II core function as opposed to the DCT-IV or FFT cores used in many existing MDCT designs in audio codecs. The modified windowing stage merges factors from a transform stage and windowing stage to obtain piece-wise symmetric windowing factors, which can be represented by a sub-set of the piece-wise symmetric windowing factors to save storage space. Such features offer appreciable reduction in complexity and less memory usage than the prior art. | 04-09-2009 |
| 20090099851 | ADAPTIVE BIT POOL ALLOCATION IN SUB-BAND CODING - In a Bluetooth™ Sub-band Codec (SBC), the size of a bit pool allocated for encoding is adapted in a manner that is dependent upon the audio content that is being encoded. In one implementation, the size of the bit pool is increased during periods when the audio input signal represents an active audio signal, such as speech or music, and decreased when the audio input signal represents background noise or silence. This has the effect of increasing the bit rate (and thus the audio quality) in the presence of speech or music but decreasing the bit rate (and thus the audio quality) in the presence of background noise or silence. By adapting the size of the bit pool in this manner, the quality of the encoded bit-stream transmitted from the SBC encoder may be improved or power may be conserved depending upon the implementation. | 04-16-2009 |
| 20090106030 | METHOD OF SIGNAL ENCODING - There is described a method of encoding a signal (s(n)) in a coder ( | 04-23-2009 |
| 20090106031 | Method and Apparatus for Re-Encoding Signals - At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction. | 04-23-2009 |
| 20090106032 | Apparatus and method of processing an audio signal - In one embodiment, the method includes receiving the audio signal including a block of audio data partitioned into N sub-blocks, and restoring a plurality of code parameters s( | 04-23-2009 |
| 20090112606 | CHANNEL EXTENSION CODING FOR MULTI-CHANNEL SOURCE - A multi-channel audio decoder reconstructs multi-channel audio of more than two physical channels from a reduced set of coded channels based on correlation parameters that specify a full power cross-correlation matrix of the physical channels, or merely preserve a partial correlation matrix (such as power of the physical channels, and some subset of cross-correlations between the physical channels, or cross-correlations of the physical channels with coded or virtual channels). | 04-30-2009 |
| 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 |
| 20090119110 | Method of Encoding and Decoding an Audio Signal - An apparatus for encoding and decoding an audio signal and method thereof are disclosed, by which compatibility with a player of a general mono or stereo audio signal can be provided in coding an audio signal and by which spatial information for a multi-channel audio signal can be stored or transmitted without a presence of an auxiliary data area. The present invention includes extracting side information embedded in non-recognizable component of audio signal components and decoding the audio signal using the extracted side information. | 05-07-2009 |
| 20090119111 | STEREO ENCODING DEVICE, AND STEREO SIGNAL PREDICTING METHOD - A prediction performance between the individual channels of a stereo signal is improved to improve the sound quality of a decoded signal. An LPF ( | 05-07-2009 |
| 20090132258 | APPARATUS, SERVER, METHOD, AND TANGIBLE MACHINE-READABLE MEDIUM THEREOF FOR PROCESSING AND RECOGNIZING A SOUND SIGNAL - An apparatus, a server, a method, and a tangible machine-readable medium thereof for processing and recognizing a sound signal are provided. The apparatus is configured to sense the sound signal of the environment and to dynamically derive and to transmit a feature signal and a sound feature message of the sound signal to the server. The server is configured to retrieve the stored sound models according to the sound feature message and to compare each of the sound models with the feature signal to determine whether the sound signal is abnormal after receiving the feature signal and the sound feature message. | 05-21-2009 |
| 20090132259 | SYSTEMS AND METHODS OF REMOTELY ENABLING SOUND ENHANCEMENT TECHNIQUES - A system and method of remotely enabling sound enhancement techniques is disclosed. In an embodiment, a watermark is embedded in an encoded multi-channel audio stream to remotely enable an enhancement decoder portion of a multi-channel audio decoder. | 05-21-2009 |
| 20090132260 | Method and Apparatus for Improving the Quality of Speech Signals - Methods and apparatus are disclosed to extend the bandwidth of a speech communication to yield a perceived higher quality speech communication for an enhanced user experience. In one aspect of the invention, for example, methods and apparatus can be used to extend the bandwidth of a speech communication beyond a band-limited region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is otherwise characterized absent such bandwidth extension. In another aspect of the invention, for example, methods and apparatus can be used to substitute for corrupt, missing or lost components of a given speech communication, or to otherwise enhance the perceived quality of a speech communication, by extending the speech communication to include one or more artificially created points within the region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is characterized. The result is a speech communication that is perceived to be of higher quality. The various aspects of the present invention can be applied, for example, to network devices or to end-terminal devices. | 05-21-2009 |
| 20090132261 | Methods for Improving High Frequency Reconstruction - The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit. | 05-21-2009 |
| 20090138271 | PARAMETRIC AUDIO CODING COMPRISING AMPLITUDE ENVELOPS - An audio encoder comprising a sinusoidal type encoder and an amplitude modulation encoder that both receive an audio input signal. The amplitude modulation encoder generates a set of sinusoidal components each having assigned individual parameter(s) relating to a time-varying amplitude envelope. The sinusoidal type encoder may be a conventional constant amplitude type encoder and generate a set of constant sinusoidal components. Based on an optimisation using a predetermined encoding efficiency criterion, such as a perceptually relevant criterion, the audio encoder decides which components from the two encoders to be included in an output bit stream. In a preferred embodiment only components from one of the two encoders are used. Preferably, the optimisation process is repeated for each audio signal segment, and preferably a flag for each segment is included in the bit stream indicating if amplitude envelope parameters are present in the segment or not. The invention in addition relates to an audio decoder, methods of encoding and decoding as well as an encoded signal and devices comprising an encoder and a decoder. Audio coding according to the invention provides a high sound quality for transient sounds echo effects while it is still hit rate efficient since amplitude envelopes are included only if proven rate efficient. | 05-28-2009 |
| 20090138272 | WIDEBAND AUDIO SIGNAL CODING/DECODING DEVICE AND METHOD - Disclosed is a wideband audio signal coding/decoding device and method that may code a wideband audio signal while maintaining a low bit rate. | 05-28-2009 |
| 20090144062 | Method and Apparatus to Facilitate Provision and Use of an Energy Value to Determine a Spectral Envelope Shape for Out-of-Signal Bandwidth Content - One provides ( | 06-04-2009 |
| 20090144063 | METHOD AND APPARATUS FOR CONTROL OF RANDERING MULTIOBJECT OR MULTICHANNEL AUDIO SIGNAL USING SPATIAL CUE - The present research relates to controlling rendering of multi-object or multi-channel audio signals. The present research provides a method and apparatus for controlling rendering of multi-object or multi-channel audio signals based on spatial cues in a process of decoding the multi-object or multi-channel audio signals. To achieve the purpose, the method suggested in the research controls rendering in a spatial cue domain in the process of decoding the multi-object or multi-channel audio signals. | 06-04-2009 |
| 20090150161 | SYNCHRONIZING PARAMETRIC CODING OF SPATIAL AUDIO WITH EXTERNALLY PROVIDED DOWNMIX - Embodiments of the present invention are directed to a binaural cue coding (BCC) scheme in which an externally provided audio signal (e.g., a studio engineering audio signal) is transmitted, along with derived cue codes, to a receiver instead of an automatically downmixcd audio signal. The cue codes are (adaptively) synchronized with the externally provided audio signal to compensate for time lags (and changes in those time lags) between the externally downmixed audio signal and the multi-channel signal used to generate the cue codes. If the receiver is a legacy receiver, then the studio engineered audio signal will typically provide a higher-quality playback than would be provided by the automatically downmixed audio signal. If the receiver is a BCC-capable receiver, then the synchronization of the cue codes with the externally provided audio signal will typically improve the quality of the synthesized playback. | 06-11-2009 |
| 20090150162 | STEREO ENCODING APPARATUS, STEREO DECODING APPARATUS, AND THEIR METHODS - A stereo audio encoding apparatus capable of preventing degradation of the sound quality of a decoded signal, while reducing the encoding bit rate. In the apparatus, a spatial information analyzing part ( | 06-11-2009 |
| 20090150163 | METHOD AND APPARATUS FOR MULTICHANNEL UPMIXING AND DOWNMIXING - Loudspeakers in domestic or automotive environments are rarely placed ideally with respect to the sources supplying them, and the stereo and surround images are seldom satisfying. According to the invention there is provided a method and apparatus for combining a precise knowledge about the relative positions of the loudspeakers that were intended (the virtual loudspeakers) and a precise knowledge about the actual placement of listening loudspeakers into a vector space that enables calculation of running corrections to the signals used in order to simulate the presence of the virtual loudspeakers. Specifically the corrections may comprise gain/attenuations determined based on the distances in vector space between the virtual and actual loudspeakers and delays determined from these distances. | 06-11-2009 |
| 20090150164 | TRI-MODEL AUDIO SEGMENTATION - Apparatus, methods, and machine readable media that segment audio streams based upon application of three models to the audio stream are disclosed. One method includes extracting audio features from an audio stream and identifying a set of candidate change points between segments of the audio stream based upon the extracted audio features. The method further includes discarding a candidate change point between a first segment and a second segment in response to determining that a single multivariate Gaussian model represents the extracted audio features of the first segment and the second segment better than a first multivariate Gaussian model represents the extracted audio features of the first segment and a second multivariate Gaussian model represents the extracted audio features of the second segment. | 06-11-2009 |
| 20090150165 | ENCODING AND DETECTING APPARATUS - An encoding data processing apparatus generates a marked version of an audio signal provided on an audio channel. The marked copy is generated by embedding data representative of a payload data word into the audio signal. The encoding data processing apparatus comprises a code word generator operable to generate a water mark code word from the payload data word and to read data representing the water mark code word into a shuffle data store. A shuffle processor is operable to generate pseudo randomly at least one address within an address space of the shuffle data store for each predetermined period and to read data representing part or parts of the water mark code word out from the data store at locations identified by the randomly generated address, and a data embedding processor operable to receive the audio signal and to embed the data representing the part or parts of the water mark code word read out from the shuffle data store into the audio signal for each predetermined period. In one example, the audio signal is one of a plurality of audio signals, each of which is provided on one of a plurality of audio channels of a media item, and the marked copy is generated by embedding the part or parts of the data representative of a payload data word into the audio signal of one or more of the audio channels. As such, even though different parts of the water mark code word may be embedded into the different audio channels, because of the pseudo-random generation of addresses, there is an increased likelihood that if there is a sufficient of the marked audio signal available, then the water mark code word can be recovered from the different audio channels or one of the audio channels. | 06-11-2009 |
| 20090157411 | METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS - An audio encoding method and apparatus and an audio decoding method and apparatus are provided. The audio signal decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating a modified downmix signal based on the downmix signal and extracted information which is extracted from the object-based side information; generating channel-based side information based on the object-based side information and control data for rendering the downmix signal; and generating a multi-channel audio signal based on the modified downmix signal and the channel-based side information. | 06-18-2009 |
| 20090157412 | Method For Streaming Through A Data Service Over A Radio Link Subsystem - An apparatus for controlling a data rate in a data client for a digital audio broadcasting system includes a buffer for storing data, a codec for coding data, and a control module for controlling a bit rate of the codec in response to a level of the data in the buffer. A method performed by the apparatus is also included. | 06-18-2009 |
| 20090164221 | METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS - Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating channel-based side information based on object-based side information and control information for rendering the downmix signal; processing the downmix signal using a decorrelated channel signal; and generating a multi-channel audio signal using the processed downmix signal and the channel-based side information. | 06-25-2009 |
| 20090164222 | METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS - Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method includes extracting a downmix signal and object-based side information from an input audio signal; generating rendering information based on input control data; and generating spatial information based on the rendering information and the object-based side information. | 06-25-2009 |
| 20090164223 | Lossless multi-channel audio codec - A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance. | 06-25-2009 |
| 20090164224 | Lossless multi-channel audio codec - A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance. | 06-25-2009 |
| 20090164225 | METHOD AND APPARATUS OF AUDIO MATRIX ENCODING/DECODING - A method to audio matrix encode/decode, which encode and decode audio signals of two or more channels into an audio signal of one or more channel while preserving the direction of a sound image includes extracting pieces of sound image information from audio signals of multi channels, encoding and allocating the extracted sound image information to an inaudible frequency domain except an audible frequency domain, and adding the sound image information allocated to the inaudible frequency domain and matrix-encoded stereo signals of the audible frequency domain. | 06-25-2009 |
| 20090164226 | Method and Apparatus for Lossless Encoding of a Source Signal Using a Lossy Encoded Data Stream and a Lossless Extension Data Stream - In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding. | 06-25-2009 |
| 20090164227 | Apparatus for Processing Media Signal and Method Thereof - The present invention relates to an apparatus for processing a media signal and method thereof. A method of processing a media signal according to the present invention includes extracting a downmix signal from a bitstream, extracting at least one of first spatial information and second spatial information from the bitstream, and generating multi-channels using the extracted spatial information and the downmix signal. And, the present invention provides a decoding method and apparatus for generating various kinds of multi-channels. | 06-25-2009 |
| 20090171671 | APPARATUS FOR ESTIMATING SOUND QUALITY OF AUDIO CODEC IN MULTI-CHANNEL AND METHOD THEREFOR - There is an apparatus for evaluating the audio quality of a multi-channel audio codec, including: a preprocessing unit for synthesizing binaural signals based on multi-channel audio signals transmitted through a multi-channel of a multi-channel audio reproduction system; an output variable calculator for calculating an interaural cross-correlation coefficient distortion (IACCDist) and other output variables of the binaural signals; and an artificial neural network circuit for outputting a grade of the perceived quality based on the interaural cross-correlation coefficient distortion (IACCDist) and other output variables calculated in the output variable calculator. | 07-02-2009 |
| 20090171672 | Method and Device for the Hierarchical Coding of a Source Audio Signal and Corresponding Decoding Method and Device, Programs and Signals - A method of hierarchically coding a source audio signal in the form of a data stream ( | 07-02-2009 |
| 20090171673 | ENCODING APPARATUS AND ENCODING METHOD - It is possible to provide an encoding device and an encoding method capable of realizing encoding with a very small information amount and a very small calculation amount when encoding higher-band spectrum data according to lower-band spectrum data in a wide-band signal. The device and the method can obtain a high-quality decoded signal even if a large quantization distortion is caused in the lower-band spectrum data. In this device, when encoding higher-band spectrum data in a signal to be encoded, according to lower-band spectrum data in the signal, only for apart (a head portion) of the higher-band spectrum data, the lower-band spectrum data after being quantized is subjected to approximate partial search and higher-band spectrum data is generated according to the search result. | 07-02-2009 |
| 20090171674 | PLAYBACK DEVICE SYSTEMS AND METHODS - A playback device configured to play back audio in response to an instruction. Prior to the instruction, decoded audio data corresponding to a header (e.g., frame number | 07-02-2009 |
| 20090171675 | DECODING REPRODUCTION APPARATUS AND METHOD AND RECEIVER - A decoding reproduction apparatus has an input memory buffer to which audio data asynchronously transmitted is input, a decoding circuit which is configured to read out and decode encoded data stored in the input memory buffer, an output memory buffer which is configured to store an output from the decoding circuit, and an output control circuit which is configured to monitor an amount of use of the input memory buffer, to output the data stored in the output memory buffer by decimation the data when the amount of use becomes larger than a predetermined upper-limit threshold value, and to output the data stored in the output memory buffer by interpolating the data when the amount of use becomes smaller than a predetermined lower-limit threshold value. Occurrence of an overflow or an underflow in the buffer as a result of D/A conversion of the audio data synchronously transmitted is prevented. | 07-02-2009 |
| 20090171676 | METHOD AND AN APPARATUS FOR DECODING AN AUDIO SIGNAL - A method of decoding for an audio signal comprises the step of receiving a downmix of an audio signal, an object information, and a mix information, the object information including an object level information, an object correlation information, and an object gain information, generating a downmix processing information using the object information and the mix information, and processing the downmix of the audio signal using the downmix processing information. Various embodiments of the present invention provide a method and an apparatus for decoding multi-object audio signals fast and efficiently by reducing process time, computer resource, thereby relieving the resource requirement like the wide bandwidth. The object parameters according to the embodiments of the present invention can provide backward compatibility in the view of the channel-oriented decoding process. | 07-02-2009 |
| 20090177478 | Method and Apparatus for Lossless Encoding of a Source Signal, Using a Lossy Encoded Data Steam and a Lossless Extension Data Stream - In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The lossy decoder also provides side information that is used to control the coefficients of a prediction filter that de-correlates the difference signal between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded, providing an extension bit stream. Instead of, or in addition to, de-correlating in the time domain, a de-correlation in the frequency domain using spectral whitening can be performed. The lossy encoded bit stream together with the lossless encoded extension bit stream form a lossless encoded bitstream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/decoding. | 07-09-2009 |
| 20090177479 | Method for Encoding and Decoding Object-Based Audio Signal and Apparatus Thereof - Methods and apparatuses for encoding and decoding an object-based audio signal are provided. The method of decoding an object-based audio signal includes extracting a down-mix signal and object-based parameter information from an input audio signal, generating an object-audio signal using the down-mix signal and the object-based parameter information, and generating an object audio signal with three-dimensional (3D) effects by applying 3D information to the object audio signal. Accordingly, it is possible to localize a sound image for each object audio signal and thus provide a vivid sense of reality during the reproduction of object audio signals. | 07-09-2009 |
| 20090187411 | METHOD FOR ENCODING A SOURCE AUDIO SIGNAL, CORRESPONDING ENCODING DEVICE, DECODING METHOD, SIGNAL, DATA CARRIER AND COMPUTER PROGRAM PRODUCT - A method is provided for coding a source audio signal, involving the transformation of an amplitude/time space into a multi-component amplitude/phase/time space, including the sinusoidal modeling of the audio signal and the delivery of the sinusoidal components that change over time. The method includes the following steps in which: the components are compared to one another in order to define at least one group with at least two components using at least one similarity criterion; and, for at least one group, at least one reference datum is coded, the reference datum being represented by an evolved phase originating from a first component of the group, known as the reference component, and at least one complement datum is coded, the complement datum being associated with at least a second component from the group and, together with the reference datum, enabling the reconstruction of at least one piece of information that is representative of at least one component. | 07-23-2009 |
| 20090192804 | METHOD AND APPARATUS FOR TIME SCALING OF A SIGNAL - A decoder receives ( | 07-30-2009 |
| 20090192805 | METHODS AND APPARATUS FOR PERFORMING VARIABLE BLACK LENGTH WATERMARKING OF MEDIA - Methods, apparatus, and articles of manufacture are disclosed in which auxiliary information is added to or removed from an audio signal. In one example, the information may be added to the audio signal using at least two frequencies that are dictated by two different frequency transformation block sizes, such that the two frequencies are not fully visible when an incorrect block size is used to perform a frequency transformation. Additionally, in another example, a decoder may compensate for time and frequency affects caused by removing old samples and adding new samples, which, in one example, alleviates the need to perform repeated frequency transformation using different frequency transformation block sizes. Other examples are described. | 07-30-2009 |
| 20090192806 | Broadband Frequency Translation for High Frequency Regeneration - An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal. | 07-30-2009 |
| 20090198498 | Method and Apparatus for Estimating High-Band Energy in a Bandwidth Extension System | 08-06-2009 |
| 20090198499 | METHOD AND APPARATUS FOR ENCODING RESIDUAL SIGNALS AND METHOD AND APPARATUS FOR DECODING RESIDUAL SIGNALS - Encoding and decoding of residual signals are provided. In a method of encoding a residual signal of an audio signal, the residual signal is divided into a plurality of sections having different sizes, based on a change of the residual signal. Then, section division information representing information about the divided sections and section-by-section residual signal information representing characteristics of the sections of the residual signal are acquired. Thereafter, the residual signal is encoded based on the section division information and the section-by-section residual signal information. | 08-06-2009 |
| 20090198500 | TEMPORAL MASKING IN AUDIO CODING BASED ON SPECTRAL DYNAMICS IN FREQUENCY SUB-BANDS - An audio coding technique based on modeling spectral dynamics is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. Each sub-band is then frequency transformed and linear prediction is applied. This results in a Hilbert envelope and a Hilbert Carrier for each of the sub-bands. Because of application of linear prediction to frequency components, the technique is called Frequency Domain Linear Prediction (FDLP). The Hilbert envelope and the Hilbert Carrier are analogous to spectral envelope and excitation signals in the Time Domain Linear Prediction (TDLP) techniques. Temporal masking is applied to the FDLP sub-bands to improve the compression efficiency. Specifically, forward masking of the sub-band FDLP carrier signal can be employed to improve compression efficiency of an encoded signal. | 08-06-2009 |
| 20090198501 | METHOD AND APPARATUS FOR ENCODING/DECODING AUDIO SIGNAL USING ADAPTIVE LPC COEFFICIENT INTERPOLATION - Provided are a method and apparatus for encoding or decoding an audio signal by adaptively interpolating a linear predictive coding (LPC) coefficient. In the method and apparatus of encoding or decoding an audio signal, LPC coefficient interpolation is selectively performed depending on whether a transient section is present in a current frame, thereby preventing noise from occurring when interpolating LPC coefficients in the transient section. | 08-06-2009 |
| 20090204412 | Method for Limiting Adaptive Excitation Gain in an Audio Decoder - Decoder for an audio signal coded by a coder including a long-term prediction filter wherein the decoder comprises: a block ( | 08-13-2009 |
| 20090204413 | METHOD AND SYSTEM FOR ASYMMETRIC INDEPENDENT AUDIO RENDERING - Methods and mobile devices are provided for asymmetric independent processing of audio streams in a system on a chip (SOC). More specifically, independent audio paths are provided for processors performing audio processing on the SOC and mixing of decoded audio samples from the processors is performed digitally on the SOC by a hardware digital mixer. | 08-13-2009 |
| 20090210234 | APPARATUS AND METHOD OF ENCODING AND DECODING SIGNALS - A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals. | 08-20-2009 |
| 20090210235 | ENCODING DEVICE, ENCODING METHOD, AND COMPUTER PROGRAM PRODUCT INCLUDING METHODS THEREOF - A disclosed encoding device converts an audio signal into frequency spectra, determines allowable error powers with respect to bands divided by the frequency of the audio signal by a predetermined with, detects a tonal frequency spectrum from the frequency spectra, and detects a band containing the frequency spectrum. Using the detection result and the allowable error powers, the encoding device performs correction such that allowable error powers determined by a power determining unit with respect to bands adjacent to the band detected by a detecting unit become smaller than the powers of the frequency spectra with respect to the adjacent bands, and quantizes each of frequency spectra having greater powers than the corrected allowable error powers. | 08-20-2009 |
| 20090210236 | METHOD AND APPARATUS FOR ENCODING/DECODING STEREO AUDIO - Provided are a method and apparatus for encoding/decoding stereo audio. In the method for encoding stereo audio, stereo audio is encoded based on at least one of the phase difference between first and second channel audios and information on an angle made by a vector on the intensity of mono-audio and a vector on the intensity of the first channel audio or a vector on the intensity of the second channel audio. Thus, the number of encoded parameters is minimized so that a compression ratio in the encoding of the stereo audio is improved. | 08-20-2009 |
| 20090210237 | FRAME COMPENSATION METHOD AND SYSTEM - A frame compensation method is provided. The method includes: obtaining a length of a lost frame and a length of a correct frame; determining that the length of the correct frame is integral power of 2 times of the length of the lost frame, and then obtaining a data sequence with the same length as the length of the lost frame according to the correct frame; and compensating the lost frame according to the data sequence to obtain a compensated data frame. A frame compensation system is also provided. Lost frames in various formats are compensated according to correct frames in various formats, so that the limitation of the related art that a lost frame in a single format can be merely compensated according to a correct frame in a single format is eliminated, and the effect of the compensated data frames is better than that of filling comfort noises. | 08-20-2009 |
| 20090210238 | Methods and Apparatuses for Encoding and Decoding Object-Based Audio Signals - An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal, which is obtained by downmixing a plurality of object signals, and object side information, extracting metadata from the object-side information and displaying an information regarding the object signals based on the metadata. | 08-20-2009 |
| 20090210239 | Method for Encoding and Decoding Object-Based Audio Signal and Apparatus Thereof - The present invention relates to a method and apparatus for encoding and decoding object-based audio signals. This audio decoding method includes extracting a first audio signal and a first audio parameter in which a music object are encoded on a channel basis and a second audio signal and a second audio parameter in which a vocal object are encoded on an object basis, from an audio signal, generating a third audio signal by employing at least one of the first and second audio signals, and generating a multi-channel audio signal by employing at least one of the first and second audio parameters and the third audio signal. Accordingly, the amount of calculation in encoding and decoding processes and the size of a bitstream that is encoded can be reduced efficiently. | 08-20-2009 |
| 20090216541 | Method of Encoding and Decoding an Audio Signal - An apparatus for encoding and decoding an audio signal and method thereof are disclosed, by which compatibility with a player of a general mono or stereo audio signal can be provided in coding an audio signal and by which spatial information for a multi-channel audio signal can be stored or transmitted without a presence of an auxiliary data area. The present invention includes extracting side information embedded in non-recognizable component of audio signal components and decoding the audio signal using the extracted side information. | 08-27-2009 |
| 20090216542 | METHOD AND APPARATUS FOR ENCODING AND DECODING AN AUDIO SIGNAL - A method and apparatus for encoding and decoding an audio signal are provided. The present invention includes receiving an audio signal including a downmix signal and a spatial information signal, if a header is included in the spatial information signal, extracting configuration information from the header, extracting spatial information included in the spatial information signal, and converting the downmix signal to a multi-channel signal using the configuration information and the spatial information. Accordingly, the header can be selectively included in the spatial information signal, thereby if the header is plurally included in the spatial information signal, it is able to decode spatial information in case of reproducing the audio signal from a random point. | 08-27-2009 |
| 20090216543 | METHOD AND APPARATUS FOR ENCODING AND DECODING AN AUDIO SIGNAL - A method and apparatus for encoding and decoding an audio signal are provided. The present invention includes receiving an audio signal including an audio descriptor, recognizing that the audio signal includes a downmix signal and a spatial information signal using the audio descriptor, and converting the downmix signal to a multi-channel signal using the spatial information signal, wherein the spatial information signal includes a header each a preset temporal or spatial interval, and the spatial information signal includes a header each a preset temporal or spatial interval thereby the header can be selectively included in the spatial information signal and if the header is plurally included in the spatial information signal, it is able to decode spatial information in case of reproducing the audio signal from a random point. | 08-27-2009 |
| 20090222272 | Controlling Spatial Audio Coding Parameters as a Function of Auditory Events - An audio encoder or encoding method receives a plurality of input channels and generates one or more audio output channels and one or more parameters describing desired spatial relationships among a plurality of audio channels that may be derived from the one or more audio output channels, by detecting changes in signal characteristics with respect to lime in one or more of the plurality of audio input channels, identifying as auditory event boundaries changes in signal characteristics with respect to lime in the one or more of the plurality of audio input channels, an audio segment between consecutive boundaries constituting an auditory event in the channel or channels, and generating all or some of the one or more parameters al least partly in response to auditory events and/or the degree of change in signal characteristics associated with the auditory event boundaries. An auditory-event-responsive audio upmixer or upmixing method is also disclosed. | 09-03-2009 |
| 20090222273 | Coding/Decoding of a Digital Audio Signal, in Celp Technique - The invention aims at constructing improved dictionaries of CELP excitation vectors for coding/decoding digital audio signals. Usually, each vector of dimension N comprises pulses capable of occupying N valid positions. The invention concerns the construction of dictionaries with particular structure by: providing a common sequence of pulses forming a base pattern; and assigning the base pattern to each excitation vector of the dictionary, based on one or more occurrences at one or more respective positions among said N valid positions. The invention also concerns a combination of dictionaries thus constructed with optionally standard multipulse dictionaries, by union or summation or cascading. | 09-03-2009 |
| 20090240503 | ACOUSTIC SIGNAL PROCESSING APPARATUS AND ACOUSTIC SIGNAL PROCESSING METHOD - To provide an acoustic signal processing apparatus which can reduce the amount of calculation in matrix arithmetic. | 09-24-2009 |
| 20090240504 | Method and Apparatus for Processing an Audio Signal - A method for processing an audio signal, comprising the steps of extracting an ancillary signal for generating the audio signal and an extension signal included in the ancillary signal from a received bit stream, reading length information for the extension signal, skipping decoding of the extension signal or not using a result of the decoding based on the length information, and generating the audio signal using the ancillary signal. Accordingly, in case of processing the audio signal by the present invention, it is able to reduce a corresponding load of operation to enable efficient processing and enhance a sound quality. | 09-24-2009 |
| 20090240505 | AUDIO DECODING - An audio decoder comprises a receiver ( | 09-24-2009 |
| 20090240506 | AUDIO BITSTREAM DATA STRUCTURE ARRANGEMENT OF A LOSSY ENCODED SIGNAL TOGETHER WITH LOSSLESS ENCODED EXTENSION DATA FOR SAID SIGNAL - Lossless compression algorithms can only exploit redundancies of the original audio signal to reduce the data rate, but not irrelevancies as identified by psycho-acoustics. Lossless audio coding schemes apply a filter or transform for decorrelation and then encode the transformed signal. The encoded bit stream comprises the parameters of the transform or filter, and the lossless representation of the transformed signal. However, in case of lossy based lossless coding the additional amount of information exceeds the amount of data for the base layer by a multiple of the base layer data amount. Therefore the additional data cannot be packed completely into the base layer data stream e.g. as ancillary data. The at least two data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal. Furthermore several intermediate quality layers are possible. However, these data streams are not independent from each other Every higher layer depends on the lower layers and can only be reasonably decoded in combination with these lower layers. According to the invention, a special combination of one-time header information with repeated header information in a block structure is used, which kind of combination depends on the type of application. Assignment information data identify the different parts or layers of the lossless format belonging to one input signal. Synchronisation data are used to combine the different data streams or parts or layers to a single lossless or intermediate output signal. These features are used in a file format and in a streaming format. | 09-24-2009 |
| 20090240507 | Method and device for transcoding audio signals - The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MPEG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain into a second parameter domain comprises linearly transforming two or more parameters of the first parameter domain to at least one parameter of the second parameter domain, wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation can be described as a matrix and implemented as a look-up table. | 09-24-2009 |
| 20090240508 | SAMPLING RATE CONVERSION APPARATUS AND METHOD THEREOF - A sampling rate conversion apparatus and a method thereof are provided which increase the sampling rate of a discrete audio signal sampled at a predetermined sampling rate by using a fractal interpolation function (FIF). An audio signal portion formed by a predetermined number of sampling data items is divided into a plurality of interpolation intervals. On the audio signal portion, mapping points are determined. The number of the mapping points is in accordance with the degree of increase in the sampling rate. For the respective interpolation intervals, mapping parameters for performing mapping using the FIF on the mapping points are calculated. In all of the interpolation intervals, the mapping using the FIF is performed on the mapping points with the use of the mapping parameters according to the respective interpolation intervals. Thereby, new sampling data items are generated. | 09-24-2009 |
| 20090240509 | APPARATUS AND METHOD FOR ENCODING AND DECODING USING BANDWIDTH EXTENSION IN PORTABLE TERMINAL - An apparatus and method for encoding and decoding using mutual information between a high band signal and a low band signal to increase a coding efficiency in a portable terminal are provided. The apparatus includes a bandwidth extender for extracting auxiliary information relating to a characteristic of a high band signal using the high band signal and a low band signal and an encoder for encoding residual high band signal obtained by subtracting auxiliary information acquired from the low band signal from auxiliary information acquired from the high band signal. | 09-24-2009 |
| 20090248423 | Apparatus and Method for Encoding/Decoding Signal - An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a three-dimensional (3D) down-mix signal from an input bitstream, generating a down-mix signal with 3D effects removed therefrom by performing a 3D rendering operation on the extracted 3D down-mix signal, and generating a 3D down-mix signal with 3D effects by performing a 3D rendering operation on the generated down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment. | 10-01-2009 |
| 20090254352 | METHOD AND SYSTEM FOR EXTRACTING AUDIO FEATURES FROM AN ENCODED BITSTREAM FOR AUDIO CLASSIFICATION - A method and system for extracting audio features from an encoded bitstream for audio classification. The method comprises partially decoding the encoded bitstream; obtaining uniform window block size spectral coefficients of the encoded bit-stream; and extracting audio features based on the uniform window block spectral coefficients. | 10-08-2009 |
| 20090254353 | METHOD AND SYSTEM FOR AN EFFICIENT IMPLEMENTATION OF THE BLUETOOTH.RTM. SUBBAND CODEC (SBC) - A method for processing audio may include performing using one or more processors and/or circuits in a Bluetooth enabled communication device, receiving audio samples that are encoded by a Bluetooth subband CODEC. The encoded audio samples may include inverse discrete cosine transformed windowed audio data. Shifted subband samples corresponding to the received encoded audio samples may be discrete cosine transformed during decoding by a receive Bluetooth subband codec. The inverse discrete cosine transformed windowed audio data may be generated by an encoding matrix operation. Windowing of the audio data may occur prior to the encoding matrix operation. A plurality of filter coefficients may be utilized for the windowing of the audio data. The shifted subband samples may be discrete cosine transformed during a decoding matrix operation. | 10-08-2009 |
| 20090259476 | DEVICE AND COMPUTER PROGRAM PRODUCT FOR HIGH FREQUENCY SIGNAL INTERPOLATION - There are provided a high frequency signal interpolation method and a high frequency signal interpolation device for forming a preferable high frequency signal with a simple configuration and favorably performing practical high frequency signal interpolation. | 10-15-2009 |
| 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 |
| 20090259478 | Audio Decoding Apparatus and Decoding Method and Program - An energy corrector ( | 10-15-2009 |
| 20090259479 | METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS - The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. | 10-15-2009 |
| 20090265176 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The preset invention includes receiving a downmix signal including at least one object, object information based on attribute of the object, preset information to render the downmix signal and preset attribute information indicating attribute of the preset information; rendering the downmix signal by applying the preset information to all data regions of the downmix signal, if the preset information is included in an extension region of a configuration information region based on the preset attribute information; and rendering the downmix signal by applying the preset information to one corresponding data region of the downmix signal, if the preset information is included in an extension region of a data region based on the preset attribute information. | 10-22-2009 |
| 20090271204 | Audio Compression - For audio encoding and decoding, in order to enhance coded audio signals, the audio signal is divided into at least a low frequency band and a high frequency band, the high frequency band is divided into at least two high frequency sub-band signals, and parameters are generated that refer at least to the low frequency band signal sections which match best with high-frequency sub-band signals. | 10-29-2009 |
| 20090276226 | METHOD AND TERMINAL FOR ENCODING AN ANALOG SIGNAL AND A TERMINAL FOR DECORDING THE ENCODED SIGNAL - An analog signal divided into time frames is encoded and a synthetic signal is formed on the model thereof in a time frame manner via a synthesis filter which is excited by an excitation signal. The excitation signal is formed by at least one adaptive code list containing a plurality of scanning values provided with a defined scanning space. For the actual excitation signal, a segment corresponding to the time frame length is selected from the plurality of scanning values via a speech-based frequency parameter which can take non-integer values and, in such a case, the values intermediate to the scanning values defined by the speech-based frequency parameter are formed in such a way that the time space between the intermediate values and the scanning values is reduced and the totality of the intermediate and the scanning values is used for forming the excitation signal. | 11-05-2009 |
| 20090276227 | FAST SYNTHESIS SUB-BAND FILTERING METHOD FOR DIGITAL SIGNAL DECODING - In order to reproduce audio signals which have been compressed or encoded for storage or transmission using, for example, MPEG audio encoding, a synthesis sub-band filter is employed which performs an inverse modified discrete cosine transform. The computational cost of the IMDCT implementation is reduced by pre-calculating arrays of sum and difference data. The arrays of sum and difference data are then used in two separate transform calculations, the results of which can be used in the generation of pulse code modulation audio data. | 11-05-2009 |
| 20090281811 | TRANSFORM CODER AND TRANSFORM CODING METHOD - A transform coder leading to reduction of degradation of perceptual sound quality even if an adequate number of bits is not assigned. Candidates of a correction scale factor stored in a correction scale factor codebook ( | 11-12-2009 |
| 20090281812 | Apparatus and Method for Encoding and Decoding Signal - Encoding and decoding apparatuses and encoding and decoding methods are provided. The decoding method includes extracting a plurality of encoded signals and division information of the encoded signals from an input bitstream, determining which of a plurality of decoding methods is to be used to decode each of the encoded signals, decoding the encoded signals using the determined decoding methods, and synthesizing the decoded signals with reference to the division information. Accordingly, it is possible to encode signals having different characteristics at an optimum bitrate by classifying the signals into one or more classes according to the characteristics of the signals and encoding each of the signals using an encoding unit that can best serve the class where a corresponding signal belongs. In addition, it is possible to efficiently encode various signals including audio and speech signals. | 11-12-2009 |
| 20090281813 | NOISE SYNTHESIS | 11-12-2009 |
| 20090281814 | METHOD AND AN APPARATUS FOR DECODING AN AUDIO SIGNAL - A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a downmix processing information using the object information and the mix information; processing the downmix signal using the downmix processing information; and, generating a multi-channel information using the object information and the mix information, wherein the number of channel of the downmix signal is equal to the number of channel of the processed downmix signal is disclosed. | 11-12-2009 |
| 20090281815 | COMPENSATION TECHNIQUE FOR AUDIO DECODER STATE DIVERGENCE - A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size. | 11-12-2009 |
| 20090287492 | Method for the dynamic range compression of an audio signal and corresponding hearing device - It should be possible to carry out dynamic range compression in hearing devices, and in particular in hearing aids, so it is free from distortion and practically in real time. For this purpose it is proposed that the modulation spectrum be obtained from the audio or input signal. The modulation spectrum is subsequently directly modified corresponding to a predefined compression function. Finally a modified or compressed output signal is recovered from the modified modulation spectrum. Alternatively a complex envelope may be obtained from the input signal, which is filtered using time-variant modulation filtering corresponding to a predefined compression rule. A distortion-free, compressed output signal may also be recovered herefrom. | 11-19-2009 |
| 20090287493 | DIRECT STREAM DIGITAL AUDIO WITH MINIMAL STORAGE REQUIREMENT - An audio coding scheme allowing PCM signal to lossless DSD signal expansion for next generation optical disc formats. The method of encoding an input DSD signal includes up-sampling a corresponding PCM signal to the DSD sample rate. Then a set of loop filter parameters for a noise-shaping loop of a sigma-delta modulator are generated, either using a random starting condition of the sigma-delta modulator or including synchronization parameters. This will allow a decoder to regenerate an almost perfect signal, but still it needs a correction signal to be able to bit identically regenerate the DSD input signal. Therefore, a correction signal is generated based on a difference between a sigma-delta modulated version of the up-sampled PCM signal and the input DSD signal, wherein the sigma-delta modulated version of the up-sampled PCM signal is obtained using the set of loop filter parameters. The correction signal may be adapted to be applied to the low bit PCM signal, to the up-sampled PCM signal or to the output bit stream. Finally, an expansion bit stream is generated where an encoded version of the set of loop filter parameters and an encoded version of the correction signal are included. The decoder can reproduce the original DSD signal based on the already available PCM signal and the described expansion bit stream. Thus, the coding scheme enables top quality audio with minimal storage overhead since the already available PCM signal is used in combination with an expansion bit stream. Since only an additional data stream is required to be stored on a disc, e.g. as part of an MPEG stream, DSD functionality is added to existing systems without causing compatibility problems. | 11-19-2009 |
| 20090287494 | Apparatus for Processing Media Signal and Method Thereof - The present invention relates to a method of processing a media signal and apparatus therefor. A media signal decoding method according to the present invention includes detecting a channel having a valid value of the multi-channels to be generated and generating the detected channel having the valid value from the downmix signal and the spatial information signal. Accordingly, the present invention is able to reduce a decoding operation quantity by detecting which one of the channels to be generated from a transferred media signal is set to a virtual value and omitting decoding for the generation of the channel set to the virtual value. | 11-19-2009 |
| 20090287495 | SPATIAL AUDIO - In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end. | 11-19-2009 |
| 20090287496 | LOUDNESS ENHANCEMENT SYSTEM AND METHOD - A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs. | 11-19-2009 |
| 20090292543 | Transmission Bandwidth And Memory Requirements Reduction In A Portable Image Capture Device - A system and method for reducing storage and transmission bandwidth requirements of a portable image capture device capable of establishing a communications connection on a network are disclosed. The method includes assigning an image identifier to captured images uploaded to a server on a network, wherein each of the images is stored in an image file having a particular size. The size of each of the image files corresponding to the uploaded images on the image capture device may then be reduced, thereby reducing storage requirements. In response to a user request to apply an action to one of the uploaded images, only the image identifier of the image and the requested action are transmitted to the server, thereby eliminating the need to retransmit the image and reducing transmission bandwidth requirements. | 11-26-2009 |
| 20090299753 | Audio Signal Transient Detection - Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment. | 12-03-2009 |
| 20090299754 | FACTORIZATION OF OVERLAPPING TRANFORMS INTO TWO BLOCK TRANSFORMS - An audio encoder/decoder uses a combination of an overlap windowing transform and block transform that have reversible implementations to provide a reversible, integer-integer form of a lapped transform. The reversible lapped transform permits both lossy and lossless transform domain coding of an audio signal having variable subframe sizes. | 12-03-2009 |
| 20090299755 | Method for Post-Processing a Signal in an Audio Decoder - A method of post-processing in an audio decoder a signal reconstructed by time and frequency shaping ( | 12-03-2009 |
| 20090299756 | Ratio of speech to non-speech audio such as for elderly or hearing-impaired listeners - A hybrid stereophonic/monophonic audio signal encoding comprises generating, in response to a discrete two-channel stereophonic audio signal, an encoded hybrid stereophonic/monophonic audio signal in which the audio signal is a discrete two-channel audio signal below a frequency f | 12-03-2009 |
| 20090299757 | METHOD AND APPARATUS FOR ENCODING AND DECODING - An method for encoding comprising: obtaining, according to a data length of a first overlapped portion between encoding data of a current frame and encoding data of a previous frame, first encoding data corresponding to the data length of the first overlapped portion from the previous frame, if the previous frame is encoded in a first encoding mode and the current frame is to be encoded in a second encoding mode; and encoding, in the second encoding mode, the first encoding data corresponding to the data length of the first overlapped portion from the previous frame and encoding data of the current frame. The corresponding decoding method, encoding and decoding apparatuses are also disclosed. | 12-03-2009 |
| 20090306992 | Method for switching rate and bandwidth scalable audio decoding rate - A method of bitrate switching on decoding an audio signal coded by a audio coding system, said decoding comprising a post-processing step depending on the bitrate. On switching from an initial bitrate to a final bitrate, said method includes a transition step of continuous change from a signal at the initial bitrate to a signal at the final bitrate, one or both of said signals being post-processed. Application to transmission of VoIP speech and/or audio signals in data packet networks. | 12-10-2009 |
| 20090306993 | METHOD AND APPARATUS FOR LOSSLESS ENCODING OF A SOURCE SIGNAL, USING A LOSSY ENCODED DATA STREAM AND A LOSSLESS EXTENSION DATA STREAM - The invention is related to lossless encoding of a source signal, using a lossy encoded data stream and a lossless extension data stream which together form a lossless encoded data stream for said source signal, whereby lossless audio compression means audio coding with bit-exact reproduction of the original PCM samples at decoder output. The lossy encoding/decoding may be an mp3 coding/decoding. The invention uses an integer MDCT and frequency domain de-correlation and time domain de-correlation for the residual signal of the base-layer lossy audio codec. The exploitation of side information from the lossy base-layer codec allows for reduction of redundancies in the gross bit stream, thus improving the coding efficiency of the lossy based lossless codec. | 12-10-2009 |
| 20090306994 | METHOD AND AN APPARATUS FOR IDENTIFYING FRAME TYPE - A method for identifying a frame type is disclosed. The present invention includes receiving current frame type information, obtaining previously received previous frame type information, generating frame identification information of a current frame using the current frame type information and the previous frame type information, and identifying the current frame using the frame identification information. | 12-10-2009 |
| 20090313027 | HIGH-QUALITY ENCODING AT LOW-BIT RATES - Methods and devices provide improved perceived quality of an audio (or other) coded signal at a low bit-rate. An input signal may be split into an outlier portion and a stationary portion. The outlier portion of the input signal may be encoded. The stationary portion may be divided into subvectors. Each subvector may be classified as trivial or non-trivial. Each trivial subvector may be encoded using a pre-defined pattern. Each non-trivial subvector may be encoded with at least one location of at least one significant sample and a sign of the significant sample. | 12-17-2009 |
| 20090313028 | METHOD, APPARATUS AND COMPUTER PROGRAM PRODUCT FOR PROVIDING IMPROVED AUDIO PROCESSING - An apparatus for performing improved audio processing may include a processor. The processor may be configured to divide respective signals of each channel of a multi-channel audio input signal into one or more spectral bands corresponding to respective analysis frames, select a leading channel from among channels of the multi-channel audio input signal for at least one spectral band, determine a time shift value for at least one spectral band of at least one channel, and time align the channels based at least in part on the time shift value. | 12-17-2009 |
| 20090313029 | Method And System For Backward Compatible Multi Channel Audio Encoding and Decoding with the Maximum Entropy - A method and system for backward compatible multi-channel audio encoding and decoding in sense of the space information maximum entropy is disclosed. The technical solution according to the invention can adopt any existing stereo channel encoding system to encode the multi-channels audio signals, so as to transmit the multi-channel audio signals at the low bit rate as that of the stereo audio signals. More importantly, the existing stereo channel reproducing systems can also decode the audio format that is encoded utilizing the encoding method according to the invention. | 12-17-2009 |
| 20090319277 | Source Coding and/or Decoding - A method of bandwidth expansion in which a low band signal is used to create an excitation signal for an LPC synthesis filter for producing a high band synthetic signal. An encoding process comprises: dividing a signal into a low band signal and a high band signal; coding the low band signal; analysing the high band audio signal to create filter coefficients; filtering the high band signal, using a filter configured by the created filter coefficients, to produce a residual signal; creating a measure of the residual signal; and outputting the coded low band signal, the created filter coefficients for the high band signal and the measure. A decoding process is similar to the reverse of the encoding process. | 12-24-2009 |
| 20090319278 | EFFICIENT CODING OF OVERCOMPLETE REPRESENTATIONS OF AUDIO USING THE MODULATED COMPLEX LAPPED TRANSFORM (MCLT) - An “Overcomplete Audio Coder” provides various techniques for overcomplete encoding audio signals using an MCLT-based predictive coder. Specifically, the Overcomplete Audio Coder uses unrestricted polar quantization of MCLT magnitude and phase coefficients. Further, quantized magnitude and phase coefficients are predicted based on properties of the audio signal and corresponding MCLT coefficients to reduce the bit rate overhead in encoding the audio signal. This prediction allows the Overcomplete Audio Coder to provide improved continuity of the magnitude of spectral components across encoded signal blocks, thereby reducing warbling artifacts. Coding rates achieved using these prediction techniques are comparable to that of encoding an orthogonal representation of an audio signal, such as with modulated lapped transform (MLT)-based coders. Finally, the Overcomplete Audio Coder provides a true magnitude-phase frequency-domain representation of the audio signal, thus allowing precise auditory models to be applied for improving compression performance, without the need for additional Fourier transforms. | 12-24-2009 |
| 20090319279 | METHOD AND SYSTEM FOR AUDIO TRANSMIT LOOPBACK PROCESSING IN AN AUDIO CODEC - Methods and systems for audio transmit loopback processing in an audio CODEC are disclosed and may include receiving digital audio signals to be transmitted via the wireless device, and looping back one or more of the signals to an output device via a switch matrix. One or more of the digital audio signals may be generated via a digital microphone, which may include a microelectromechanical (MEMS) microphone. The received digital audio signals may be filtered via decimation filters, which may include poly-phase filters. The received digital audio signals may be switched between phases of the poly-phase filters via an input switch, which may include CMOS transistors. One or more of the received digital audio signals may be generated from a received analog signal via an analog to digital converter (ADC), which may include a multi-channel ADC. The output device may include a wireless headset, loudspeaker and/or audio test equipment. | 12-24-2009 |
| 20090319280 | 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. | 12-24-2009 |
| 20090326958 | Methods and Apparatuses for Encoding and Decoding Object-Based Audio Signals - An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving first and second audio signals, which are object-encoded; generating third object energy information based on first object energy information included in the first audio signal and second object energy information included in the second audio signal; and generating a third audio signal by combining the first and second object signals and the third object energy information. | 12-31-2009 |
| 20090326959 | GENERATION OF DECORRELATED SIGNALS - In a case of transient audio input signals, in a multi-channel audio reconstruction, uncorrelated output signals are generated from an audio input signal in that the audio input signal is mixed with a representation of the audio input signal delayed by a delay time such that, in a first time interval, a first output signal corresponds to the audio input signal, and a second output signal corresponds to the delayed representation of the audio input signal, wherein, in a second time interval, the first output signal corresponds to the delayed representation of the audio input signal, and the second output signal corresponds to the audio input signal. | 12-31-2009 |
| 20090326960 | ENCODING AND DECODING OF AUDIO OBJECTS - An audio system comprises an encoder ( | 12-31-2009 |
| 20090326961 | EFFICIENT AND SECURE FORENSIC MARKING IN COMPRESSED DOMAIN - Methods, devices, and computer program products enable the embedding of forensic marks in a host content that is in compressed domain. These and other features are achieved by preprocessing of a host content to provide a plurality of host content versions with different embedded watermarks that are subsequently compressed. A host content may then be efficiently marked with forensic marks in response to a request for such content. The marking process is conducted in compressed domain, thus reducing the computational burden of decompressing and re-compressing the content, and avoiding further perceptual degradation of the host content. In addition, methods, devices and computer program products are disclosed that obstruct differential analysis of such forensically marked content. | 12-31-2009 |
| 20090326962 | QUALITY IMPROVEMENT TECHNIQUES IN AN AUDIO ENCODER - An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits. | 12-31-2009 |
| 20100004936 | AUDIO OUTPUT APPARATUS CAPABLE OF SUPPRESSING POP NOISE - An audio output apparatus includes an audio codec receiving a digital audio signal from a system controller and outputting an analog audio signal corresponding to the digital audio signal to an amplifier through a capacitor, and a switch controller triggered by a trigger signal to output a control signal to the amplifier. The amplifier is disabled to cease outputting the analog audio signal in response to the control signal. The trigger signal is generated by one of the system controller, the audio codec, and a power circuit supplying electric power to the system controller, the audio codec, the switch controller and the amplifier upon occurrence of a condition associated with pop noise, and is outputted to the switch controller before the pop noise is generated, such that the amplifier is disabled in response to the control signal from the switch controller and is unable to output the pop noise. | 01-07-2010 |
| 20100010818 | Method and an Apparatus for Decoding an Audio Signal - A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a downmix processing information using the object information and the mix information; processing the downmix signal using the downmix processing information; and, generating a multi-channel information using the object information and the mix information, wherein the number of channel of the downmix signal is equal to the number of channel of the processed downmix signal is disclosed. | 01-14-2010 |
| 20100010819 | Method and an Apparatus for Decoding an Audio Signal - A method for processing an audio signal, comprising: receiving a downmix signal in time domain; if the downmix signal corresponds to a mono signal, bypassing the downmix signal; if the number of channel of the downmix signal corresponds to at least two, decomposing the downmix signal into a subband signal, and processing the subband signal using a downmix processing information, wherein the downmix processing information is estimated based on an object information and a mix information is disclosed. | 01-14-2010 |
| 20100010820 | Method and an Apparatus for Decoding an Audio Signal - A method for processing an audio signal, comprising: receiving a downmix signal and a downmix processing information; and, processing the downmix signal using a downmix processing information, comprising: de-correlating the downmix signal; and, mixing the downmix signal and the de-correlated signal in order to output the processed downmix signal, wherein the downmix processing information is estimated based on an object information and a mix information is disclosed. | 01-14-2010 |
| 20100010821 | Method and an Apparatus for Decoding an Audio Signal - A method for processing an audio signal, comprising: receiving a downmix signal, a first multi-channel information, and an object information; processing the downmix signal using the object information and a mix information; and, transmitting one of the first multi-channel information and a second multi-channel information according to the mix information, wherein the second channel information is generated using the object information and the mix information is disclosed. | 01-14-2010 |
| 20100017213 | DEVICE AND METHOD FOR POSTPROCESSING SPECTRAL VALUES AND ENCODER AND DECODER FOR AUDIO SIGNALS - For postprocessing spectral values which are based on a first transformation algorithm for converting the audio signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided. Hereupon, a weighted addition of spectral values of the sequence of blocks of spectral values is performed in order to obtain a sequence of blocks of postprocessed spectral values, wherein the combination is performed such that for calculating a postprocessed spectral value for a frequency band and a time duration a spectral value of the sequence of blocks for the frequency band and the time duration and a spectral value for another frequency band or another time duration are used, wherein the combination is further performed such that such weighting factors are used that the postprocessed spectral values are an approximation to the spectral values as they are obtained by converting the audio signal into a spectral representation using a second transformation algorithm which is different from the first transformation algorithm. The postprocessed spectral values are in particular used for a difference formation within a scalable encoder or for an addition within a scalable decoder, respectively. | 01-21-2010 |
| 20100023333 | High frequency signal interpolating method and high frequency signal interpolating - A quality high frequency signal is generated through simple processing, and practical high frequency signal interpolation is carried out. A digital audio signal reproduced by an apparatus, which carries out compression, is provided to an input terminal | 01-28-2010 |
| 20100023334 | AUDIO CODING APPARATUS, AUDIO CODING METHOD AND RECORDING MEDIUM - An audio encoding apparatus includes: a sub-band division part dividing a quantized value into sub-bands; an integrated codeword length table including a plurality of codeword length tables storing codeword lengths of individual code books and a plurality of codeword length tables; a code book selection part selecting, from the plurality of code books, the given combination of code books; a codeword length table selection part selecting a codeword length table; an index value calculation part sequentially calculating an index value to be used by the code book; a codeword length calculation part collectively obtaining codeword lengths for each code book; and a coding part determining a code book of a codeword length of a minimum accumulated result and encoding the quantized values in the sub-band based on the determined code book. | 01-28-2010 |
| 20100023335 | LOW COMPLEXITY PARAMETRIC STEREO DECODER - A stereo audio decoder with low complexity is provided. A high stereo sound quality can be obtained with a limited computational power and is thus suitable for miniature and mobile equipment. The stereo decoder generates a set of stereo output channels (C | 01-28-2010 |
| 20100030563 | APPARATUS AND METHOD FOR GENERATING AN AMBIENT SIGNAL FROM AN AUDIO SIGNAL, APPARATUS AND METHOD FOR DERIVING A MULTI-CHANNEL AUDIO SIGNAL FROM AN AUDIO SIGNAL AND COMPUTER PROGRAM - An apparatus for generating an ambient signal from an audio signal includes a compressor for lossy compression of a representation of the audio signal so as to obtain a compressed representation of the audio signal describing a compressed audio signal. The apparatus for generating the ambient signal further includes a calculator for calculating a difference between the compressed representation of the audio signal and the representation of the audio signal so as to obtain a discrimination representation. The apparatus further includes a provider for providing the ambient signal using the discrimination representation. An apparatus for deriving a multi-channel audio signal from an audio signal includes an apparatus for generating an ambient signal from an audio signal, an apparatus for providing the audio signal as a front-loudspeaker signal and an apparatus for providing the ambient signal as a back-loudspeaker signal. | 02-04-2010 |
| 20100042415 | AUDIO SIGNAL CODING METHOD AND DECODING METHOD - It possible not only to reduce a delay, but also to enhance the coding efficiency and reduce audio artifact upon coding. | 02-18-2010 |
| 20100042416 | CODING/DECODING METHOD, SYSTEM AND APPARATUS - An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal. | 02-18-2010 |
| 20100042417 | MULTIPLEXING APPARATUS, MULTIPLEXING METHOD, PROGRAM, AND RECORDING MEDIUM - A multiplexing apparatus for multiplexing audio data into transport stream (TS) packets includes a first encoding section encoding the audio data by a first encoding method; a second encoding section encoding the audio data by a second encoding method, which is a variable-length encoding method and which differs from the first encoding method, for attaching a timing value indicating a timing used when audio data is decoded in units of predetermined audio data; a packetization section packetizing the audio data encoded by the first encoding section and the audio data encoded by the second encoding section into TS packets and for attaching the same ID to a plurality of packetized TS packets; a determination section determining a TS packet to be multiplexed from among the plurality of TS packets packetized by the packetization section; and a multiplexing section multiplexing the TS packet determined by the determination section. | 02-18-2010 |
| 20100049530 | UNIVERSAL CONTAINER FOR AUDIO DATA - Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent. | 02-25-2010 |
| 20100049531 | UNIVERSAL CONTAINER FOR AUDIO DATA - Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent. | 02-25-2010 |
| 20100049532 | 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. | 02-25-2010 |
| 20100057471 | METHOD AND SYSTEM FOR PROCESSING AUDIO SIGNALS VIA SEPARATE INPUT AND OUTPUT PROCESSING PATHS - Aspects of a method and system for processing audio signals via separate input and output processing paths are provided. In this regard, a hardware audio CODEC comprising one or more audio inputs and one or more audio outputs and may be enabled to route, via one or more switching elements, audio signals from any of the inputs to any of the outputs. The CODEC may be enabled to simultaneously process a plurality of audio signals based on a configuration of the switching elements. Upstream from the switching elements, received audio signals may be processed independent of an output to which the may be communicated. Downstream from said switching elements audio signals may be processed independent of an input via which the signals were received. | 03-04-2010 |
| 20100057472 | METHOD AND SYSTEM FOR FREQUENCY COMPENSATION IN AN AUDIO CODEC - In a method and system for frequency compensation in an audio CODEC, a filter in a hardware audio CODEC may be configured based on power consumption and based on a frequency response of an active output device to which the filter is communicatively coupled. The filter may comprise a plurality of filter stages, which may be, for example, biquads, and the filter may be configured by enabling or disabling one or more of the stages. In this manner, power consumption of the filter may be managed by enabling and/or disabling one or more stages. Configuration of the filter may be performed dynamically depending on whether one or more audio output devices may be active. In this regard, which output device is active and its frequency response may be determined and filter coefficients may be reconfigured upon a change in which output device may be active. | 03-04-2010 |
| 20100057473 | METHOD AND SYSTEM FOR DUAL VOICE PATH PROCESSING IN AN AUDIO CODEC - Aspects of a method and system for dual voice path processing in an audio CODEC may enable selecting two or more signals received via one or more audio input devices, and filtering and down-sampling each of the selected signals via two or more signal processing branches. Furthermore, an output sample rate of each of the signal processing branches may be configured independently. The signal processing branches may each comprise one or more IIR filters with configurable coefficients and one or more cascaded-integrate-comb (CIC) decimation filters having a configurable decimation ratio. A first of the selected signals, may be filtered and/or down-sampled to generate a signal having a first, lower, sample rate, and a second of the signals may be filtered and/or down-sampled to generate a signal having a second, higher, sample rate. One or more post-processing algorithms such as audio beamforming may also be performed on the selected signals. | 03-04-2010 |
| 20100057474 | METHOD AND SYSTEM FOR DIGITAL GAIN PROCESSING IN A HARDWARE AUDIO CODEC FOR AUDIO TRANSMISSION - In a hardware audio CODEC which processes audio signals from a plurality of inputs, voltage and/or power levels of the input audio signals may be adjusted such that the digitally adjusted levels are approximately equal for each of the plurality of inputs. The digital adjustment may comprise, for each audio sample of one of the input audio signals, adding the audio sample to one or more right shifted versions of the audio sample and selecting a portion of a summed audio signal resulting from the addition. The portion of the summed audio that is selected may be determined based on the type of audio content being processed. The one or more right shifted versions of the audio sample may be selected via one or more switching elements which may be controlled via a digital control word which may be dynamically generated. | 03-04-2010 |
| 20100057475 | METHOD AND SYSTEM FOR DIGITAL GAIN CONTROL IN AN AUDIO CODEC - Aspects of a method and system for digital gain control in an audio CODEC are provided. In this regard, in a hardware audio CODEC, a plurality of gain values in decibel format may be generated and may be summed to generate an overall gain value in decibels. Gain values may be ramped over a plurality of audio samples. The overall gain value in decibels may then be utilized to scale one or more audio signals. A first portion of the overall gain value in decibels may be converted to a scalar and a remaining portion of the overall gain value in decibels may be converted to a shift count. Accordingly, the scaling may comprise multiplying one or more audio signals by the scalar to generate one or more scaled audio signals and shifting the scaled audio signals by a number of bits equal to the shift count. | 03-04-2010 |
| 20100057476 | SIGNAL BANDWIDTH EXTENSION APPARATUS - A signal bandwidth extension apparatus includes a determination unit which determines whether or not a peak component of the input signal is lacked in the band to be extended, and a control unit which controls to extend the bandwidth when the determination unit determines that the peak component of the input signal is lacked in the band to be extended, and not to extend the bandwidth when the determination unit determines that the peak component is not lacked. | 03-04-2010 |
| 20100063824 | APPARATUS AND METHOD FOR WIDENING AUDIO SIGNAL BAND - An audio signal band expanding apparatus ( | 03-11-2010 |
| 20100063825 | Systems and Methods for Memory Management and Crossfading in an Electronic Device - Systems and methods are disclosed for the management of memory used in a crossfading operation in an electronic device. In one embodiment, a processor is used to alternately decode two audio streams, one which is being faded out and one which is being faded in to implement a crossfade. The two audio streams may be encoded in the same or different formats and may be alternately decoded such that resource usage is reduced. The amount of decoded data of both audio streams and other parameters may determine which audio stream is to be actively decoded. In certain embodiments, the decoded data may be stored in a circular buffer, and a delta is determined between the decoded data and the empty space of the buffer. | 03-11-2010 |
| 20100063826 | COMPUTATION APPARATUS AND METHOD, QUANTIZATION APPARATUS AND METHOD, AUDIO ENCODING APPARATUS AND METHOD, AND PROGRAM - A computation apparatus includes: a range calculation section for calculating a range of an input value that can give a predetermined discrete value obtained by discretizing a computation result of a nonlinear operation; and a discrete value output section for outputting, when the input value is input, the predetermined discrete value corresponding to the range in which the input value that has been input is contained. | 03-11-2010 |
| 20100063827 | Selective Bandwidth Extension - A method of receiving an audio signal includes measuring a periodicity of the audio signal to determine a checked periodicity. At least one best available subband is determined. At least one extended subband is composed, wherein composing includes reducing a ratio of composed harmonic components to composed noise components if the checked periodicity is lower than a threshold, and scaling a magnitude of the at least one extended subband based on a spectral envelope on the audio signal. | 03-11-2010 |
| 20100070284 | METHOD AND AN APPARATUS FOR PROCESSING A SIGNAL - An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. | 03-18-2010 |
| 20100070285 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - The present invention includes receiving a plurality of frame data including first frame data and second frame data encoded by at least one coding schemes, obtaining first flag information indicating whether the first frame data and the second frame data are encoded by frequency domain transform coding scheme, respectively, decoding the first frame data by frequency domain transform coding scheme based on the first flag information when the first frame data is encoded by frequency domain transform coding scheme, obtaining second flag information indicating whether subframe data is encoded by time domain transform coding scheme or time-frequency domain coding scheme when the second frame data is not encoded by frequency domain transform coding scheme, the at least two subframe data being included in the second frame data, decoding the subframe data by time domain transform coding scheme or time-frequency domain transform coding scheme based on the second flag information, and compensating for discontinuity existing between the first frame data decoded by frequency domain transform coding scheme and the subframe data decoded by time domain transform coding scheme, wherein the time-frequency domain coding scheme is time domain coding scheme including frequency domain transform. | 03-18-2010 |
| 20100070286 | Technique for controlling codec selection along a complex call path - The invention relates to a technique of operating a call control node controlling at least one section of a call path. The call path includes between two opposite edge nodes a multi-section harmonization path along which codec selection is to be harmonized. A method embodiment of the technique, wherein the call control node is a transfer node in the harmonization path between the edge nodes, comprises the steps of determining if the call control node is a transfer node of the harmonization path; determining if a codec used for the at least one section controlled by the call control node fulfils a predefined harmonization criterion; and providing, in case the used codec does not fulfill the harmonization criterion, a harmonization trigger indication to at least one of the edge nodes of the harmonization path for initiating harmonization. | 03-18-2010 |
| 20100070287 | ADAPTING MASKING THRESHOLDS FOR ENCODING A LOW FREQUENCY TRANSIENT SIGNAL IN AUDIO DATA - An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal. | 03-18-2010 |
| 20100076772 | Methods and Apparatuses for Encoding and Decoding Object-Based Audio Signals - An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal and object-based side information, the downmix signal comprising at least two downmix channel signals; extracting gain information from the object-based side information and generating modification information for modifying the downmix channel signals on a channel-by-channel basis based on the gain information; and modifying the downmix channel signals by applying the modification information to the downmix channel signals. | 03-25-2010 |
| 20100076773 | SECURE AUDIO STREAM SCRAMBLE SYSTEM - A process for distributing digital audio sequences according to a nominal flux format including a succession of fields, each of which includes at least one digital block clusterizing a selected number of coefficients corresponding to single audio elements that are digitally coded inside the flux and utilized by audio decoders that are able to play it to be able to decode it correctly, including a preparatory step including modifying at least one of the coefficients, and a transmission step including a primary flux in compliance with a nominal format including blocks that were modified during the preparatory step and by a route separated from the primary flux by an additional piece of digital information which allows reconstruction of the original flux starting with a calculation, on recipient equipment, as a function of the primary flux and of the additional information. | 03-25-2010 |
| 20100082352 | 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. | 04-01-2010 |
| 20100088102 | AUDIO CODING AND REPRODUCING APPARATUS - In order to reduce the problem that sound cuts out due to overflow of audio output data caused by a delay in shifting to an audio reproducing process, an audio coding and reproducing apparatus includes: an input data storage unit in which PCM audio signals are stored; an output data storage unit in which data to be outputted is stored; an audio output unit configured to output the audio data; an audio coding unit configured to code the audio data; a coded data storage unit configured to store the audio data coded by the audio coding unit; a bitrate control unit configured to control a bitrate at which the coded data is outputted, based on an amount of free space of the output data storage unit; and a data memory unit configured to retain the coded data. | 04-08-2010 |
| 20100088103 | PLAYBACK APPARATUS AND PLAYBACK METHOD - Noise is prevented when decoding an audio stream not containing syncwords or CRC bits in the elementary stream. When decoding a current frame, the private header of the next frame is analyzed and the current frame is muted if the private header of the next frame is not valid. When there is a data discontinuity caused by editing, decoding resumes from the start address of the next frame determined. | 04-08-2010 |
| 20100094637 | ARBITRARY SHAPING OF TEMPORAL NOISE ENVELOPE WITHOUT SIDE-INFORMATION - In a first aspect, arbitrary shaping of the temporal envelope of noise is provided in spectral domain coding systems without the need of side-information. In the encoding, a filtered measure of quantization error is applied as a feedback signal to the frequency-domain representation of a discrete time-domain signal prior to quantization, so that the filtering parameters of said filtering affect the shaping of quantization noise in the time domain of the quantized frequency-domain representation of the discrete time-domain signal when it is inversely transformed from the frequency domain back to the time domain in decoding. This may be accomplished with respect to each of a plurality of frequency bins or groups of bins. In another aspect, frequency-domain noise-feedback quantizing in digital audio encoding is provided. | 04-15-2010 |
| 20100094638 | APPARATUS AND METHOD FOR DECIDING ADAPTIVE NOISE LEVEL FOR BANDWIDTH EXTENSION - An apparatus and method for deciding an adaptive noise level for bandwidth extension are provided. The apparatus includes a noise level decider for deciding a high-band noise level for bandwidth extension according to tonality of an input signal, a pitch frequency analyzer for detecting a pitch frequency of the input signal and analyzing correlation between the detected pitch frequency and a frequency channel, and a noise level controller for adaptively controlling the decided high-band noise level based on the analyzed correlation of the pitch frequency and the frequency channel. | 04-15-2010 |
| 20100094639 | METHODS AND ARRANGEMENTS EMPLOYING DIGITAL CONTENT ITEMS - Methods and arrangements for identifying content, and employing such identification, are detailed. One method embeds a plural-bit digital watermark into content, but first checks to see if the content is previously watermarked. Another method applies a digital watermark detection procedure to only a sub-portion of a digital content item. Yet another arrangement involves plural-portion content, where one portion is watermarked with first data governing its rights management, and another portion is watermarked with second data governing its rights management. Still another method concerns distribution of content items, where each is watermarked with a unique ID as part of the distribution process. Yet another method concerns deriving an identifier from content, and using the content to access related metadata from a remote computer system. Still other methods concern arrangements for recognizing content, and then providing links to information about the content creator, etc., in response. A variety of other technologies and improvements are also detailed. | 04-15-2010 |
| 20100094640 | AUDIO ENCODING METHOD AND DEVICE - Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that can be applied to the entire broadened signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter allowing the reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal. makes it possible to obtain a signal of good perceived quality. | 04-15-2010 |
| 20100094641 | AUDIO SIGNAL PROCESSING CIRCUIT - An audio signal processing circuit is provided which comprises an ADC which samples an audio signal at a predetermined sampling frequency, a high-band compensation processor which compensates a signal sampled by the ADC to a frequency band which is higher than a signal band sampled by the sampling frequency, and an encoding unit which encodes a signal processed by the high-band compensation processor. | 04-15-2010 |
| 20100094642 | METHOD OF LOST FRAME CONSEALMENT AND DEVICE - A device for lost frame concealment comprises: a lost frame detector for detecting whether a voice frame is lost, a decoding module for decoding the current voice frame, a low band delay module for delaying the low band signal, a low band signal recovering module for recovering the lost low band signal, a high band lost frame concealment module for processing the lost frame concealment for the high band signal, and a QMF synthesis filter for synthetically filtering the low band signal and the high band signal. The invention makes full use of the delay of the coding/decoding device itself, enhances the effect of lost frame concealment for the low band signal and the high band signal, and introduces no nearby delay during the process of lost frame concealment. | 04-15-2010 |
| 20100100389 | System for Signal Sample Rate Conversion - An apparatus and method for converting a source signal at a first rate to a re-sampled signal at a second rate using an array of processors. A decoder decomposes the source signal into left and right source values and sends an aperture signal to a coefficient control unit upon decomposition completion. A transfer unit controllably receives and passes the left and right source values on to a re-sampler. The coefficient control unit calculates a polyphase offset based on the aperture signal and a clock signal. A coefficient server selectively passes coefficients to the re-sampler based on the polyphase offset. And the re-sampler generates the re-sampled signal based on the left and right source values and the coefficients. | 04-22-2010 |
| 20100106508 | SYSTEM FOR PROVIDING AUDIO RECORDINGS - Disclosed systems provide audio recordings from digital television provider networks. Audio recordings are streamed to a customer premises equipment device of a user and, in response to user input, an identifier for the audio recording is transmitted to a music provider. A copy of the audio recording is delivered to a user designated device. | 04-29-2010 |
| 20100106509 | AUDIO ENCODING METHOD, AUDIO DECODING METHOD, AUDIO ENCODING DEVICE, AUDIO DECODING DEVICE, PROGRAM, AND AUDIO ENCODING/DECODING SYSTEM - An audio encoding device ( | 04-29-2010 |
| 20100106510 | METHODS AND APPARATUS TO PERFORM AUDIO WATERMARKING AND WATERMARK DETECTION AND EXTRACTION - Methods and apparatus to audio watermarking and watermark detection and extracted are described herein. An example method includes receiving a media content signal, sampling the media content signal to generate samples, storing the samples in a buffer, determining a first sequence of samples in the buffer, determining a second sequence of samples in the buffer, wherein the second sequence of samples is of substantially equal length as the first sequence of samples, calculating an average of the first sequence of samples and the second sequence of samples to generate an average sequence of samples, extracting an identifier from the average sequence of samples, and storing the identifier in a tangible memory. | 04-29-2010 |
| 20100106511 | Encoding apparatus and encoding method - An encoding apparatus converts an input signal into a frequency-domain spectrum signal, divides the converted spectrum signal into an arbitrary number of segments with respect to a time axis and a frequency axis, calculates a spectrum power of each segment and a feature parameter that represents a feature of the corresponding spectrum power, calculates a masking threshold using the calculated spectrum power of each segment, detects a segment having a spectrum power equal to or less than the calculated masking threshold, corrects the spectrum power of the detected segment, and encodes both the spectrum power of the corrected segment and the calculated parameter. | 04-29-2010 |
| 20100114581 | Method for encoding, method for decoding, encoder, decoder and computer program products - A method for encoding a plurality of signal values is described, wherein the signal values are grouped into a first subgroup and a second subgroup, the signal values of the first subgroup are compared to the signal values of the second subgroup and based on the result of the comparison it is decided whether the signal values of the first subgroup are bit-plane encoded with higher priority than the signal values of the second subgroup. | 05-06-2010 |
| 20100114582 | APPARATUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL INCLUDING INFORMATION BITSTREAM CONVERSION - Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals. | 05-06-2010 |
| 20100114583 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a spectral data of lower band and type information indicating a particular band extension scheme for a current frame of the audio signal from among a plurality of band extension schemes including a first band extension scheme and a second band extension scheme, by an audio processing apparatus; when the type information indicates the first band extension scheme for the current frame, generating a spectral data of higher band in the current frame using the spectral data of lower band by performing the first band extension scheme; and when the type information indicates the second band extension scheme for the current frame, generating the spectral data of higher band in the current frame using the spectral data of lower band by performing the second band extension scheme, wherein the first band extension scheme is based on a first data area of the spectral data of lower band, and wherein the second band extension scheme is based on a second data area of the spectral data of lower band. | 05-06-2010 |
| 20100114584 | AUDIO DEVICE AND AUDIO PROCESSING METHOD - An audio device and an audio processing method are provided. The audio device includes a digital microphone module and an audio codec. The digital microphone module captures an external audio source according to a spread-spectrum clock, in order to generate a digital audio source signal. The audio codec includes a clock generation module, a storage unit, and an audio codec core. The clock generation module generates a clock signal and the spread-spectrum clock. The storage unit temporarily stores a first digital audio source signal from the digital microphone module according to the spread-spectrum clock, and outputs the first digital audio source signal according to the clock signal. The audio codec core comprises a digital-to-analog (D/A) converter and an analog-to-digital (A/D) converter. | 05-06-2010 |
| 20100114585 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - An apparatus for processing an audio signal and method thereof are disclosed, by which a method for processing an audio signal, comprising: extracting noise filling flag information indicating whether noise filling is used to a plurality of frames; extracting coding scheme information indicating whether a current frame included in the plurality of frames is coded in either a frequency domain or a time domain; when the noise filling flag information indicates that the noise filling is used to for the plurality of frames and the coding scheme information indicates that the current frame is coded in the frequency domain, extracting noise level information for the current frame; when a noise level value corresponding to the noise level information meets a predetermined level, extracting noise offset information for the current frame; and, when the noise offset information is extracted, performs the noise-filling for the current frame based on the noise level value and the noise offset information. | 05-06-2010 |
| 20100121646 | CODING/DECODING OF DIGITAL AUDIO SIGNALS - The invention relates to the coding/decoding of a signal into several sub-bands, in which at least a first and a second sub-bands which are adjacent are transform coded ( | 05-13-2010 |
| 20100121647 | APPARATUS AND METHOD FOR CODING AND DECODING MULTI OBJECT AUDIO SIGNAL WITH MULTI CHANNEL - Provided are an apparatus and method for coding and decoding a multi object audio signal with multi channel. The apparatus includes: a multi channel encoding means for down-mixing an audio signal including a plurality of channels, generating a spatial cue for the audio signal including the plurality of channels, and generating first rendering information including the generated spatial cue; and a multi object encoding unit for down-mixing an audio signal including a plurality of objects, which includes the down-mixed signal from the multi channel encoding unit, generating a spatial cue for the audio signal including the plurality of objects, and generating second rendering information including the generated spatial cue, wherein the multichannel encoding unit generates a spatial cue for the audio signal including the plurality of objects regardless of a Coder-DECoder (CODEC) scheme the limits the multi channel encoding unit. | 05-13-2010 |
| 20100121648 | AUDIO FREQUENCY ENCODING AND DECODING METHOD AND DEVICE - An audio encoding method and a corresponding decoding method are provided according to the present invention. Accordingly, the pre-echo effect of the audio transient signal is eliminated and the distortion of the transient signal is mitigated. The technical solution includes performing time-domain processing on an input audio transient signal; dividing sampling points x | 05-13-2010 |
| 20100145711 | METHOD AND AN APPARATUS FOR DECODING AN AUDIO SIGNAL - A method of processing an audio signal is disclosed. The present invention includes receiving downmix information, object information and mix information, generating and transferring multi-channel information using at least one of the downmix information, the object information and the mix information, and selectively generating and transferring either first gain information or extra multi-channel information including second gain information in accordance with a decoding mode using at least one of the object information and the mix information. | 06-10-2010 |
| 20100145712 | CODING OF DIGITAL AUDIO SIGNALS - The invention concerns an encoder for an input audio signal (S(z)) comprising a combination module combining the input audio signal with an intermediate counter-reaction signal forming a modified input signal and a quantification module scalable for the rate ( | 06-10-2010 |
| 20100145713 | ADDING ADDITIONAL DATA TO ENCODED BIT STREAMS - A method of adding additional data to encoded bit streams may include receiving a signal containing an encoded data frame, where the encoded data frame includes a plurality of data blocks. The method may further include transforming the encoded data frame into a modified encoded data frame by inserting at least one additional data block between a synchronization information block and an error check block, where the at least one additional data block includes the additional data, and modifying data in at least one of the synchronization information block and the error check block to account for the inserting of the at least one additional data block. | 06-10-2010 |
| 20100145714 | METHODS AND APPARATUSES FOR BIT STREAM DECODING IN MP3 DECODER - A decoding method for MP3 bit streams, which replaces a buffer required in the decoding process by manipulating the order of data decoding. The decoding method includes reading the head and side information of the current frame, and calculating a main data's start address of the current frame. While decoding the main data, the head and side information of subsequent frames are skipped if the reading of the main data is not yet completed. The start address of the next frame is calculated and directly accessed after finished reading the main data of the current frame. An optimum method for accessing frequency lines utilizes the characteristics of the MP3 frequency line, instead of inserting a plurality of zeros in the rzero zone containing successive zeros, the initial boundary address of the rzero zone is memorized. | 06-10-2010 |
| 20100153118 | AUDIO ENCODING AND DECODING - A multi-channel audio encoder ( | 06-17-2010 |
| 20100153119 | APPARATUS AND METHOD FOR CODING AUDIO DATA BASED ON INPUT SIGNAL DISTRIBUTION CHARACTERISTICS OF EACH CHANNEL - Provided is an audio coding apparatus and method that can selectively apply a operation mode of a coding module for stereo or multi-channel representation according to input signal characteristics of each channel, when voice or music signals are transmitted using an audio codec in portable terminals capable of stereo or multi-channel input and output. The audio coding apparatus includes a down-mixer for down-mixing multi-channel audio signals into mono signals; a coder for coding the mono signals; an input channel correlation analyzer for deciding whether to give them stereo effect based on their signal distribution characteristics, and outputting a control signal indicating whether to perform stereo representation process; and a stereo representation unit for performing stereo representation process onto the multi-channel audio signals when the control signal indicating to perform stereo representation process. | 06-17-2010 |
| 20100153120 | AUDIO DECODING APPARATUS AUDIO DECODING METHOD, AND RECORDING MEDIUM - An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal. | 06-17-2010 |
| 20100153121 | INFORMATION CODING APPARATUS - An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal. The feedback signal generator is configured by a pole-zero filter that includes a filter coefficient of an all-pole filter which is based on spectral envelope information estimated by the input audio signal, a parameter for adjusting a peak level in the frequency characteristic of the quantization noise caused by the all-pole filter, and the predictive filter coefficient. | 06-17-2010 |
| 20100161340 | Dynamic Codec Switching - In one embodiment, a method and apparatus for processing an audio signal are provided. In one example of the invention, an audio signal is received. The audio signal is analyzed to determine an interference level of the audio signal relative to a threshold interference level. Then the audio signal is processed with a lower quality codec or a higher quality codec responsive to the determination of the interference level of the audio signal relative to the threshold interference level. | 06-24-2010 |
| 20100161341 | DISCRIMINATOR FOR DISCRIMINATING EMPLOYED MODULATION TECHNIQUE, SIGNAL DEMODULATOR, MUSICAL INSTRUMENT AND METHOD OF DISCRIMINATION - A signal modulator includes a discriminator for discriminating a modulation technique through which a carrier signal was modulated to a quasi audio signal and a signal demodulation module for reproducing a continuous data stream from the quasi audio signal through a demodulating technique corresponding to the discriminated modulation technique; the discriminator includes a sampling circuit for extracting groups of samples from the quasi audio signal during each period of the carrier signal, an integrator calculating an integrated value on each group of samples, a comparator comparing the integrated value with a threshold for a neighborhood of zero so as to determine the groups of samples with the integrated value less than the threshold and a determiner measuring the time period between the groups of two modulation period and discriminating 16DPSK when the time period is equal to the modulation period. | 06-24-2010 |
| 20100161342 | CODING APPARATUS AND METHOD, DECODING APPARATUS AND METHOD, AND PROGRAM STORAGE MEDIUM - In order to obtain coded data which does not strike viewers and listeners as being incongruous, when plural audio data are to be coded, a coding program groups the respective audio data into one audio data, codes the grouped audio data in sequence with a predetermined number of samples being treated as units, and sets delimitations corresponding to the delimitations of the plural audio data in the coded data at coding units of the coded data. | 06-24-2010 |
| 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 |
| 20100169102 | LOW COMPLEXITY MPEG ENCODING FOR SURROUND SOUND RECORDINGS - The invention provides for the encoding of surround sound produced by any coincident microphone techniques with coincident-to-virtual microphone signal matrixing. An encoding scheme provides significantly lower computational demand, by deriving the spatial parameters and output downmixes from the coincident microphone array signals and the coincident-to-surround channel-coefficients matrix, instead of the multi-channel signals. | 07-01-2010 |
| 20100169103 | METHOD AND APPARATUS FOR ENHANCEMENT OF AUDIO RECONSTRUCTION - An audio signal having at least one audio channel and associated direction parameters indicating a direction of origin of a portion of the audio channel with respect to a recording position is reconstructed to derive a reconstructed audio signal. A desired direction of origin with respect to the recording position is selected. The portion of the audio channel is modified for deriving a reconstructed portion of the reconstructed audio signal, wherein the modifying includes increasing an intensity of the portion of the audio channel having direction parameters indicating a direction of origin close to the desired direction of origin with respect to another portion of the audio channel having direction parameters indicating a direction of origin further away from the desired direction of origin. | 07-01-2010 |
| 20100169104 | Partially Complex Modulated Filter Bank - An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal. | 07-01-2010 |
| 20100174547 | Speech coding - A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal; and from the speech signal, deriving a spectral envelope signal representing the modelled filter and a remaining signal representing the modelled source. At intervals during the encoding, the method further comprises determining a period between portions of the remaining signal having a degree of repetition and determining a correlation between said portions based on that period, thus producing a respective vector of the correlation for each interval. Once every number of said intervals, the method further comprises selecting a codebook from a plurality of codebooks for quantizing the vectors, quantizing the vectors of that number of intervals according to the selected codebook, and transmitting the quantized vectors along with an indication of the selected codebook. | 07-08-2010 |
| 20100179814 | PARTIALLY COMPLEX MODULATED FILTER BANK - An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal. | 07-15-2010 |
| 20100185450 | METHOD AND SYSTEM FOR EFFICIENT OPTIMIZATION OF AUDIO SAMPLING RATE CONVERSION - A controller for outputting audio having a predetermined output sample rate. The controller is programmed to receive an audio signal having a first sample rate, identify the first sample rate of the audio signal, select a converter based on the identification of the first sample rate, and convert the sample rate of the received audio signal to the predetermined output sample rate. | 07-22-2010 |
| 20100191536 | AUDIO CODING SELECTION BASED ON DEVICE OPERATING CONDITION - A sensor is configured to determine at least one operating condition of a device and a selector is configured to select an audio coding process for the device, based on the operating condition. The operating condition may include remaining battery life of the device and/or ambient noise level. The selected audio coding process may consume less power than another possible audio coding process during audio processing. The audio may include voice and/or audio playback, e.g., music playback. | 07-29-2010 |
| 20100191537 | BINAURAL OBJECT-ORIENTED AUDIO DECODER - A binaural object-oriented audio decoder comprising decoding means for decoding and rendering at least one audio object based on head-related transfer function parameters is proposed. Said decoding means are being arranged for positioning an audio object in a virtual three-dimensional space. Said head-related transfer function parameters are being based on an elevation parameter, an azimuth parameter, and a distance parameter. Said parameters are corresponding to the position of the audio object in the virtual three-dimensional space. The binaural object-oriented audio decoder is configured for receiving the head-related transfer function parameters, whereby said received head-related transfer function parameters are varying for the elevation parameter and the azimuth parameter only. Said binaural object-oriented audio decoder is characterized by distance processing means for modifying the received head-related transfer function parameters according to a received desired distance parameter. Said modified head-related transfer function parameters are being used to position the audio object in the three-dimensions at the desired distance. Said modification of the head-related transfer function parameters is based on a predetermined distance parameter for said received head-related function parameters. | 07-29-2010 |
| 20100191538 | HIERARCHICAL CODING OF DIGITAL AUDIO SIGNALS - The invention relates to a method for scalar quantization-based coding of the samples of a digital audio signal (S), the samples being coded over a pre-determined number of bits in order to obtain a binary frame of quantization indices (I | 07-29-2010 |
| 20100198601 | AUDIO ENCODING AND DECODING METHOD AND ASSOCIATED AUDIO ENCODER, AUDIO DECODER AND COMPUTER PROGRAMS - The invention relates to a method for ordering spectral parameters of ambisonic components to be encoded (A | 08-05-2010 |
| 20100198602 | METHOD AND AN APPARATUS FOR DECODING AN AUDIO SIGNAL - The present invention relates to an apparatus for processing an audio signal and method thereof. The present invention includes receiving a downmix signal comprising plural objects, and a bitstream including object information and downmix gain information, obtaining level guide flag information for all frames indicating whether level guide information is present in the bitstream, obtaining the level guide information representing a limitation of object level applied to at least one object of the plural objects, from the bitstream, based on the level guide flag information, receiving mix information, generating modified mix information by modifying the mix information based on the level guide information and the downmix gain information, and generating at least one of downmix processing information and multi-channel information based on the modified mix information and the object information, wherein the mix information is estimated using object level for at least one object of the plural objects, and wherein the object information and the downmix gain information are determined when the downmix signal is generated. | 08-05-2010 |
| 20100198603 | SUB-BAND PROCESSING COMPLEXITY REDUCTION - A sub-band processing system that reduces computational complexity and memory requirements includes a processor and a local or distributed memory. Logic stored in the memory partitions a frequency spectrum of bins into a smaller number of sub-bands. The logic enables a lossy compression by designating a magnitude and a designated or derived phase of each bin in the frequency spectrum as representative. The logic renders a lossless compression by decompressing the lossy compressed data and providing lost data based on original spectral relationships contained within the frequency spectrum. | 08-05-2010 |
| 20100204995 | Compander System - A compressor device for a compander system has a level detecting/control device and a pre-emphasis device for carrying out an adaptive pre-emphasis filtering. The invention is also directed to an expander device for a compander system with a level detecting/control device and a de-emphasis device for carrying out an adaptive de-emphasis filtering. | 08-12-2010 |
| 20100204996 | METHOD AND SYSTEM FOR DYNAMIC RANGE CONTROL IN AN AUDIO PROCESSING SYSTEM - Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC. | 08-12-2010 |
| 20100204997 | ADAPTIVE TUNING OF THE PERCEPTUAL MODEL - Methods of encoding a signal using a perceptual model are described in which a signal to mask ratio parameter within the perceptual model is tuned. The signal to mask ratio parameter is tuned based on a function of the bitrate of the part of the signal which has already been encoded and the target bitrate for the encoding process. The tuned signal to mask ratio parameter is used to compute a masking threshold for the signal which is then used to quantise the signal. | 08-12-2010 |
| 20100204998 | Time Warped Modified Transform Coding of Audio Signals - A representation of an audio signal having a first frame, a second frame following the first frame, and a third frame following the second frame, is derived by estimating first warp information for the first and the second frame and second warp information for the second frame and the third frame, the warp information describing a pitch information of the audio signal. First spectral coefficients for the first and the second frame are derived using the first warp information and a first weighted representation of the first and the second frame, the first weighted representation derived by applying a first window function to the first and the second frames, wherein the first window function depends on the first warp information. Second spectral coefficients for the second and the third frame are derived using the second warp information and a second weighted representation of the second and the third frame, the second weighted representation derived by applying a second window function to the second and the third frames, wherein the second window function depends on the second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients. | 08-12-2010 |
| 20100211398 | VECTOR QUANTIZER, VECTOR INVERSE QUANTIZER, AND THE METHODS - A vector quantizer which improves the accuracy of vector quantization in switching over a vector quantization codebook on a first stage depending on the type of feature having the correlation with a quantization target vector. In the vector quantizer, a classifier ( | 08-19-2010 |
| 20100211399 | Spectral Translation/Folding in the Subband Domain - The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications. | 08-19-2010 |
| 20100217605 | METHODS AND DEVICES FOR PERFORMING A FAST MODIFIED DISCRETE COSINE TRANSFORM OF AN INPUT SEQUENCE - An improved fast N-point MDCT process and encoder/decoder is disclosed. The N-point MDCT may be realized through an N/2-point DCT algorithm. The N/2 DCT transform matrix is directly factored and the factored DCT transform matrices are used to develop a set of equations for realizing the N-point MDCT coefficients from an input sequence. The factoring of the DCT transform matrix may include expressing the DCT transform as a multiplication of matrices and exploiting mirror images within the matrices. It may further include simplifying at least one of the matrices by eliminating a variable based on trigonometric identity | 08-26-2010 |
| 20100217606 | SIGNAL BANDWIDTH EXPANDING APPARATUS - A signal bandwidth expanding apparatus configured to expand a bandwidth of an input signal. The apparatus includes: a time acquiring section configured to acquire time information; a priority holding section configured to hold priority information of processes, each process divided from a process of bandwidth expansion; a controller configured to: sequentially perform the processes from a process having a higher priority using the priority information held by the priority holding section, calculate a time taken for the process using the time acquiring section when the process is ended, and control whether or not a next process having a secondary priority is performed according to the time taken for the process; and a frequency balance correcting section configured to change a frequency characteristic of a signal expanded in a bandwidth according to the process performed by the controller. | 08-26-2010 |
| 20100217607 | Audio Decoder, Audio Encoder, Methods for Decoding and Encoding an Audio Signal and Computer Program - An audio decoder for providing a decoded representation of an audio content on the basis of an encoded representation of the audio content comprises a linear-prediction-domain decoder core configured to provide a time-domain representation of an audio frame on the basis of a set of linear-prediction domain parameters associated with the audio frame and a frequency-domain decoder core configured to provide a time-domain representation of an audio frame on the basis of a set of frequency-domain parameters, taking into account a transform window out of a set comprising a plurality of different transform windows. The audio decoder comprises a signal combiner configured to overlap-and-add-time-domain representations of subsequent audio frames encoded in different domains, in order to smoothen a transition between the time-domain representations of the subsequent frames. The set of transform windows comprises one or more windows specifically adapted for a transition between a frequency-domain core mode and a linear-prediction-domain core mode. | 08-26-2010 |
| 20100217608 | SOUND DECODER AND SOUND DECODING METHOD WITH DEMULTIPLEXING ORDER DETERMINATION - A sound encoder multiplexes a plurality of codes into a sound code in an order determined by a multiplexing order determination unit ( | 08-26-2010 |
| 20100217609 | CODING APPARATUS, DECODING APPARATUS, CODING METHOD, AND DECODING METHOD - A down-sampler | 08-26-2010 |
| 20100223061 | Method and Apparatus for Audio Coding - In accordance with an example embodiment of the present invention, there is provided an apparatus for encoding an audio signal in two or more encoding stages, the audio signal comprising a set of frequency components. The apparatus comprises a frequency component selection unit configured to select a number of frequency components from the set for encoding in a current encoding stage, the selected frequency components being components of the set that have not been encoded to a non-zero value in a preceding encoding stage; and an encoding unit configured to encode at least one of the selected frequency components to a non-zero value using a number of bits less than or equal to a predetermined number of bits allocated for the current encoding stage. | 09-02-2010 |
| 20100223062 | METHODS AND APPARATUS TO PERFORM AUDIO WATERMARKING AND WATERMARK DETECTION AND EXTRACTION - Methods and apparatus to audio watermarking and watermark detection and extracted are described herein. According to an example method, an identifier is encoded in media content when a different identifier has been previously encoded. According to another example method, messages decoded from media content are validated to provide improved decoding accuracy. In another example method, decoded symbols are stored in memory and synchronization symbols are located to detect a message encoded in media content. | 09-02-2010 |
| 20100228551 | Encoding/Decoding of Digital Signals, Especially in Vector Quantization with Permutation Codes - The invention relates to the encoding/decoding of digital signals, especially using transposition codes involving a calculation of combinatorial expressions. According to the invention, the combinatorial expressions are represented by prime factor power decompositions, and determined by a preliminary reading of pre-recorded representations of decompositions of selected whole numbers. | 09-09-2010 |
| 20100228552 | Audio decoding apparatus and audio decoding method - An audio decoding apparatus and method are provided. The audio decoding apparatus includes a spectrum converting part configured to divide the first frequency spectrum in each channel of the first audio signal in a time direction or in a frequency direction to calculate a first signal sequence having the same time resolution and the same frequency resolution in all the channels of the first audio signal, a down-mixing part configured to perform weighted addition on the signals at the same time and within the same frequency band included in the first signal sequence in all the channels to calculate a second signal sequence having channels of a second number different from the first number of channels. | 09-09-2010 |
| 20100228553 | COMMUNICATION TERMINAL DEVICE, COMMUNICATION SYSTEM, AND COMMUNICATION METHOD - Provided is a communication terminal device which performs audio encoding by using downloaded audio codec while using an existing radio communication system infrastructure, a channel codec, and an error correction/detection function as they are. In the communication terminal device, if a judgment unit ( | 09-09-2010 |
| 20100228554 | MULTI-OBJECT AUDIO ENCODING AND DECODING METHOD AND APPARATUS THEREOF - Provided are a multi-object audio encoding and decoding method and an apparatus thereof. The multi-object encoding method includes generating a down-mix signal and a residual signal by down-mixing a foreground audio object and a background audio object, and generating a bitstream including the down-mix signal and the residual signal. | 09-09-2010 |
| 20100228555 | Partially Complex Modulated Filter Bank - An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal. | 09-09-2010 |
| 20100228556 | Quantization for Audio Encoding - Disclosed herein is a quantization method and apparatus of an audio encoder. The quantization method comprises calculating an absolute value of a maximum frequency spectrum of a first frame, externally received, by analyzing frequency spectrum data of the first frame, setting an initial value of a common scale factor to be used to quantize the first frame based on the absolute value of the maximum frequency spectrum of the first frame and an absolute value of a maximum frequency spectrum of a second frame, which has previously been calculated, and quantizing the frequency spectrum data of the first frame based on the set initial value of the common scale factor. Accordingly, before quantization is performed, an initial value of a common scale factor which is almost close to a value of an actual common scale factor can be previously set. | 09-09-2010 |
| 20100228557 | Method and apparatus for audio decoding - A method for decoding an audio signal includes: obtaining a lower-band signal component of an audio signal corresponding to a received code stream when the audio signal switches from a first bandwidth to a second bandwidth which is narrower than the first bandwidth; extending the lower-band signal component to obtain higher-band information; performing a time-varying fadeout process on the higher-band information to obtain a processed higher-band signal component; and synthesizing the processed higher-band signal component and the obtained lower-band signal component. With the methods provided in the embodiments of the invention, when an audio signal has a switch from broadband to narrowband, a series of processes such as bandwidth detection, artificial band extension, time-varying fadeout process, and bandwidth synthesis, may be performed to make the switch to have a smooth transition from a broadband signal to a narrowband signal so that a comfortable listening experience may be achieved. | 09-09-2010 |
| 20100235171 | AUDIO DECODER - Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder ( | 09-16-2010 |
| 20100235172 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - A method for processing an audio signal, comprising: receiving the audio signal; and processing the received audio signal, wherein the audio signal is processed according to a scheme comprising: comparing a size information of at least two blocks of A+1 level with a size information of a block of A level corresponding to the at least two of A+1 level; and, determining the at least two blocks of A+1 level as an optimum block if the size information of the at least two blocks of A+1 level is less than the size information of the block of A level is disclosed. A method for processing an audio signal, comprising: receiving the audio signal; and processing the received audio signal, wherein the audio signal is processed according to a scheme comprising: comparing a size information of a block of A level with a size information of at least two blocks of A+1 level; and, determining the block of A level as an optimum block if the size information of the block of A level is less than the size information of the at least two blocks of A+1 level is disclosed. | 09-16-2010 |
| 20100235173 | FIXED CODEBOOK SEARCH METHOD AND SEARCHER - A fixed codebook search method includes: initializing a counter; searching for pulses and calculating the value of a cost function Qk; initializing the counter if the Qk value increases; increasing the value of the counter if the Qk value does not increase; judging whether the value of the counter is greater than the threshold value; continuing the search process if the value of the counter is not greater than the threshold value; and ending the whole search process if the value of the counter is greater than the threshold value. The present invention reduces the search count and improves the search efficiency. | 09-16-2010 |
| 20100235174 | DIGITAL AUDIO SIGNAL COMPRESSION METHOD AND APPARATUS - Compression of audio signal data is described herein. In various embodiments, the compression of each unit of the audio signal data includes the employment of a distribution substantially representative of a subblock of residual data of the unit of audio signal data, to reduce the amount of data having to be transmitted to transmit the unit of audio signal data to a recipient. | 09-16-2010 |
| 20100241433 | AUDIO ENCODER, AUDIO DECODER AND AUDIO PROCESSOR HAVING A DYNAMICALLY VARIABLE WARPING CHARACTERISTIC - An audio encoder, an audio decoder or an audio processor includes a filter ( | 09-23-2010 |
| 20100241434 | MULTI-CHANNEL DECODING DEVICE, MULTI-CHANNEL DECODING METHOD, PROGRAM, AND SEMICONDUCTOR INTEGRATED CIRCUIT - The multi-channel decoding device ( | 09-23-2010 |
| 20100241435 | Apparatus for efficiently mixing narrowband and wideband voice data and a method therefor - A voice mixing apparatus decodes input encoded narrowband voice data and encoded voice data for narrowband region of input encoded wideband voice data, and detects a speaker in accordance with the decoded voice signals of the entire narrowband. When encoded voice data from a speaker is included in the narrowband, a signal in a region outside the narrowband of the expanded data is encoded. When the data is included in the wideband, encoded voice data of the region outside the narrowband is extracted for output. When the destination terminal is compatible with the encoded narrowband voice data, the narrowband voice signal mixed is encoded and output. When the destination terminal is compatible with wideband, the narrowband voice signal mixed is encoded for the narrowband region, and the voice data of the speaker is used as the encoded voice data for the region outside the narrowband. | 09-23-2010 |
| 20100241436 | Apparatus and method for encoding and decoding multi-channel signal - Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal. | 09-23-2010 |
| 20100241437 | METHOD AND DEVICE FOR NOISE FILLING - A method for perceptual spectral decoding comprises decoding of spectral coefficients recovered from a binary flux into decoded spectral coefficients of an initial set of spectral coefficients. The initial set of spectral coefficients are spectrum filled. The spectrum filling comprises noise filling of spectral holes by setting spectral coefficients in the initial set of spectral coefficients not being decoded from the binary flux equal to elements derived from the decoded spectral coefficients. The set of reconstructed spectral coefficients of a frequency domain formed by the spectrum filling is converted into an audio signal of a time domain. A perceptual spectral decoder comprises a noise filler, operating according to the method for perceptual spectral decoding. | 09-23-2010 |
| 20100241438 | METHOD AND AN APPARATUS OF DECODING AN AUDIO SIGNAL - A method of decoding an audio signal is disclosed, The present invention includes the steps of receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal, extracting the ambient component signal and the source component signal of each of the channels based on correlation between the channel signals, modifying the ambient component signal using surround effect information, and generating the audio signal including a plurality of channels using the modified ambient component signal and the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal. And, a signal output unit for outputting a ambient component signal is arranged to have an output direction different from that of another signal output unit for outputting a source component signal, whereby a listener can be provided with an audio signal of which ambient sound is enhanced. | 09-23-2010 |
| 20100241439 | METHOD, MODULE AND COMPUTER SOFTWARE WITH QUANTIFICATION BASED ON GERZON VECTORS - The invention relates to a method for encoding the components (X | 09-23-2010 |
| 20100250258 | Method for Correcting Metadata Affecting the Playback Loudness of Audio Information - A coded signal conveys encoded audio information and metadata that may be used to control the loudness of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values. | 09-30-2010 |
| 20100250259 | METHOD AND AN APPARATUS OF DECODING AN AUDIO SIGNAL - The present invention includes an audio signal receiving unit receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal; an ambient component signal extracting unit extracting the ambient component signal of each of the channels based on correlation between the channel signals; an ambient component signal modifying unit modifying the ambient component signal using surround effect information; a source component signal extracting unit extracting the source component signal of each of the channels based on the correlation between the channel signals; a first signal output unit outputting the modified ambient component signal and the source component signal; and a second signal output unit outputting the audio signal or the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal. And, a signal output unit for outputting an ambient component signal is arranged to have an output direction different from that of another signal output unit for outputting a source component signal, whereby a listener can be provided with an audio signal of which ambient sound is enhanced. | 09-30-2010 |
| 20100250260 | Encoder - A method including generating from a first audio signal, and via a first encoding and decoding of the first audio signal, a second audio signal; determining at least one energy difference value between the first audio signal and the second audio signal; and calculating at least one signal shaping factor dependent on the at least one energy difference value. | 09-30-2010 |
| 20100250261 | Encoder - An apparatus including at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to select at least two single frequency components; generate an indicator, the indicator being configured to represent the at least two single frequency components and is configured to be dependent on the frequency separation between the two single frequency components. | 09-30-2010 |
| 20100250262 | METHOD AND APPARATUS FOR CODING OR DECODING WIDEBAND SPEECH - A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result. | 09-30-2010 |
| 20100250263 | METHOD AND APPARATUS FOR CODING OR DECODING WIDEBAND SPEECH - A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result. | 09-30-2010 |
| 20100250264 | SPECTRAL ENHANCING METHOD AND DEVICE - The invention concerns a method for spectral enhancement and a device therefor. The inventive method is a method for enhancing spectral content of a signal having an incomplete spectrum including a first spectral band, the method including the following steps: at least transposing the spectral content of the first band into a second spectral band not included in the spectrum to generate a transposed spectrum signal, with spectrum limited to the second spectral band; transforming the spectrum of the transposed spectrum signal to obtain an enhancing signal; combining the incomplete spectrum signal and the enhancing signal to produce a spectrum enhanced signal. The invention is characterised in that the spectral content is subjected to a whitening step. | 09-30-2010 |
| 20100262427 | LOW COMPLEXITY SPECTRAL BAND REPLICATION (SBR) FILTERBANKS - A complex analysis filterbank is implemented by obtaining an input audio signal as a plurality of N time-domain input samples. Pair-wise additions and subtractions of the time-domain input samples is performed to obtain a first and second groups of intermediate samples, each group having N/2 intermediate samples. The signs of odd-indexed intermediate samples in the second group are then inverted. A first transform is applied to the first group of intermediate samples to obtain a first group of output coefficients in the frequency domain. A second transform is applied to the second group of intermediate samples to obtain an intermediate second group of output coefficients in the frequency domain. The order of coefficients in the intermediate second group of output coefficients is then reversed to obtain a second group of output coefficients. The first and second groups of output coefficients may be stored and/or transmitted as a frequency domain representation of the audio signal. | 10-14-2010 |
| 20100268540 | SYSTEM AND METHOD FOR UTILIZING AUDIO BEACONING IN AUDIENCE MEASUREMENT - An audio beacon system, apparatus and method for collecting information on a panelist's exposure to media. An audio beacon is configured as on-device encoding technology that is operative in a panelist's processing device (e.g., cell phone, PDA, PC) to enable the device to encode and/or process media data and acoustically transmit it for a predetermined period of time. The acoustically transmitted data is received and processed by a portable audience measurement device, such as Arbitron's Personal People Meter™ (“PPM”), or other specially equipped portable device to enable audience measurement systems to achieve higher levels of detail on panel member activity and greater association of measurement devices to their respective panelists. | 10-21-2010 |
| 20100268541 | AUDIO PROCESSING METHOD, SYSTEM, AND CONTROL SERVER - An audio processing method includes: after the terminal accesses the control server, the control server obtains audio capabilities of the terminal through capability negotiation; and the control server forwards the coded audio data to each terminal according to the audio capabilities. An audio processing system and a control server are disclosed. In the embodiments of the present disclosure, the audio data does not need to undergo an operation of audio coding and decoding every time when the audio data passes through a control server, and the number of coding and decoding operations performed by the control server are reduced drastically. Especially, in the case that only one control server exists, the audio delay between terminals only derives from network transmission, coding of the sending terminal and decoding of the receiving terminal, and the control server extracts and reassembles the packets of the audio data only. | 10-21-2010 |
| 20100274564 | COORDINATED ANR REFERENCE SOUND COMPRESSION - Apparatus and method of an ANR circuit providing both feedforward-based and feedback-based ANR, possibly of a personal ANR device, compressing both feedforward and feedback reference sounds detected by feedforward and feedback microphones, respectively, in response to the acoustic energy of the feedforward reference noise sound reaching a predetermined level. | 10-28-2010 |
| 20100274565 | Method and Apparatus for Performing Packet Loss or Frame Erasure Concealment - A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window. | 10-28-2010 |
| 20100280830 | DECODER - A decoder for decoding an encoded audio signal from a first part of the encoded audio signal, wherein the decoder is configured to: receive a first part of an encoded audio signal; determine at least one scaling factor dependent on the first part of the encoded audio signal; scale the first part of the encoded audio signal dependent on the at least one scaling factor to produce a scaled encoded audio signal; and decode the scaled encoded audio signal. | 11-04-2010 |
| 20100280831 | Method and Device for Fast Algebraic Codebook Search in Speech and Audio Coding - A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions. A codevector of the algebraic codebook is computed using the positions of the pulses determined in the first and subsequent stages, wherein a number of the first and subsequent stages corresponds to the number of pulses in the codevectors of the algebraic codebook. | 11-04-2010 |
| 20100280832 | Packet Generator - A packet generator for generating packets from an input signal configured to: generate at least one first signal, dependent on the input signal, the first signal comprising a first relative time value; generate at least one second signal, dependent on the input signal and associated with the at least one first signal; and generate at least one indicator associated with each of the at least one second signal, each indicator dependent on the first relative time value. | 11-04-2010 |
| 20100280833 | ENCODING DEVICE, DECODING DEVICE, AND METHOD THEREOF - Provided is an encoding device which can suppress quality degradation of a decoded signal in a band extension for estimating a high range from a low range of a decoded signal. The encoding device includes: a first layer encoding unit ( | 11-04-2010 |
| 20100280834 | ENCODING DEVICE AND DECODING DEVICE - An encoding device ( | 11-04-2010 |
| 20100286988 | Hybrid Permanent/Reversible Dynamic Range Control System - A technique for controlling audio dynamic range in a manner that can be permanent, reversible, or anywhere in between, and can accomplish this goal in the baseband PCM or encoded domains. | 11-11-2010 |
| 20100286989 | RECORDING/REPRODUCTION DEVICE - An audio data processor ( | 11-11-2010 |
| 20100286990 | AUDIO ENCODER AND DECODER - The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal. | 11-11-2010 |
| 20100286991 | AUDIO ENCODER AND DECODER - The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit. | 11-11-2010 |
| 20100292992 | METHOD AND APPARATUS FOR MULTIPLEX SIGNAL DECODING - A stereo audio decoder decoding a multiplex signal into a stereo audio signal. A first filter module filters the multiplex signal to generate a summation signal. A sub-carrier module modulates the multiplex signal according to a sub-carrier frequency to generate a sub-carrier mixed signal having a first high frequency and a first low frequency component. The first low frequency component has a sub-carrier phase offset between the stereo audio decoder and the multiplex signal. The second filter module filters out the first high frequency component of the sub-carrier mixed signal to generate a sub-carrier pure signal having only the first low frequency component. A corrector generates a difference signal according to a correction signal and the multiplex signal. A channel separator obtains a left channel signal and a right channel signal of the stereo audio signal by decoding the summation signal and the difference signal. | 11-18-2010 |
| 20100292993 | Method and Device for Efficient Quantization of Transform Information in an Embedded Speech and Audio Codec - A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec while reducing a quantization noise comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients. Therefore, a method and device for reducing the quantization noise produced during coding of the error signal in the at least one upper layer comprises providing the spectral mask and, in the at least one upper layer, applying the spectral mask to the coding coefficients prior to quantizing the coding coefficients. | 11-18-2010 |
| 20100292994 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - A method of processing an audio signal is disclosed. The present invention includes receiving spectral data corresponding to a first band in a frequency band including the first band and a second band, determining a copy band based on frequency information of the copy band corresponding to a partial band of the first band, and generating spectral data of a target band corresponding to the second band using the spectral data of the copy band, wherein the copy band exists in an upper part of the first band. | 11-18-2010 |
| 20100299151 | Advanced Encoding Of Music Files - Example embodiments allow for the creation, distribution, and use of flexible media formats. Example embodiments may allow individual content files to be rendered in multiple formats and versions. In addition, example embodiments may provide for granular rights management, which may allow users to access content files on a feature-by-feature basis. | 11-25-2010 |
| 20100305952 | AUDIO ENCODING AND DECODING METHOD AND ASSOCIATED AUDIO ENCODER, AUDIO DECODER AND COMPUTER PROGRAMS - The invention relates to a method for sequencing spectral components of elements to be encoded (A | 12-02-2010 |
| 20100305953 | GENERATING A FRAME OF AUDIO DATA - A method of generating a frame of audio data for an audio signal from preceding audio data for the audio signal that precede the frame of audio data, the method comprising the steps of: predicting a predetermined number of data samples for the frame of audio data based on the preceding audio data, to form predicted data samples; identifying a section of the preceding audio data for use in generating the frame of audio data; and forming the audio data of the frame of audio data as a repetition ( | 12-02-2010 |
| 20100305954 | ENCODING METHOD AND SYSTEM - An encoding system includes a sampling unit, a computing unit, a comparing unit, a quantifying unit, and an encoding unit. The sampling unit obtains first sample data of a current sampling point and second sample data of a previous sampling point. The computing unit computes a data difference between the first sample data and the second sample data. The data difference includes a numeral and a sign. The comparing unit determines whether the data difference is more than or equal to 0 and outputs a determining result. The quantifying unit quantifies the numeral of the data difference. The encoding unit encodes the numeral of the data difference with or without the sign according to the determining result. | 12-02-2010 |
| 20100305955 | ENCODING METHOD, APPARATUS AND DEVICE AND DECODING METHOD - The present invention relates to encoding technology. The encoding method includes selecting a second encoding mode for encoding an input frame signal according to an analysis on signal characteristic of the input frame signal; obtaining coding demand values for a preset first encoding mode and the second encoding mode which are used to encode the input frame signal; determining, from the above encoding modes based on the coding demand values, an encoding mode for encoding the input frame signal; and multiplexing information of the determined encoding mode and encoded data which are encoded according to the determined encoding mode. Hence, the compatibility and the prioritization in terms of the encoding modes can be achieved. | 12-02-2010 |
| 20100305956 | METHOD AND AN APPARATUS FOR PROCESSING A SIGNAL - A method and apparatus for processing a signal are discussed. According to an embodiment, the method includes receiving extension information and a downmix signal decoded by either an audio coding scheme or a speech coding scheme, the downmix signal having a bandwidth of a low frequency signal; generating an upmixing signal from the downmix signal by using channel extension; determining an extension base signal corresponding to partial band of the upmixing signal based on the extension information; and generating an extended upmixing signal by applying the extension information to the extension base signal, the extended upmixing signal having a bandwidth extended by reconstructing a high frequency signal. | 12-02-2010 |
| 20100318367 | METHOD AND DEVICE FOR UPDATING STATUS OF SYNTHESIS FILTERS - A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal. | 12-16-2010 |
| 20100318368 | QUANTIZATION AND INVERSE QUANTIZATION FOR AUDIO - An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding. | 12-16-2010 |
| 20100324911 | CVSD DECODER STATE UPDATE AFTER PACKET LOSS - A system and method is described for updating the state of an audio decoder, such as a CVSD decoder, after a packet loss has occurred. In response to the loss of a packet, the system and method encodes audio samples produced by a packet loss concealment (PLC) algorithm and effectively passes the encoded audio samples through the audio decoder in lieu of the contents of the lost packet. This operation brings the state of the audio decoder into better synchronization with the state of a remote audio encoder, thereby reducing or minimizing the degrading effect of the packet loss on the perceived quality of an output audio signal produced by a voice processing system that includes the audio decoder. | 12-23-2010 |
| 20100324912 | Context-based arithmetic encoding apparatus and method and context-based arithmetic decoding apparatus and method - Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code. | 12-23-2010 |
| 20100324913 | Method and System for Block Adaptive Fractional-Bit Per Sample Encoding - A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, computing a data value range L of the N samples, determining a first encoding block size for the frame, mapping the N samples to normalized data values, computing a first block polynomial value for a block of samples in the frame of the first encoding block size, and encoding the first block polynomial value. | 12-23-2010 |
| 20100324914 | Adaptive Encoding of a Digital Signal with One or More Missing Values - A method of encoding samples in a digital signal is provided that includes receiving a plurality of samples of the digital signal, and encoding the plurality of samples, wherein an output number of bits is adapted for coding efficiency when a value in a range of possible distinct data values of the plurality of samples is not found in the plurality of samples. | 12-23-2010 |
| 20100324915 | ENCODING AND DECODING APPARATUSES FOR HIGH QUALITY MULTI-CHANNEL AUDIO CODEC - Provided is an encoding apparatus for a High Quality Multi-channel Audio Codec (HQMAC) and a decoding apparatus for the HQMAC. The encoding/decoding apparatuses for the HQMAC may perform a High Quality Multi-channel Audio Codec-Channel Based (HQMAC-CB) encoding or an HQMAC-CB decoding in accordance with characteristics of inputted audio signals to provide compatibility with a lower channel. | 12-23-2010 |
| 20100324916 | REMOVING TIME DELAYS IN SIGNAL PATHS - The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths. | 12-23-2010 |
| 20100332238 | Method and System for Lossless Value-Location Encoding - A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L−1 remaining possible distinct data values, wherein each of the L−1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L−1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L−1 remaining possible distinct data values in the N samples, and encoding the L−1 coding vectors. | 12-30-2010 |
| 20100332239 | Apparatus and method of encoding audio data and apparatus and method of decoding encoded audio data - An apparatus and method encode audio data, and an apparatus and method decode encoded audio data. An audio data encoding apparatus includes: a scalable encoding unit dividing audio data into a plurality of layers, representing the audio data in predetermined numbers of bits in each of the plurality of layers, and encoding a lower layer prior to encoding an upper layer and an upper bit of each layer prior to encoding a lower bit of each layer; an SBR encoding unit generating spectral band replication (SBR) data that has information with respect to audio data in a frequency band of frequencies equal to or greater than a predetermined frequency among the audio data to be encoded, and encoding the SBR data; and a bitstream production unit generating a bitstream using the encoded SBR data and the encoded audio data corresponding to a predetermined bitrate. | 12-30-2010 |
| 20110004478 | METHOD AND APPARATUS FOR TRANSFORMING BETWEEN DIFFERENT FILTER BANK DOMAINS - Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain. | 01-06-2011 |
| 20110004479 | HARMONIC TRANSPOSITION - The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L | 01-06-2011 |
| 20110015933 | SIGNAL ENCODING APPARATUS, SIGNAL DECODING APPARATUS, SIGNAL PROCESSING SYSTEM, SIGNAL ENCODING PROCESS METHOD, SIGNAL DECODING PROCESS METHOD, AND PROGRAM - Provided is a signal encoding apparatus including: an encoding unit which encodes a quantization value of a frequency spectrum in an input signal through a plurality of encoding algorithms; an amplitude change amount calculation unit which calculates an amplitude change amount with respect to the frequency spectrum based on a spectrum envelope of the frequency spectrum; and an encoding selection unit which selects the encoding algorithm according to a degree of deflection of an occurrence probability distribution of the quantization value in the amplitude change amount among the plurality of the encoding algorithms. | 01-20-2011 |
| 20110022397 | SLOT POSITION CODING OF TTT SYNTAX OF SPATIAL AUDIO CODING APPLICATION - Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with a fixed number of bits or a variable number of bits based on the data structure type. | 01-27-2011 |
| 20110022398 | METHOD AND APPARATUS FOR TRANSCODING AUDIO DATA - A method and apparatus for transcoding audio data. The method includes determining if AAC joint stereo exists, running a reference AC-3 rematrixing when the AAC joint stereo does not exist, when AAC joint stereo does exist, enabling rematrixing when the number of corresponding AAC bands is greater than half the size of the band, otherwise, running reference AC-3 rematrixing. | 01-27-2011 |
| 20110022399 | Auto Detection Method for Frame Header - A method for auto-detecting a frame header is provided. By searching and comparing content of input frames and predetermined sync words, decoding efficiency is increased and the probability of incurring program errors is reduced. Once decoding errors occur, an auto-recovery mechanism soon recovers the audio decoding system operation. | 01-27-2011 |
| 20110022400 | AUDIO RESUME PLAYBACK DEVICE AND AUDIO RESUME PLAYBACK METHOD - An audio playback device includes an interruption-information storage memory which holds frame information at the time of an interruption of playback, a compressed-stream control section which calculates a start position of reading the compressed stream from a recording medium, and reads the compressed stream accordingly, a compressed-stream decode section which decodes the compressed audio stream and transmits decoding information on the frame to the compressed-stream control section as additional resume information, and an output control section which outputs a decoding result. When playback is restarted after an interruption, playback can be restarted without any section not played back in terms of time and without any frequency range not played back, by calculating the start position of reading the compressed stream based on both the additional resume information and the frame information at the time of the interruption, and by reading the compressed stream accordingly. | 01-27-2011 |
| 20110022401 | SLOT POSITION CODING OF OTT SYNTAX OF SPATIAL AUDIO CODING APPLICATION - Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type. | 01-27-2011 |
| 20110029317 | DYNAMIC TIME SCALE MODIFICATION FOR REDUCED BIT RATE AUDIO CODING - Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder. | 02-03-2011 |
| 20110035225 | ENTROPY CODING USING ESCAPE CODES TO SWITCH BETWEEN PLURAL CODE TABLES - An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding. | 02-10-2011 |
| 20110035226 | COMPLEX-TRANSFORM CHANNEL CODING WITH EXTENDED-BAND FREQUENCY CODING - An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform. | 02-10-2011 |
| 20110035227 | METHOD AND APPARATUS FOR ENCODING/DECODING AN AUDIO SIGNAL BY USING AUDIO SEMANTIC INFORMATION - An audio signal encoding method and apparatus and an audio signal decoding method and apparatus are provided. The audio signal encoding method includes: transforming an audio signal into a signal of a frequency domain; extracting semantic information from the audio signal; variably reconfiguring one or more sub-bands included in the audio signal by segmenting or grouping the one or more sub-bands using the extracted semantic information; and generating a quantized bitstream by calculating a quantization step size and a scale factor with respect to a reconfigured sub-band of the one or more sub-bands. | 02-10-2011 |
| 20110040566 | METHOD AND APPARATUS FOR ENCODING AND DECODING RESIDUAL SIGNAL - A residual signal decoding method including dividing a signal into a plurality of sub-bands in a frequency domain, wherein the signal is in a residual signal and is encoded with respect to an effective residual signal exceeding a masking curve that is generated with respect to a multi-channel audio signal, transforming the frequency domain into a time domain, and restoring the signal as an effective residual signal by synthesizing signals of the domain-transformed sub-bands. | 02-17-2011 |
| 20110046963 | MULTI-CHANNEL AUDIO DECODING METHOD AND APPARATUS THEREFOR - Provided is a multi-channel audio decoding method and apparatus therefor, the method involving decoding filter bank coefficients of a plurality of bands from a bitstream having a predetermined format; performing frequency transformation on the decoded filter bank coefficients of the plurality of bands, with respect to each of the plurality of bands; compensating for a phase of each of the plurality of bands according to a predetermined phase compensation value, and serially band-synthesizing the frequency-transformed coefficients of each of the plurality of phase-compensated bands on a frequency domain; and decoding a multi-channel audio signal from the band-synthesized frequency-transformed coefficients. | 02-24-2011 |
| 20110046964 | METHOD AND APPARATUS FOR ENCODING MULTI-CHANNEL AUDIO SIGNAL AND METHOD AND APPARATUS FOR DECODING MULTI-CHANNEL AUDIO SIGNAL - A method and apparatus which encode multi-channel audio signals and a method and apparatus which decode multi-channel audio signals. When encoding, a downmixed audio signal, first additional information for restoring multi-channel audio signals from the downmixed audio signal, and second additional information representing characteristics of a residual signal are multiplexed. When decoding, restored multi-channel audio signals having a predetermined phase difference are combined using the second additional information, and an audio signal of each channel is corrected, in order to improve quality of the restored audio signals. | 02-24-2011 |
| 20110054911 | Enhanced Audio Decoder - Methods, systems, and apparatus are presented for decoding an audio signal that includes bandwidth extension data. An audio signal that includes core audio data and bandwidth extension data can be received in a decoder. The core audio data can be associated with a core portion of an audio signal, such as the frequency range below a cutoff frequency, and the bandwidth extension data can be associated with an extended portion of the audio signal, such as a frequency range above the cutoff frequency. The core audio data can be decoded to generate a decoded core audio signal in a time domain representation. Further, an extended portion of the audio signal can be reconstructed in accordance with extension data and decoded core audio signal. Additionally, the decoded core audio signal can be lowpass filtered and the extended portion can be highpass filtered before being combined to generate a decoded output signal. | 03-03-2011 |
| 20110054912 | SYSTEM AND METHOD OF STORING TELEPHONE CONVERSATIONS - A method and system of storing telephone conversation data to a third party database storage unit is disclosed. The method includes detecting a telephone call initiated in a mobile telephone, recording telephone conversation data, detecting a termination of the telephone call, and transferring the recorded telephone conversation data to a third party database storage unit. The system includes an application program interface installed on a mobile telephone, an internet gateway, and a third party database storage unit configured to store data recorded by the application program interface. | 03-03-2011 |
| 20110054913 | ASYNCHRONOUS SAMPLING RATE CONVERTER FOR AUDIO APPLICATIONS - In recent years, it has become commonplace for portable devices to generate analog audio signals from numerous sources, meaning that the codecs employed in these portable devices need to be able to utilize various digital bit streams at different sampling rates. To date, however, the circuitry for asynchronous sampling rate conversions for multiple bit streams has been complex, rigid, and power hungry. Here, a codec is provided which uses miniDSP cores to perform asynchronous sampling rate conversion efficiently and with reduced power consumption compared to other conventional codecs. | 03-03-2011 |
| 20110054914 | Method for Reduction of Aliasing Introduced by Spectral Envelope Adjustment in Real-Valued Filterbanks - The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. | 03-03-2011 |
| 20110054915 | COMPUTING CIRCUITS AND METHOD FOR RUNNING AN MPEG-2 AAC OR MPEG-4 AAC AUDIO DECODING ALGORITHM ON PROGRAMMABLE PROCESSORS - The present invention relates to computing circuits and method for running an MPEG-2 AAC or MPEG-4 AAC algorithm efficiently, which is used as an audio compression algorithm in multi-channel high-quality audio systems, on programmable processors. In accordance with the present invention, the IMDCT process which takes large part of the amount of the operations in implementation of an MPEG-2/4 AAC algorithm can be performed in efficient. In addition, while the architecture of the existing digital signal processor is still used, the performance can be improved by means of the addition of the architecture of the address generator, Huffman decoder, and bit processing architecture. After all, to design and change the programmable processor is facilitated. | 03-03-2011 |
| 20110054916 | MULTI-CHANNEL AUDIO ENCODING AND DECODING - An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes. | 03-03-2011 |
| 20110060593 | OUTPUT CIRCUIT FOR AUDIO CODEC CHIP - An output circuit for audio codec chip includes a noise eliminating circuit electrically coupled to the audio codec chip for eliminating noise signals. The noise eliminating circuit includes a first switch and a second switch. When the audio codec chip output signals jump from low voltage level to high voltage level, the noise signals are grounded via the first and second switches respectively. | 03-10-2011 |
| 20110060594 | APPARATUS AND METHOD FOR ADAPTIVE AUDIO CODING - An audio encoder capable of implementing a plurality of encoding functions, wherein an adaptation controller adjusts the implementation of the encoding functions in response to feedback received by the adaptation controller during use. The adjustment may involve adapting encoding algorithms or selecting alternative encoding algorithms. The encoder may also include an operations scheduler to adjust the order in which the encoding functions are applied. The feedback may be received from internally of the encoder, for example from the currently implemented encoding functions, or from externally of the encoder. A corresponding decoder is also provided. | 03-10-2011 |
| 20110060595 | APPARATUS AND METHOD FOR ADAPTIVE AUDIO CODING - An audio encoder capable of implementing a plurality of encoding functions, wherein an adaptation controller adjusts the implementation of the encoding functions in response to feedback received by the adaptation controller during use. The adjustment may involve adapting encoding algorithms or selecting alternative encoding algorithms. The encoder may also include an operations scheduler to adjust the order in which the encoding functions are applied. The feedback may be received from internally of the encoder, for example from the currently implemented encoding functions, or from externally of the encoder. A corresponding decoder is also provided. | 03-10-2011 |
| 20110060596 | Method for decoding an audio signal that has a base layer and an enhancement layer - An audio signal may have a BL and an EL, wherein the EL represents additional information for enhancing the quality of the BL audio content. Decoding of such dual-layer signals usually comprises partial decoding of the BL data, wherein frequency bins of the BL are restored, mapping the restored frequency bins to the MDCT domain, adding them to the decoded EL and performing inverse Integer MDCT. A low-complexity method for decoding comprises reverse mapping of the decoded EL data, adding the reverse mapped EL data to the partially decoded BL data and filtering the sum, using the inverse BL filter bank. | 03-10-2011 |
| 20110060597 | MULTI-CHANNEL AUDIO ENCODING AND DECODING - An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes. | 03-10-2011 |
| 20110082699 | SIGNAL CODING AND DECODING - An encoding device ( | 04-07-2011 |
| 20110082700 | SIGNAL CODING AND DECODING - An encoding device ( | 04-07-2011 |
| 20110087494 | Apparatus and method of encoding audio signal by switching frequency domain transformation scheme and time domain transformation scheme - Provided are an apparatus of encoding an audio signal by switching a time domain transformation scheme and a frequency domain transformation scheme, and an apparatus of decoding an audio signal by switching the time domain transformation scheme and the frequency domain transformation scheme. When any one transformation scheme of the time domain transformation scheme and the frequency domain transformation is switched into the other transformation scheme, an audio signal before and after the switching is encoded using an additionally transformed audio signal, or an encoded audio signal is decoded. | 04-14-2011 |
| 20110093275 | CODING METHOD, DECODING METHOD, CODEC METHOD, CODEC SYSTEM AND RELEVANT APPARATUSES - A coding method, a decoding method, a coding-decoding (codec) method, a codec system and relevant apparatuses are disclosed. The coding method includes: obtaining an amplitude vector and a length vector corresponding to a vector to be coded; sorting elements of the amplitude vector and elements of the length vector; and obtaining a position index value according to the sorted amplitude vector and the sorted length vector. A decoding method, a codec system, and relevant apparatuses are also provided. | 04-21-2011 |
| 20110093276 | APPARATUS - A method comprising receiving at a user equipment encrypted content. The content is stored in said user equipment in an encrypted form. At least one key for decryption of said stored encrypted content is stored in the user equipment. | 04-21-2011 |
| 20110099018 | Apparatus and Method for Calculating Bandwidth Extension Data Using a Spectral Tilt Controlled Framing - An apparatus for calculating bandwidth extension data of an audio signal in a bandwidth extension system, in which a first spectral band is encoded with a first number of bits and a second spectral band different from the first spectral band is encoded with a second number of bits, the second number of bits being smaller than the first number of bits, has a controllable bandwidth extension parameter calculator for calculating bandwidth extension parameters for the second frequency band in a frame-wise manner for a sequence of frames of the audio signal. Each frame has a controllable start time instant. The apparatus additionally includes a spectral tilt detector for detecting a spectral tilt in a time portion of the audio signal and for signaling the start time instant for the individual frames of the audio signal depending on spectral tilt. | 04-28-2011 |
| 20110099019 | USER ATTRIBUTE DISTRIBUTION FOR NETWORK/PEER ASSISTED SPEECH CODING - Systems, methods and apparatuses are described herein for distributing user attribute information about users of a communications system to communication terminals, which use the user attribute information to configure a speech codec to operate in a speaker-dependent manner during a communication session, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals. | 04-28-2011 |
| 20110099020 | Method for Dynamically Adjusting Audio Decoding Process - A method for dynamically arranging DSP tasks. The method comprises receiving an audio bit stream, checking a remaining execution time as the DSP transforms the audio information into spectral information, simplifying the step of transforming the audio information when the DSP detects that the remaining execution time is shorter then a predetermined interval, and skipping one section of the audio information and decoding the remaining section when the execution time is less than a predetermined interval. | 04-28-2011 |
| 20110106540 | STEREO CODING AND DECODING METHOD AND APPARATUS THEREOF - A method of encoding input signals (l, r) to generate encoded data ( | 05-05-2011 |
| 20110106541 | Partially Complex Modulated Filter Bank - An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal. | 05-05-2011 |
| 20110106542 | Audio Signal Decoder, Time Warp Contour Data Provider, Method and Computer Program - An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation having a time warp contour evolution information has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value on the basis of a time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded audio signal representation and using the rescaled version of the time warp contour. | 05-05-2011 |
| 20110106543 | SPATIAL SYNTHESIS OF MULTICHANNEL AUDIO SIGNALS - A method and associated device are provided for spatial synthesis of a sum signal to obtain at least two output signals, the sum signal as well as the spatialization parameters being output from a parametric coding by matrixing of an original multi-channel signal. The method comprises: decorrelation of the sum signal to obtain a decorrelated signal; applying a synthesis matrix, whose coefficients depend on the spatialization parameters, to the decorrelated signal and to the sum signal to obtain said output signals, wherein for at least one range of value of at least one spatialization parameter, the coefficients of the synthesis matrix are determined according to a criterion of minimizing a quantitative function, relating to the quantity of decorrelated signal in each of the output signals obtained by applying the synthesis matrix. | 05-05-2011 |
| 20110106544 | ADAPTING MASKING THRESHOLDS FOR ENCODING A LOW FREQUENCY TRANSIENT SIGNAL IN AUDIO DATA - An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal. | 05-05-2011 |
| 20110106545 | TEMPORAL AND SPATIAL SHAPING OF MULTI-CHANNEL AUDIO SIGNALS - A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel. On the other hand, the shaping time scale is larger than the time scale of the sampling values, significantly reducing the amount of data needed by the wave form parameter representation. | 05-05-2011 |
| 20110112841 | Apparatus - An apparatus configured to model an encoded signal to estimate at least one distribution of the signal, rotate the signal with respect to a lattice, for lattice quantization of the signal, dependent on the at least one distribution of the signal, and quantize the signal rotated with respect to the lattice. | 05-12-2011 |
| 20110112842 | METHOD AND APPARATUS FOR EDITING AUDIO OBJECT IN SPATIAL INFORMATION-BASED MULTI-OBJECT AUDIO CODING APPARATUS - Disclosed is an audio object editing apparatus of a multi-object audio coding apparatus. The audio object editing apparatus of the multi-object audio coding apparatus may include an object information extracting unit to receive an object bit stream and to extract object information from the object bit stream, a downmix processing unit to receive a downmix signal, and to control the downmix signal using object editing information and the object information, and a bit stream processing unit to edit the object information according to the object editing information, and to generate a controlled object bit stream based on the edited object information. | 05-12-2011 |
| 20110112843 | SIGNAL ANALYZING DEVICE, SIGNAL CONTROL DEVICE, AND METHOD AND PROGRAM THEREFOR - Provided is a signal analyzing device comprising a separate information calculating unit for generating separate information to separate an input signal mixed with a sound source signal, into the sound source signal. The signal-analyzing device is characterized by sending out the input signal and the separate information. | 05-12-2011 |
| 20110112844 | METHOD AND APPARATUS FOR ESTIMATING HIGH-BAND ENERGY IN A BANDWIDTH EXTENSION SYSTEM | 05-12-2011 |
| 20110112845 | METHOD AND APPARATUS FOR ESTIMATING HIGH-BAND ENERGY IN A BANDWIDTH EXTENSION SYSTEM | 05-12-2011 |
| 20110119064 | AUDIO DATA BIT RATE DETECTOR - A detector for determining an audio data bit rate of a pre-compressed audio elementary stream (AES), in which the pre-compressed AES includes a plurality of preamble length fields is provided. The detector includes an analyzer module configured to determine one or more candidate audio data bit rates of the AES from the plurality of preamble length fields, a detector module configured to determine whether one or more of the candidate audio data bit rates are correct, and a selector module configured to select one of the one or more candidate audio data bit rates determined to be correct as the audio data bit rate of the pre-compressed AES. | 05-19-2011 |
| 20110119065 | EMBODIED MUSIC SYSTEM - An embodied music system. The system creates an interactive interface between a listener and the external environment. The system includes a physical device located in the environment that provides sensory input to the listener. An audio signal of the system is adapted to be heard by the listener. An encoder embeds inaudible control data into the audio signal. A decoder extracts the control data from the audio signal and transmits the control data to the physical device, thereby controlling operation of the device. Finally, an audio reproduction device is connected to the decoder and plays the audio signal for the listener. The embodied music system allows the listener to experience multi-sensory compositions. | 05-19-2011 |
| 20110119066 | SIGNAL ENCODING DEVICE AND SIGNAL ENCODING METHOD, SIGNAL DECODING DEVICE AND SIGNAL DECODING METHOD, PROGRAM, AND RECORDING MEDIUM - A signal encoding device for encoding an input time-series signal includes: partitioning means; low-frequency encoding means; high-frequency gain information generating means; low-frequency reference value information generating means; high-frequency gain difference information generating means; high-frequency gain difference information encoding means; and multiplexing means. | 05-19-2011 |
| 20110119067 | APPARATUS FOR SIGNAL STATE DECISION OF AUDIO SIGNAL - A module capable of appropriately selecting a linear predictive coding (LPC)-based or a code excitation linear prediction (CELP)-based speech or audio encoder and a transform-based audio encoder according to a feature of an input signal is a module that performs as a bridge for overcoming a performance barrier between a conventional LPC-based encoder and an audio encoder. Also, an integral audio encoder that provides consistent audio quality regardless of a type of the input audio signal can be designed based on the module. | 05-19-2011 |
| 20110125505 | Method and Device for Efficient Frame Erasure Concealment in Speech Codecs - A method and device for concealing frame erasures caused by frames of an encoded sound signal erased during transmission from an encoder to a decoder and for recovery of the decoder after frame erasures comprise, in the encoder, determining concealment/recovery parameters including at least phase information related to frames of the encoded sound signal. The concealment/recovery parameters determined in the encoder are transmitted to the decoder and, in the decoder, frame erasure concealment is conducted in response to the received concealment/recovery parameters. The frame erasure concealment comprises resynchronizing, in response to the received phase information, the erasure-concealed frames with corresponding frames of the sound signal encoded at the encoder. When no concealment/recovery parameters are transmitted to the decoder, a phase information of each frame of the encoded sound signal that has been erased during transmission from the encoder to the decoder is estimated in the decoder. Also, frame erasure concealment is conducted in the decoder in response to the estimated phase information, wherein the frame erasure concealment comprises resynchronizing, in response to the estimated phase information, each erasure-concealed frame with a corresponding frame of the sound signal encoded at the encoder. | 05-26-2011 |
| 20110125506 | RATE-DISTORTION OPTIMIZATION FOR ADVANCED AUDIO CODING - A method for optimization of rate-distortion for Advanced Audio Coding (AAC). The method provides for the identification of quantized spectral coefficient sequences for optimization of rate-distortion. The method also provides joint optimization of scale factors, Huffman codebooks and quantized spectral coefficient sequences for minimization of a rate-distortion cost. The method provides an iterative rate-distortion optimization algorithm for AAC encoding. In each iteration, the method first finds the optimal scale factors and quantized spectral coefficients when Huffman codebooks are fixed, then updates Huffman codebooks and quantized spectral coefficients given the optimized scale factors. The iterations may be applied until a predetermined threshold is attained. | 05-26-2011 |
| 20110125507 | Method and System for Frequency Domain Postfiltering of Encoded Audio Data in a Decoder - A decoder configured to generate decoded audio data (e.g., decoded speech data) and including a postfilter coupled and configured to filter encoded audio data in the frequency domain, methods for frequency domain postfiltering of encoded audio data in a decoder, and methods for decoding encoded audio data in a decoder including by postfiltering encoded audio data in the frequency domain in the decoder. In some embodiments, the decoder is configured to decode input encoded audio without performing any time-to-frequency domain transform on encoded audio data to prepare data for postfiltering. Typically, the postfiltering improves the quality of the decoded audio signal by attenuating spectral valley regions thereof to remove excess quantization noise present in the encoded input audio while preserving formants of the decoded audio signal to avoid introducing unnecessary distortion. | 05-26-2011 |
| 20110137659 | Frequency Band Extension Apparatus and Method, Encoding Apparatus and Method, Decoding Apparatus and Method, and Program - The present invention relates to a frequency band extension apparatus and method, an encoding apparatus and method, a decoding apparatus and method, and a program, with which a music signal can be reproduced with higher sound quality by means of frequency band extension. | 06-09-2011 |
| 20110137660 | Encoding and decoding speech signals - A method and apparatus for transmitting an audio signal over a communication channel comprising encoding the audio signal with an encoder | 06-09-2011 |
| 20110137661 | QUANTIZING DEVICE, ENCODING DEVICE, QUANTIZING METHOD, AND ENCODING METHOD - A quantizing device for more efficient quantization realized by lessening the computational complexity of quantization of a balance weighting factor. The device includes a power/correlation calculating unit ( | 06-09-2011 |
| 20110137662 | Audio Signal Transformatting - This invention relates to reformatting a plurality of audio input signals from a first format to a second format by applying them to a dynamically-varying transformatting matrix. In particular, this invention obtains information attributable to the direction and intensity of one or more directional signal components, calculates the transformatting matrix based on the first and second rules, and applies the audio input signals to the transformatting matrix to produce output signals. | 06-09-2011 |
| 20110145003 | Simultaneous Time-Domain and Frequency-Domain Noise Shaping for TDAC Transforms - A frequency-domain noise shaping method and device interpolates a spectral shape and a time-domain envelope of a quantization noise in a windowed and transform-coded audio signal. In the method and device, transform coefficients of the windowed and transform-coded audio signal are split into a plurality of spectral bands. For each spectral band, a first gain representing a spectral shape of the quantization noise at a first transition between a first time window and a second time window is calculated, a second gain representing a spectral shape of the quantization noise at a second transition between the second time window and a third time window is calculated, and the transform coefficients of the second time window are filtered based on the first and second gains, to interpolate between the first and second transitions the spectral shape and the time-domain envelope of the quantization noise. | 06-16-2011 |
| 20110153333 | Forward Time-Domain Aliasing Cancellation with Application in Weighted or Original Signal Domain - The present invention relates to methods and devices for forward time-domain aliasing cancellation in a coded signal transmitted from a coder to a decoder. Information related to correction of the time-domain aliasing in the coded signal is calculated at the coder and added in a bitstream sent from the coder to the decoder. The decoder receives the bitstream and cancels the time-domain aliasing in the coded signal in response to the information comprised in the bitstream. The information may be representative of a difference between a frame of audio signal to be encoded in a first coding mode and a decoded signal from the frame including time-domain aliasing effects. | 06-23-2011 |
| 20110153334 | METHOD FOR EXTRACTING PROBABILITY MODEL VALUE FROM PROBABILITY MODEL TABLE AND METHOD AND APPARATUS FOR DECODING SYMBOL VALUE BY USING THE SAME - A method for extracting a probability model value from a probability model table and a method and apparatus for decoding a symbol value using the same are provided. The method for extracting a probability model value from a probability model table includes: segmenting and reducing a probability model table including a plurality of probability model values; disposing indexes on the basis of the segmented and reduced probability model table; and searching the probability model table for a probability model value by using the disposed indexes. | 06-23-2011 |
| 20110153335 | METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNALS - An audio signal processing method is disclosed. The audio signal processing method includes receiving a residual and long term prediction information, performing inverse frequency mapping with respect to the residual to generate a synthesized residual, and performing long term synthesis based on the synthesized residual and the long term prediction information to generate a synthesized audio signal of a current frame, wherein the long term prediction information comprises a final prediction gain and a final pitch lag, the final pitch lag has a range starting with 0, and the long term synthesis is performed based on a synthesized audio signal of a frame comprising a preceding frame. | 06-23-2011 |
| 20110153336 | MULTI-MODE SCHEME FOR IMPROVED CODING OF AUDIO - The present invention relates to an improved scheme for coding of audio. In particular, the present invention relates to an encoder device and a method for coding an input signal in an encoder system. The method comprises applying a first mode to the input signal to form a first output and applying a second mode to the input signal to form a second output. A first processed output is then formed from at least a part of the first output, and a second processed output is formed from at least a part of the second output. Forming a second processed output comprises estimating a part of the input signal from at least a part of the second output. Then, an optimum mode is determined based on the firstprocessedoutput and the secondprocessedoutput, and the output according to the optimum mode is selected. | 06-23-2011 |
| 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 |
| 20110161088 | Time Warp Contour Calculator, Audio Signal Encoder, Encoded Audio Signal Representation, Methods and Computer Program - A time warp contour calculator for use in an audio signal decoder is configured to receive an encoded warp ratio information, to derive a sequence of warp ratio values from the encoded warp ratio information, and to obtain warp contour node values starting from a time warp contour start value. | 06-30-2011 |
| 20110166864 | QUANTIZATION MATRICES FOR DIGITAL AUDIO - Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode. | 07-07-2011 |
| 20110166865 | COMPUTATIONALLY EFFICIENT AUDIO CODER - The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal. The audio coder further compares the computed short-block characteristics to a set of threshold values to detect presence of an attack in each of the short-blocks and changes the long-block frame length of one or more short-blocks upon detecting the attack in the respective one or more short-blocks. | 07-07-2011 |
| 20110166866 | SIGNAL SYNTHESIZING - A method of synthesizing a first (L) and a second (R) output signal from an input signal (x). The method comprises: filtering ( | 07-07-2011 |
| 20110166867 | MULTI-OBJECT AUDIO ENCODING AND DECODING APPARATUS SUPPORTING POST DOWN-MIX SIGNAL - A multi-object audio encoding and decoding apparatus supporting a post downmix signal may be provided. The multi-object audio encoding apparatus may include: an object information extraction and downmix generation unit to generate object information and a downmix signal from input object signals; a parameter determination unit to determine a downmix information parameter using the extracted downmix signal and the post downmix signal; and a bitstream generation unit to combine the object information and the downmix information parameter, and to generate an object bitstream. | 07-07-2011 |
| 20110173004 | Device and Method for Noise Shaping in a Multilayer Embedded Codec Interoperable with the ITU-T G.711 Standard - A device and method for shaping noise during encoding of an input sound signal comprise pre-emphasizing the input signal or a decoded signal from a given sound signal codec to produce a pre-emphasized signal, computing a filter transfer function based on the pre-emphasized signal, and shaping the noise by filtering the noise through the transfer function to produce a shaped noise signal, wherein the noise shaping comprises producing a noise feedback. A device and method for noise shaping in a multilayer codec, including at least Layer 1 and 2, comprise: at an encoder, producing an encoded sound signal in Layer 1 including Layer 1 noise shaping, and producing a Layer 2 enhancement signal; at a decoder, decoding the Layer 1 encoded sound signal to produce a synthesis signal, decoding the enhancement signal, computing a filter transfer function based on the synthesis signal, filtering the enhancement signal through the transfer function to produce a Layer 2 filtered enhancement signal, and adding the filtered enhancement signal to the synthesis signal to produce an output signal including contributions from Layer 1 and 2. | 07-14-2011 |
| 20110173005 | Efficient Use of Phase Information in Audio Encoding and Decoding - An efficient encoded representation of a first and a second input audio signal can be derived using correlation information indicating a correlation between the first and the second input audio signals, when a signal characterization information, indicating at least a first or a second, different characteristic of the input audio signal is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic. | 07-14-2011 |
| 20110173006 | Audio Signal Synthesizer and Audio Signal Encoder - An audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation. | 07-14-2011 |
| 20110173007 | Audio Encoder and Audio Decoder - An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream. | 07-14-2011 |
| 20110173008 | Audio Encoder and Decoder for Encoding Frames of Sampled Audio Signals - An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame has a number of time domain audio samples, having a predictive coding analysis stage for determining information on coefficients of a synthesis filter and information on a prediction domain frame based on a frame of audio samples. The audio encoder further has a frequency domain transformer for transforming a frame of audio samples to the frequency domain to obtain a frame spectrum and an encoding domain decider for deciding whether encoded data for a frame is based on the information on the coefficients and on the information on the prediction domain frame, or based on the frame spectrum. Moreover, the audio encoder has a controller for determining an information on a switching coefficient when the encoding domain decider decides that encoded data of a current frame is based on the information on the coefficients and the information on the prediction domain frame when encoded data of a previous frame was encoded based on a previous frame spectrum and a redundancy reducing encoder for encoding the information on the prediction domain frame, the information on the coefficients, the information on the switching coefficient and/or the frame spectrum. | 07-14-2011 |
| 20110173009 | Apparatus and Method for Encoding/Decoding an Audio Signal Using an Aliasing Switch Scheme - An apparatus for encoding an audio signal includes the windower for windowing a first block of the audio signal using an analysis window having an aliasing portion and a further portion. The apparatus furthermore includes a processor for processing the first sub-block of the audio signal associated with the aliasing portion by transforming the sub-block from a domain into a different domain subsequent to windowing the first sub-block to obtain the processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain into the different domain before windowing the second sub-block to obtain a processed second sub-block. The apparatus furthermore includes a transformer for converting the processed first sub-block and the processed second sub-block from the different domain into a further different domain using the same block transform rule to obtain a converted first block which may then be compressed using any of the well-known data compression algorithms. Thus, a critically sampled switch between two coding modes can be obtained, since aliasing portions occurring in two different domains are matched to each other. | 07-14-2011 |
| 20110173010 | Audio Encoder and Decoder for Encoding and Decoding Audio Samples - An audio encoder for encoding audio samples has a first time domain aliasing introducing encoder configured to decode audio samples in a first encoding domain and having a first framing rule, a start window and a stop window. The audio encoder further has a second encoder configured to encode samples in a second encoding domain and having a predetermined frame size number of audio samples, and a coding warm-up period number of audio samples, the second encoder having a different second framing rule, a frame of the second encoder being an encoded representation of a number of successive audio samples that is equal to the predetermined frame size number of audio samples. The audio encoder further has a controller switching from the first to the second encoder and for modifying the second framing rule or for modifying the start or the stop window of the first encoder. | 07-14-2011 |
| 20110173011 | Audio Encoder and Decoder for Encoding and Decoding Frames of a Sampled Audio Signal - An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra. | 07-14-2011 |
| 20110173012 | Noise Filler, Noise Filling Parameter Calculator Encoded Audio Signal Representation, Methods and Computer Program - A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter. Accordingly, an encoded audio signal representation representing the audio signal can be obtained. | 07-14-2011 |
| 20110173013 | Adaptive Variable Bit Rate Audio Encoding - A method and apparatus for producing a variable bit rate audio signal is disclosed. An audio signal is encoded into a plurality of encoded audio signals at different bit rates. A variable bit rate audio signal is produced by selecting between the plurality of encoded audio frames of different bit rates in accordance with a selection criterion. | 07-14-2011 |
| 20110173014 | Audio Decoding - Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. Each frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied, and (iii) window information. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes. Subband samples are then generated by dequantizing the decoded quantization indexes, and a sequence of different window functions that were applied within a single frame of the audio data is identified based on the window information. Time-domain audio data are obtained by inverse-transforming the subband samples and using the plural different window functions indicated by the window information. | 07-14-2011 |
| 20110178806 | ENCODER, ENCODING SYSTEM, AND ENCODING METHOD - An encoding device includes, an estimation unit to estimate a decoded signal of a plurality of channels based on a down-mix signal obtained by down-mixing an input signal of the plurality of channels, similarity between the channels of the input signal, and an intensity difference between the channels of the input signal; an analysis unit to analyze a phase of the input signal and a phase of the decoded signal; a calculation unit to calculate phase information based on the phase of the input signal and the phase of the decoded signal; and a coding unit to encode the similarity between the channels of the input signal, the intensity difference between the channels of the input signal, and the phase information. | 07-21-2011 |
| 20110178807 | METHOD AND APPARATUS FOR DECODING AUDIO SIGNAL - Provided are a method and an apparatus for decoding an audio signal. A method for decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses includes decoding the encoded audio signal, setting a smoothing frequency band of the decoded audio signal according to a layer structure of the layered sinusoidal pulse coding scheme, dividing the smoothing frequency band into one or more subbands, and smoothing the decoded audio signal on a subband-by-subband basis. Accordingly, a decoding operation time can be reduced and the quality of a synthesized signal can be improved by variably setting a frequency band to be smoothed, when decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses. | 07-21-2011 |
| 20110178808 | 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. | 07-21-2011 |
| 20110178809 | CRITICAL SAMPLING ENCODING WITH A PREDICTIVE ENCODER - A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding. | 07-21-2011 |
| 20110191110 | Multi-parameter physical audio signal decoding system - A multi-parameter physical audio signal decoding system includes an audio capturing device, a signal pre-processor, a signal decoder, a data modulator and a display device. The audio capturing device receives audio signals, and transforms them into digital signals in real time. The digital signals are transmitted to the signal pre-processor. The signal decoder decodes the digital signals. The data modulator removes signal noises. The display device displays the modulated digital signals. In a non-real-time decoding process, an audio file recorded with physical signals is transmitted to the pre-processor to proceed the decoding process. | 08-04-2011 |
| 20110191111 | Audio Packet Loss Concealment by Transform Interpolation - In audio processing for an audio or video conference, a terminal receives audio packets having transform coefficients for reconstructing an audio signal that has undergone transform coding. When receiving the packets, the terminal determines whether there are any missing packets and interpolates transform coefficients from the preceding and following good frames. To interpolate the missing coefficients, the terminal weights first coefficients from the preceding good frame with a first weighting, weights second coefficients from the following good frame with a second weighting, and sums these weighted coefficients together for insertion into the missing packets. The weightings can be based on the audio frequency and/or the number of missing packets involved. From this interpolation, the terminal produces an output audio signal by inverse transforming the coefficients. | 08-04-2011 |
| 20110191112 | ENCODER - An encoder for encoding an audio signal comprising at least two channels, the encoder configured to determine a first indicator dependent on the relative energies of a first and a second of the at least two channels for a first time period, determine at least two second indicators dependent on the relative energies of the first and the second of the at least two channels for the first time period, and generate an encoded signal comprising at least one part dependent on the first indicator and the at least two second indicators. | 08-04-2011 |
| 20110191113 | METHOD AND APPARATUS FOR CANONICAL NONLINEAR ANALYSIS OF AUDIO SIGNALS - The present invention is directed to systems and methods designed to ascertain the structure of acoustic signals. The approach involves an alternative transform of an acoustic input signal, utilizing a network of nonlinear oscillators in which each oscillator is tuned to a distinct frequency. Each oscillator receives input and interacts with the other oscillators in the network, yielding nonlinear resonances that are used to identify structure in an acoustic input signal. The output of the nonlinear frequency transform can be used as input to a system that will provide further analysis of the signal. According to one embodiment, the nonlinear responses are defined as a network of n expanded canonical oscillators z | 08-04-2011 |
| 20110196684 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique. | 08-11-2011 |
| 20110196685 | METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS - Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method generating a third downmix signal by combining a first downmix signal extracted from a first audio signal and a second downmix signal extracted from a second audio signal; generating third object-based side information by combining first object-based side information extracted from the first audio signal and second object-based side information extracted from the second audio signal; converting the third object-based side information into channel-based side information; and generating a multi-channel audio signal using the third downmix signal and the channel-based side information. | 08-11-2011 |
| 20110196686 | SPECTRUM CODING APPARATUS, SPECTRUM DECODING APPARATUS, ACOUSTIC SIGNAL TRANSMISSION APPARATUS, ACOUSTIC SIGNAL RECEPTION APPARATUS AND METHODS THEREOF - A spectrum coding apparatus capable of performing coding at a low bit rate and with high quality is disclosed. This apparatus is provided with a section that performs the frequency transformation of a first signal and calculates a first spectrum, a section that converts the frequency of a second signal and calculates a second spectrum, a section that estimates the shape of the second spectrum in a hand of FL≦k| 08-11-2011 | |
| 20110196687 | 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. | 08-11-2011 |
| 20110202352 | Apparatus and a Method for Generating Bandwidth Extension Output Data - An apparatus for generating bandwidth extension output data for an audio signal has a noise floor measurer, a signal energy characterizer and a processor. The audio signal has components in a first frequency band and components in a second frequency band, the bandwidth extension output data are adapted to control a synthesis of the components in the second frequency band. The noise floor measurer measures noise floor data of the second frequency band for a time portion of the audio signal. The signal energy characterizer derives energy distribution data, the energy distribution data characterizing an energy distribution in a spectrum of the time portion of the audio signal. The processor combines the noise floor data and the energy distribution data to obtain the bandwidth extension output data. | 08-18-2011 |
| 20110202353 | Apparatus and a Method for Decoding an Encoded Audio Signal - An apparatus for decoding an encoded audio signal having first and second portions encoded in accordance with first and second encoding algorithms, respectively, BWE parameters for the first and second portions and a coding mode information indicating a first or a second decoding algorithm, includes first and second decoders, a BWE module and a controller. The decoders decode portions in accordance with decoding algorithms for time portions of the encoded signal to obtain decoded signals. The BWE module has a controllable crossover frequency and is configured for performing a bandwidth extension algorithm using the first decoded signal and the BWE parameters for the first portion, and for performing a bandwidth extension algorithm using the second decoded signal and the bandwidth extension parameter for the second portion. The controller controls the crossover frequency for the BWE module in accordance with the coding mode information. | 08-18-2011 |
| 20110202354 | Low Bitrate Audio Encoding/Decoding Scheme Having Cascaded Switches - An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and wherein the second encoding branch furthermore has a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder such as a spectral domain decoding branch, a second domain decoder such as an LPC domain decoding branch for decoding a signal such as an excitation signal in the second domain, and a third domain decoder such as an LPC-spectral decoder branch and two cascaded switches for switching between the decoders. | 08-18-2011 |
| 20110202355 | Audio Encoding/Decoding Scheme Having a Switchable Bypass - An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate. | 08-18-2011 |
| 20110202356 | Methods and Apparatuses for Encoding and Decoding Object-Based Audio Signals - An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal, which is obtained by downmixing a plurality of object signals, and object side information, extracting metadata from the object-side information and displaying an information regarding the object signals based on the metadata. | 08-18-2011 |
| 20110202357 | Methods and Apparatuses for Encoding and Decoding Object-Based Audio Signals - An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal, which is obtained by downmixing a plurality of object signals, and object side information, extracting metadata from the object-side information and displaying an information regarding the object signals based on the metadata. | 08-18-2011 |
| 20110208528 | SIGNAL CLIPPING PROTECTION USING PRE-EXISTING AUDIO GAIN METADATA - The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values ( | 08-25-2011 |
| 20110224991 | 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. | 09-15-2011 |
| 20110224992 | SET-TOP-BOX WITH INTEGRATED ENCODER/DECODER FOR AUDIENCE MEASUREMENT - Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider. | 09-15-2011 |
| 20110224993 | APPARATUS AND METHOD FOR PROCESSING MULTI-CHANNEL AUDIO SIGNAL USING SPACE INFORMATION - An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal. | 09-15-2011 |
| 20110224994 | Energy Conservative Multi-Channel Audio Coding - The invention relates to the technical field of audio encoding and/or decoding technologies, and thus concerns an overall encoding procedure and associated decoding procedure. The encoding procedure involves at least two signal encoding processes (S | 09-15-2011 |
| 20110224995 | CODING WITH NOISE SHAPING IN A HIERARCHICAL CODER - A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided. | 09-15-2011 |
| 20110231195 | HIGH-FREQUENCY BANDWIDTH EXTENSION IN THE TIME DOMAIN - A system extends the high-frequency spectrum of a narrowband audio signal in the time domain. The system extends the harmonics of vowels by introducing a non linearity in a narrow band signal. Extended consonants are generated by a random-noise generator. The system differentiates the vowels from the consonants by exploiting predetermined features of a speech signal: | 09-22-2011 |
| 20110231196 | DUAL-MODE ENCODER, SYSTEM INCLUDING SAME, AND METHOD FOR GENERATING INFRA-RED SIGNALS - A dual-mode encoder. The encoder includes a logic device. The logic device includes a first input terminal for receiving a signal associated with an audio source, a second input terminal for receiving a mode instruction signal, and an output terminal for outputting an encoded signal. The logic device is configured to operate in a first mode by encoding the received associated signal according to a first protocol, and to operate in a second mode by encoding the received associated signal according to a second protocol. | 09-22-2011 |
| 20110238424 | Method and apparatus for encoding and decoding excitation patterns from which the masking levels for an audio signal encoding and decoding are determined - For the quantisation of spectral data in an audio transform encoder psycho-acoustic information is required, i.e. an approximation of the true masking threshold. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window/transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied. | 09-29-2011 |
| 20110238425 | Multi-Resolution Switched Audio Encoding/Decoding Scheme - An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter. | 09-29-2011 |
| 20110238426 | Audio Decoder, Audio Encoder, Method for Decoding an Audio Signal, Method for Encoding an Audio Signal, Computer Program and Audio Signal - An audio decoder for providing a decoded audio information on the basis of an entropy encoded audio information includes a context-based entropy decoder configured to decode the entropy-encoded audio information in dependence on a context, which context is based on a previously-decoded audio information in a non-reset state-of-operation. The context-based entropy decoder is configured to select a mapping information, for deriving the decoded audio information from the encoded audio information, in dependence on the context. The context-based entropy decoder includes a context resetter configured to reset the context for selecting the mapping information to a default context, which default context is independent from the previously-decoded audio information, in response to a side information of the encoded audio information. | 09-29-2011 |
| 20110238427 | SIGNAL CLASSIFICATION PROCESSING METHOD, CLASSIFICATION PROCESSING DEVICE, AND ENCODING SYSTEM - A signal classification processing method, a classification processing device, and an encoding system are provided. The signal classification processing method includes: obtaining a high band input signal; determining a signal type of the obtained high band input signal according to a time domain characteristic parameter and/or a frequency domain characteristic parameter of the high band input signal; and determining an encoding mode corresponding to the signal type. The classification processing device includes: a receiving unit, configured to obtain a high band input signal; and a processing unit, configured to determine a signal type of the obtained high band input signal according to a time domain characteristic parameter and/or a frequency domain characteristic parameter of the high band input signal and determine an encoding mode corresponding to the signal type. An encoding system is also provided. Therefore, type subdivision and processing are performed on the high band input signal, so as to facilitate encoding and decoding processing of the signal. | 09-29-2011 |
| 20110246205 | METHOD FOR DETECTING AUDIO SIGNAL TRANSIENT AND TIME-SCALE MODIFICATION BASED ON SAME - A method for detecting a transient in an audio signal that has been broken up into frames includes obtaining a time domain feature of the frames and comparing the domain feature with a predetermined value. If the time domain feature is greater than the predetermined value, the frames are taken as transient and if the time domain feature is less than the predetermined value, the frames are taken as non-transient. The method has a low computational intensity and is thus very suitable for devices with limited processing resources. | 10-06-2011 |
| 20110246206 | AUDIO DECODING SYSTEM AND AN AUDIO DECODING METHOD THEREOF - An audio decoding system including a decoder decoding a first part of audio data, and an audio buffer compressor compressing and storing the decoded first part of audio data in a first time interval and decompressing the stored first part of audio data in a second time interval. | 10-06-2011 |
| 20110246207 | APPARATUS FOR PLAYING AND PRODUCING REALISTIC OBJECT AUDIO - Disclosed is an apparatus for playing and producing realistic object audio. The apparatus for playing realistic object audio includes: a deformatter unit individually separating scene description (SD) compression data and object audio compression data from inputted audio files; an SD decoding unit decoding the SD compression data to restore SD information; an object audio decoding unit decoding the object audio compression data to restore object audio signals which are respective audio signals of a plurality of objects; and an object audio effect unit adding an audio effect for each object to the object audio signals according to SD information for each object corresponding to the object audio signals among the SD information to produce a realistic object audio signal corresponding to each of the object audio signals. | 10-06-2011 |
| 20110251846 | Transient Signal Encoding Method and Device, Decoding Method and Device, and Processing System - A transient signal encoding method and device, decoding method and device, and processing system, where the transient signal encoding method includes: obtaining a reference sub-frame where a maximal time envelope having a maximal amplitude value is located from time envelopes of all sub-frames of an input transient signal; adjusting an amplitude value of the time envelope of each sub-frame before the reference sub-frame in such a way that a first difference is greater than a preset first threshold, in which the first difference is a difference between the amplitude value of the time envelope of each sub-frame before the reference sub-frame and the amplitude value of the maximal time envelope; and writing the adjusted time envelope into bitstream. | 10-13-2011 |
| 20110257978 | Time Series Filtering, Data Reduction and Voice Recognition in Communication Device - A computer implemented method for processing audio data communicated between a first device and a second device over a data communication network, where one or more processors are programmed to perform steps include at a first device: receiving time series audio data comprising audio data over a time period; partitioning the audio data in a plurality of time segments; transforming the audio data in the plurality of time segments into a plurality of feature values; transmitting a subset of plurality of feature values over a data communication network; and at a second device: receiving the transmitted plurality of feature values from the data communication network; and transforming the feature values into the time domain to reproduce the time series audio data. | 10-20-2011 |
| 20110257979 | Time/Frequency Two Dimension Post-processing - In accordance with an embodiment, a time-frequency post-processing method of improving perceptual quality of a decoded audio signal, the method includes determining a time-frequency representation (such as filter bank analysis and synthesis) of an audio signal, estimating a time-frequency energy distribution of an audio signal from a time-frequency filter bank, computing a modification gain for each time-frequency representation point to have a modified time-frequency representation, and outputting audio signal from a modified time-frequency representation. | 10-20-2011 |
| 20110257980 | Bandwidth Extension System and Approach - A method of performing BandWidth Extension (BWE) includes a frequency band shifting approach to generate an extended high band signal in time domain and a gain determination approach of controlling the energy of the extended high band. The proposed approach allows shifting any size of low band to any size of high band. The BWE scaling gain is estimated by using available filter bank coefficients with extremely low bit rate or without costing any bit, combining three possible gain factors. | 10-20-2011 |
| 20110257981 | LPC RESIDUAL SIGNAL ENCODING/DECODING APPARATUS OF MODIFIED DISCRETE COSINE TRANSFORM (MDCT)-BASED UNIFIED VOICE/AUDIO ENCODING DEVICE - Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP). | 10-20-2011 |
| 20110257982 | AUDIO SIGNAL LOUDNESS DETERMINATION AND MODIFICATION IN THE FREQUENCY DOMAIN - Methods of, apparatuses for, and computer readable media having instructions thereon that when executed cause carrying out methods of determining and modifying the perceived loudness of a frequency domain audio signal where the frequency resolution, and corresponding temporal coverage of the frequency domain information is not constant. The frequency (and thus temporal) resolution of the perceived loudness processing is maintained constant at the longest block size. One method includes a block combiner and a loudness modification interpolator. | 10-20-2011 |
| 20110264454 | Adaptive Transition Frequency Between Noise Fill and Bandwidth Extension - A method for spectrum recovery in spectral decoding of an audio signal, comprises obtaining ( | 10-27-2011 |
| 20110264455 | METHODS, APPARATUS AND ARTICLES OF MANUFACTURE TO PERFORM AUDIO WATERMARK DECODING - Example methods, apparatus and articles of manufacture to perform audio watermark decoding are disclosed. A disclosed example method includes receiving an audio signal including an audience measurement code embedded therein using a first plurality of frequency components, sampling the audio signal, transforming the sampled audio signal into a first frequency domain representation, determining whether the code is detectable in the first plurality of frequency components of the first frequency domain representation, and when the code is not detected in the first plurality of frequency components, examining a second plurality of frequency components of a second frequency domain representation to determine whether the code is detected, the second plurality of frequency components being offset from the first plurality of frequency components by a first offset, the first offset corresponding to a sampling frequency mismatch. | 10-27-2011 |
| 20110264456 | BINAURAL RENDERING OF A MULTI-CHANNEL AUDIO SIGNAL - Binaural rendering a multi-channel audio signal into a binaural output signal is described. The multi-channel audio signal has a stereo downmix signal into which a plurality of audio signals are downmixed, and side information having a downmix information, as well as object level information of the plurality of audio signals and inter-object cross correlation information. Based on a first rendering prescription, a preliminary binaural signal is computed from the first and second channels of the stereo downmix signal. A decorrelated signal is generated as an perceptual equivalent to a mono downmix of the first and second channels of the stereo downmix signal being, however, decorrelated to the mono downmix. Depending on a second rendering prescription, a corrective binaural signal is computed from the decorrelated signal and the preliminary binaural signal is mixed with the corrective binaural signal to obtain the binaural output signal. | 10-27-2011 |
| 20110264457 | ENCODER, DECODER, ENCODING METHOD, AND DECODING METHOD - An encoding apparatus and method for generating low-frequency-band encoding information and high-frequency-band encoding information from an original signal. The encoding apparatus includes a first spectrum calculator that calculates a first spectrum of a low frequency band from a decoded signal of the low-frequency-band encoding information, a second spectrum calculator that calculates a second spectrum from the original signal, an estimator that divides a high frequency band of the second spectrum into a plurality of bands and estimates the second spectrum included in each band, using the first spectrum, and a first error component encoder that encodes a first error component between the high frequency band of the second spectrum and an estimated spectrum. A corresponding decoding apparatus and method provides decoding. | 10-27-2011 |
| 20110270616 | SPECTRAL NOISE SHAPING IN AUDIO CODING BASED ON SPECTRAL DYNAMICS IN FREQUENCY SUB-BANDS - A technique of spectral noise shaping in an audio coding system is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. The tonality of each sub-band is determined. If a sub-band is tonal, time domain linear prediction (TDLP) processing is applied to the sub-band, yielding a residual signal and linear predictive coding (LPC) coefficients of an all-pole model representing the sub-band signal. The residual signal is further processed using a frequency domain linear prediction (FDLP) method. The FDLP parameters and LPC coefficients are transferred to a decoder. At the decoder, an inverse-FDLP process is applied to the encoded residual signal followed by an inverse TDLP process, which shapes the quantization noise according to the power spectral density of the original sub-band signal. Non-tonal sub-band signals bypass the TDLP process. | 11-03-2011 |
| 20110282674 | MULTICHANNEL AUDIO CODING - An encoder for encoding an audio signal comprising at least two channels configured to determine at least one audio signal image position value for the at least two channels of the audio signal; and calculate at least one audio signal image gain value associated with the at least one audio signal image position value. | 11-17-2011 |
| 20110282675 | Apparatus and Method for Generating a Synthesis Audio Signal and for Encoding an Audio Signal - An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal. The spectral domain patch generator is furthermore configured to select a first spectral domain patching algorithm from the plurality of patching algorithms for a first time portion and a second spectral domain patching algorithm from the plurality of patching algorithm for a second different time portion in accordance with the patching control signal to obtain the modified spectral representation. | 11-17-2011 |
| 20110282676 | Method and System for Dual Mode Subband Acoustic Echo Canceller with Integrated Noise Suppression - Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal. | 11-17-2011 |
| 20110282677 | METHOD AND SYSTEM FOR REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES AND GENERAL PURPOSE AUDIO CODEC - A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms. | 11-17-2011 |
| 20110288872 | STEREO ACOUSTIC SIGNAL ENCODING APPARATUS, STEREO ACOUSTIC SIGNAL DECODING APPARATUS, AND METHODS FOR THE SAME - Disclosed is a stereo acoustic signal encoding apparatus in which the signal quality does not deteriorate if there are a plurality of sound sources. A peak tracing unit ( | 11-24-2011 |
| 20110288873 | AUDIO ENCODER AND BANDWIDTH EXTENSION DECODER - An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter fulfils a predefined criterion. | 11-24-2011 |
| 20110295608 | METHODS FOR IMPROVING HIGH FREQUENCY RECONSTRUCTION - The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit. | 12-01-2011 |
| 20110301960 | CODING APPARATUS, CODING METHOD, DECODING APPARATUS, DECODING METHOD, AND PROGRAM - A coding apparatus includes a generation unit configured to generate first coding information used for first coding of a first audio signal and second coding information used for second coding of a second audio signal, and generate third coding information used for the first coding of the second audio signal and fourth coding information used for the second coding of a third audio signal; a first coding unit configured to generate first data and second data; a second coding unit configured to generate third data and fourth data by performing the second coding on the third audio signal; and a multiplexing unit configured to generate a stream of the first audio signal and a stream of the second audio signal. The third data is decoded in place of the second data in a case where a loss or an error has occurred in the stream of the second audio signal. | 12-08-2011 |
| 20110301961 | METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO SIGNAL USING ADAPTIVE SINUSOIDAL CODING - A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently. | 12-08-2011 |
| 20110307261 | Quantizing a Joint-Channel-Encoded Audio Signal - Provided are, among other things, systems, methods and techniques for quantizing a joint-channel-encoded audio signal, e.g., by: (a) obtaining an audio signal that includes a plurality of channels, with each channel including a block of samples; (b) segmenting the samples within each of a plurality of the blocks into quantization units; (c) jointly sum/difference encoding at least one pair of corresponding quantization units in different channels to produce a sum channel quantization unit and a difference channel quantization unit; (d) initializing quantization step sizes among the quantization units across the plurality of channels; (e) quantizing the samples in the quantization units using the assigned quantization step sizes; (f) calculating quantization errors for the quantization units; (g) based on the quantization errors, identifying a target quantization unit, from among the quantization units, for reduction of quantization step size; (h) determining whether the target quantization unit has been jointly sum/difference encoded with another quantization unit; (i) if the target quantization unit has not been jointly sum/difference encoded with another quantization unit, re-quantizing the target quantization unit using a decreased quantization step size; (j) if the target quantization unit has been jointly sum/difference encoded with another quantization unit, then: (i) designating the sum channel quantization unit as a target S/D quantization unit if the sum channel quantization unit has a greater quantization error than the difference channel quantization unit, (ii) designating the difference channel quantization unit as the target S/D channel quantization unit if the difference channel quantization unit has a greater quantization error than the sum channel quantization unit, and (iii) re-quantizing the target S/D channel quantization using a decreased quantization step size; (k) recalculating the quantization error for the target quantization unit; and (l) repeating (g)-(k) until a specified criterion is satisfied. | 12-15-2011 |
| 20110313777 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR OBTAINING A PARAMETER DESCRIBING A VARIATION OF A SIGNAL CHARACTERISTIC OF A SIGNAL - An apparatus for obtaining a parameter describing a variation of a signal characteristic of a signal on the basis of actual transform-domain parameters describing the audio signal in transform-domain includes a parameter determinator. The parameter determinator is configured to determine one or more model parameters of a transform-domain variation model describing an evolution of the transform-domain parameters in dependence on one or more model parameters representing a signal characteristic, such that a model error, representing a deviation between a modeled temporal evolution of the transform-domain parameters and an evolution of the actual transform-domain parameters, is brought below a predetermined threshold value or minimized. | 12-22-2011 |
| 20110313778 | 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. | 12-22-2011 |
| 20110320209 | FREQUENCY DOMAIN MULTIBAND DYNAMICS COMPRESSOR WITH AUTOMATICALLY ADJUSTING FREQUENCY BAND BOUNDARY LOCATIONS - A multiband dynamics compressor implements a frequency-domain solution for addressing unwanted magnitude peaks which may occur at the crossover frequency (boundary) between two adjacent frequency bands. The solution proposes making slight adjustments to the frequency band boundary locations, for example on a frame-by-frame basis, in order to prevent a spectral peak in the input signal from being located midway between two frequency bands. The adjustment to the boundary location pushes the energy of the spectral peak substantially into one frequency band for compression. | 12-29-2011 |
| 20110320210 | MULTIBAND DYNAMICS COMPRESSOR WITH SPECTRAL BALANCE COMPENSATION - A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band. | 12-29-2011 |
| 20110320211 | METHOD AND APPARATUS FOR PROCESSING SIGNAL - A method and an apparatus for processing a signal are provided. The method includes: obtaining an energy average value of each sub-band for a current frame frequency-domain signal; obtaining a current frame modification coefficient of each sub-band for the current frame frequency-domain signal according to a spectral envelope and the energy average value of each sub-band; obtaining a weighted modification coefficient of each sub-band for the current frame frequency-domain signal by using the current frame modification coefficient and a relevant frame modification coefficient; and modifying the spectral envelope of each sub-band for the current frame frequency-domain signal by using the weighted modification coefficient. | 12-29-2011 |
| 20110320212 | AUDIO SIGNAL ENCODING METHOD, AUDIO SIGNAL DECODING METHOD, ENCODING DEVICE, DECODING DEVICE, AUDIO SIGNAL PROCESSING SYSTEM, AUDIO SIGNAL ENCODING PROGRAM, AND AUDIO SIGNAL DECODING PROGRAM - When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized. | 12-29-2011 |
| 20110320213 | TIME-WARPING OF DECODED AUDIO SIGNAL AFTER PACKET LOSS - A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain. | 12-29-2011 |
| 20110320214 | DUAL STREAMING WITH EXCHANGE OF FEC STREAMS BY AUDIO SINKS - A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links. | 12-29-2011 |
| 20120004918 | Full-Band Scalable Audio Codec - A scalable audio codec for a processing device determines first and second bit allocations for each frame of input audio. First bits are allocated for a first frequency band, and second bits are allocated for a second frequency band. The allocations are made on a frame-by-frame basis based on the energy ratio between the two bands. For each frame, the codec transform codes both frequency bands into two sets of transform coefficients, which are then packetized based on the bit allocations. The packets are then transmitted with the processing device. Additionally, the frequency regions of the transform coefficients can be arranged in order of importance determined by power levels and perceptual modeling. Should bit stripping occur, the decoder at a receiving device can produce audio of suitable quality given that bits have been allocated between the bands and the regions of transform coefficients have been ordered by importance. | 01-05-2012 |
| 20120004919 | THREE-DIMENSIONAL GLASSES WITH BLUETOOTH AUDIO DECODE - Audio associated with three-dimensional image content is enabled to be heard by a user without interfering with other users. A device network includes a display system (master) and a wearable device (slave). The wearable device includes a glasses frame, earphones, and left and right eye shuttering lenses. The wearable device receives from the display system a frame sync signal and audio content associated with three-dimensional image content displayed by the display system as alternating left and right images. The audio content is played using the earphone. The left and right eye shuttering lens are shuttered in synchronism with the alternating left and right images according to the frame sync signal to enable a wearer of the wearable device to perceive the alternating left and right images as a three-dimensional image. Additional wearable devices may join the device network to be delivered independent audio and three-dimensional content in a similar manner. | 01-05-2012 |
| 20120004920 | DATA EMBEDDING SYSTEM - A data hiding system is described for hiding data within an audio signal. The system can be used for watermarking, data communications, audience surveying etc. The system hides data in an audio signal by adding artificial echoes whose polarity varies with the data to be hidden. In one embodiment, each data value is represented by a positive and a negative echo having different delays. A receiver can then remove the effects of natural echoes and/or periodicities in the audio signal by differencing measurements obtained at the different delays. | 01-05-2012 |
| 20120004921 | METHOD FOR RETRIEVING AUDIO SIGNAL STORED ON PHOTOGRAPH - A method of decoding coded data provided on a photograph using a reader includes steps of irradiating an image-side of the photograph from a first end thereof to an opposing second end thereof with infra-red illumination; receiving infra-red illumination reflected from the image; processing the reflected infra-red illumination to locate within the photograph a first target boundary; processing the reflected infra-red illumination received after location of the first target boundary to obtain audio data; storing the audio data in a memory; processing the reflected infra-red illumination to locate within the image a second target boundary downstream of the first target boundary; ceasing the processing of the reflected infra-red illumination to obtain coded data upon location of the second target boundary; and decoding the audio data stored in the memory to obtain an audio signal. | 01-05-2012 |
| 20120010891 | APPARATUS AND METHOD FOR ENCODING/DECODING MULTICHANNEL SIGNAL - An apparatus and method for encoding/decoding a multi-channel signal may be provided. The apparatus of encoding a multi-channel signal may insert information about whether to encode a phase parameter indicating phase information of a plurality of channels, included in the multi-channel signal, in a bitstream of the multi-channel signal. The apparatus of decoding a multi-channel signal may determine whether to up-mix a mono signal using the phase parameter based on the information about whether to encode. | 01-12-2012 |
| 20120016679 | ADAPTING MASKING THRESHOLDS FOR ENCODING AUDIO DATA - According to one embodiment, an improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal. | 01-19-2012 |
| 20120016680 | AUDIO DECODER AND DECODING METHOD USING EFFICIENT DOWNMIXING - A method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M| 01-19-2012 | |
| 20120022877 | Dynamic Range Improvement Technique - Apparatus and methods are disclosed for detecting and progressively attenuating specific frequencies prevalent in an audio signal. In contrast to conventional wide-band enhancement techniques over long time frames, narrow bandwidths and short attenuation times employed are commensurate with resonances and timing typical of speech. Apparent dynamic range is therefore increased through attenuation of longer-duration elements with declining informational contribution. | 01-26-2012 |
| 20120022878 | SIGNAL DE-NOISING METHOD, SIGNAL DE-NOISING APPARATUS, AND AUDIO DECODING SYSTEM - In the field of audio encoding/decoding technologies, a signal de-noising method is provided. The method includes: selecting, according to a degree of inter-frame correlation of a frame where a spectral coefficient to be adjusted resides, at least two spectral coefficients having high correlation with the spectral coefficient to be adjusted; performing weighting on the at least two selected spectral coefficients and the spectral coefficient to be adjusted to acquire a predicted value of the spectral coefficient to be adjusted; and adjusting a spectrum of a decoded signal by using the acquired predicted value, and outputting the adjusted decoded signal. A signal de-noising apparatus corresponding to the signal de-noising method and an audio decoding system using the signal de-noising apparatus are also provided. | 01-26-2012 |
| 20120022879 | METHODS AND APPARATUS FOR EMBEDDING CODES IN COMPRESSED AUDIO DATA STREAMS - Methods and apparatus for embedding codes in compressed audio data streams are disclosed. An example apparatus disclosed herein to embed a code in a compressed audio data stream comprises an unpacking unit to determine a plurality of transform coefficients associated with the compressed audio data stream, the plurality of transform coefficients being represented by a respective plurality of mantissas and a respective plurality of scale factors, and an embedding unit to modify a mantissa in the plurality of mantissas and a corresponding scale factor in the plurality of scale factors to embed the code in the compressed audio data stream. | 01-26-2012 |
| 20120029923 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR CODING OF HARMONIC SIGNALS - A scheme for coding a set of transform coefficients that represent an audio-frequency range of a signal uses a harmonic model to parameterize a relationship between the locations of regions of significant energy in the frequency domain. | 02-02-2012 |
| 20120029924 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR MULTI-STAGE SHAPE VECTOR QUANTIZATION - A multistage shape vector quantizer architecture uses information from a selected first-stage codebook vector to generate a rotation matrix. The rotation matrix is used to rotate the direction of the input vector to support shape quantization of the first-stage quantization error. | 02-02-2012 |
| 20120029925 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR DYNAMIC BIT ALLOCATION - A dynamic bit allocation operation determines a bit allocation for each of a plurality of vectors, based on a corresponding plurality of gain factors, and compares each allocation to a threshold value that is based on a dimensionality of the vector. | 02-02-2012 |
| 20120029926 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR DEPENDENT-MODE CODING OF AUDIO SIGNALS - A scheme for coding a set of transform coefficients that represent an audio-frequency range of a signal uses information from a reference frame that describes a previous frame of the signal to determine frequency-domain locations of regions of significant energy in a target frame of the signal. | 02-02-2012 |
| 20120035936 | INFORMATION REUSE IN LOW POWER SCALABLE HYBRID AUDIO ENCODERS - A system and method of reusing information in low power scalable hybrid audio encoders. The system and method provides a transform coder and parameterization of high frequency spectrum (SBR). | 02-09-2012 |
| 20120035937 | DECODING METHOD AND DECODING APPARATUS THEREFOR - A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients. | 02-09-2012 |
| 20120035938 | AUDIO REPRODUCING METHOD, AUDIO REPRODUCING APPARATUS THEREFOR, AND INFORMATION STORAGE MEDIUM - An audio reproducing method for quickly and correctly extracting extra data, including: receiving a data stream including the extra data including an end marker disposed immediately before main data and data length information, which is length information of the extra data, disposed immediately before the end marker; checking the presence/absence of the end marker; and if the end marker exists, extracting the extra data by using the data length information. | 02-09-2012 |
| 20120035939 | METHOD OF PROCESSING SIGNAL, ENCODING APPARATUS THEREOF, DECODING APPARATUS THEREOF, AND SIGNAL PROCESSING SYSTEM - A method processing a signal, an encoding apparatus, and a decoding apparatus are provided. The method of processing a signal includes restoring a down-mixed original signal using a re-quantized prediction parameter to generate a restored signal in an encoding apparatus; generating mute information indicating whether the down-mixed original signal has been muted, according to a value of the restored signal; and transmitting the mute information and the down-mixed original signal from the encoding apparatus to a decoding apparatus. | 02-09-2012 |
| 20120035940 | AUDIO SIGNAL PROCESSING METHOD, ENCODING APPARATUS THEREFOR, AND DECODING APPARATUS THEREFOR - An audio signal processing method includes: receiving an audio signal comprising consecutive frames; generating a first encoding parameter corresponding to a first frame among the consecutive frames and a second encoding parameter corresponding to a second frame adjacent to the first frame; and generating at least one interpolated parameter based on the first encoding parameter and the second encoding parameter. | 02-09-2012 |
| 20120035941 | QUANTIZATION AND INVERSE QUANTIZATION FOR AUDIO - An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding. | 02-09-2012 |
| 20120046954 | EFFICIENT BEAT-MATCHED CROSSFADING - Methods and devices to enable efficient beat-matched, DJ-style crossfading are provided. For example, such a method may involve determining beat locations of a first audio stream and a second audio stream and crossfading the first audio stream and the second audio stream such that the beat locations of the first audio stream are substantially aligned with the beat locations of the second audio stream. The beat locations of the first audio stream or the second audio stream may be determined based at least in part on an analysis of frequency data unpacked from one or more compressed audio files. | 02-23-2012 |
| 20120046955 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR NOISE INJECTION - A scheme for injecting noise at uncoded elements of a spectrum is controlled according to a measure of a distribution of energy of the original spectrum among the locations of the uncoded elements. | 02-23-2012 |
| 20120046956 | UNIVERSAL CONTAINER FOR AUDIO DATA - Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent. | 02-23-2012 |
| 20120059659 | Codebook Segment Merging - Provided are, among other things, systems, methods and techniques for merging entropy codebook application ranges within an audio signal. According to one embodiment, an audio signal is obtained, the audio signal including quantization indexes, identification of segments of said quantization indexes, and indexes of entropy codebooks that have been assigned to such segments, with a single entropy codebook index having been assigned to each said segment; potential merging operations in which specified segments potentially would be merged with each other are identified; bit penalties are estimated for the potential merging operations; then, the potential merging operation having the lowest estimated the penalty is performed. | 03-08-2012 |
| 20120065983 | Efficient Combined Harmonic Transposition - The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular; a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described, The system may comprise an analysis filter bank ( | 03-15-2012 |
| 20120065984 | DECODING DEVICE AND DECODING METHOD - Provided is a decoding device that can reduce abrupt changes in the number of channels in a decoded signal when transmission errors occur as a result of lost frames in an encoding/decoding system for multichannel signals. Said decoding device is also capable of per-sample smoothing and can reduce degradation of audio quality. In the provided device, a demultiplexer ( | 03-15-2012 |
| 20120072225 | SYSTEMS AND METHODS FOR ENCODING AND DECODING - Systems and methods for encoding and decoding are disclosed. The systems and methods include multimedia decoder instantiation systems and multimedia processing engines which are capable of being upgraded or reconfigured to support a new or previously-unsupported compression format, without the need for platform-specific software or hardware upgrades. | 03-22-2012 |
| 20120072226 | PARCOR COEFFICIENT QUANTIZATION METHOD, PARCOR COEFFICIENT QUANTIZATION APPARATUS, PROGRAM AND RECORDING MEDIUM - On a criterion to minimize the entropy of the linear prediction residual of the input signal used for calculation of the input PARCOR coefficient sequence, PARCOR coefficients with larger absolute values are quantized with higher quantization precisions so as to reduce the increase of the code amount of the linear prediction residual caused by the quantization error of the PARCOR coefficients. If the PARCOR coefficient is represented by a value formed by a predetermined number of bits, the number of effective bits from the most significant bit toward the least significant bit included in the output value increases with the absolute value of the PARCOR coefficient. | 03-22-2012 |
| 20120078640 | AUDIO ENCODING DEVICE, AUDIO ENCODING METHOD, AND COMPUTER-READABLE MEDIUM STORING AUDIO-ENCODING COMPUTER PROGRAM - An audio encoding device includes, a time-frequency transformer that transforms signals of channels, a first spatial-information determiner that generates a frequency signal of a third channel, a second spatial-information determiner that generates a frequency signal of the third channel, a similarity calculator that calculates a similarity between the frequency signal of the at least one first channel and the frequency signal of the at least one second channel, a phase-difference calculator that calculates a phase difference between the frequency signal of the at least one first channel and the signal of the at least one second channel, a controller that controls determination of the first spatial information when the similarity and the phase difference satisfy a predetermined determination condition, a channel-signal encoder that encodes the frequency signal of the third channel, and a spatial-information encoder that encodes the first spatial information or the second spatial information. | 03-29-2012 |
| 20120078641 | COMPRESSION CODING AND DECODING METHOD, CODER, DECODER, AND CODING DEVICE - The embodiments of the present invention relate to a compression coding and decoding method, a coder, a decoder and a coding device. The compression coding method includes: extracting sign information of an input signal to obtain an absolute value signal of the input signal; obtaining a residual signal of the absolute value signal by using a prediction coefficient, where the prediction coefficient is obtained by prediction and analysis that are performed according to a signal characteristic of the absolute value signal of the input signal; and multiplexing the residual signal, the sign information and a coding parameter to output a coding code stream, after the residual signal, the sign information and the coding parameter are respectively coded, so as to improve compression efficiency of a voice and audio signal. | 03-29-2012 |
| 20120078642 | ENCODING METHOD AND ENCODING DEVICE, DECODING METHOD AND DECODING DEVICE AND TRANSCODING METHOD AND TRANSCODER FOR MULTI-OBJECT AUDIO SIGNALS - A method of encoding a multi-object audio signal and an encoding apparatus, a decoding method and a decoding apparatus, and a transcoding method and a transcoder are provided. A multi-object audio signal encoding apparatus may encode object signals obtained by excluding ForeGround Objects (FGOs) from a plurality of input object signals, and may encode the FGOs, thereby providing a listener with a satisfactory sound quality. | 03-29-2012 |
| 20120084089 | PROGRESSIVE ENCODING OF AUDIO - The present disclosure includes processing a signal to generate a first sub-set of data, transmitting the first sub-set of data for generation of a reconstructed audio signal, the reconstructed audio signal having a fidelity relative to the signal, processing the signal to generate a second sub-set of data and a third sub-set of data, the second sub-set of data defining a second portion of the signal and comprising data that is different than data of the first sub-set of data, and the third sub-set of data defining a third portion of the signal and comprising data that is different than data of the first and second sub-sets of data, comparing a priority of the second sub-set of data to a priority of the third sub-set of data, and transmitting one of the second sub-set of data and the third sub-set of data over the network for improving the fidelity. | 04-05-2012 |
| 20120095769 | AUDIO DECODING METHOD AND AUDIO DECODER - Embodiments of the present invention disclose an audio decoding method, including: determining that bitstreams to be decoded are monophony coding layer and first stereo enhancement layer bitstreams; decoding the monophony coding layer to obtain a monophony decoded frequency-domain signal; reconstructing left and right channel frequency-domain signals in a first sub-band region by utilizing the monophony decoded frequency-domain signal after an energy adjustment; and reconstructing left and right channel frequency-domain signals in a second sub-band region by utilizing the monophony decoded frequency-domain signal without the energy adjustment. | 04-19-2012 |
| 20120101824 | PITCH-BASED PRE-FILTERING AND POST-FILTERING FOR COMPRESSION OF AUDIO SIGNALS - Systems and methods for enhancing the quality of an audio signal produced by an audio codec are described herein. In accordance with the systems and methods, a pitch-based pre-filter adaptively filters an input audio signal to produce a filtered audio signal. An audio encoder encodes the filtered audio signal to generate a compressed audio bit stream. An audio decoder decodes the compressed audio bit stream to generate a decoded audio signal. A pitch-based post-filter adaptively filters the decoded audio signal to produce an output audio signal, wherein adaptively filtering the decoded audio signal comprises undoing at least part of a signal-shaping effect of the pitch-based pre-filter. | 04-26-2012 |
| 20120101825 | METHOD AND APPARATUS FOR ENCODING/DECODING AUDIO DATA WITH SCALABILITY - Method and apparatus for encoding/decoding audio data with scalability are provided. The method includes slicing audio data to correspond to a plurality of layers, obtaining scale band information and coding band information corresponding to each of the plurality of layers, coding additional information containing scale factor information and coding model information based on scale band information and coding band information corresponding to a first layer, obtaining quantized samples by quantizing audio data corresponding to the first layer with reference to the scale factor information, coding the obtained plurality of quantized samples in units of symbols in order from respective symbols formed with most significant bits (MSB) to least significant bits (LSB) based on the coding model information, and repeating, until coding for the plurality of layers is finished. Accordingly, fine grain scalability can be provided with lower complexity and better audio quality even in a lower layer. | 04-26-2012 |
| 20120101826 | DECOMPOSITION OF MUSIC SIGNALS USING BASIS FUNCTIONS WITH TIME-EVOLUTION INFORMATION - Decomposition of a multi-source signal using a basis function inventory and a sparse recovery technique is disclosed. | 04-26-2012 |
| 20120101827 | METHODS AND APPARATUS TO EXTRACT DATA ENCODED IN MEDIA CONTENT - Methods and apparatus to extract data encoded in media content are disclosed. An example method includes sampling a media content signal to generate digital samples, determining a frequency domain representation of the digital samples, determining a first rank of a first frequency in the frequency domain representation, determining a second rank of a second frequency in the frequency domain representation, combining the first rank and the second rank with a set of ranks to create a combined set of ranks, comparing the combined set of ranks to a set of reference sequences, determining a data represented by the combined set of ranks based on the comparison, and storing the data in a memory device. | 04-26-2012 |
| 20120109659 | Compensator and Compensation Method for Audio Frame Loss in Modified Discrete Cosine Transform Domain - The invention provides a compensation method for audio frame loss in a MDCT domain, the method comprising: when a frame currently lost is a P | 05-03-2012 |
| 20120116780 | ACOUSTIC SIGNAL PROCESSING SYSTEM, ACOUSTIC SIGNAL DECODING APPARATUS, PROCESSING METHOD IN THE SYSTEM AND APPARATUS, AND PROGRAM - The amount of computation in an acoustic signal decoding apparatus for a signal transform process from a frequency domain to a time domain is reduced while realizing the generation of appropriate output acoustic signals. | 05-10-2012 |
| 20120116781 | ENCODING APPARATUS, ENCODING METHOD, AND PROGRAM - An encoding apparatus includes a noise detector configured to detect noise included in a certain band in accordance with an audio signal, a gain controller configured to perform gain control on the audio signal so that components in the certain band of the audio signal are attenuated when the noise is detected by the noise detector, a bit allocation calculation unit configured to calculate the numbers of bits to be allocated to frequency spectra of the audio signal which have been subjected to the gain control performed by the gain controller in accordance with the frequency spectra, and a quantization unit configured to quantize the frequency spectra of the audio signal which have been subjected to the gain control in accordance with the numbers of the bits. | 05-10-2012 |
| 20120123787 | AUDIOAUDIO FORMAT CONVERTING APPARATUS AND AUDIOAUDIO FORMAT CONVERTING METHOD - According to one embodiment, there is provided a audio format converting apparatus including a audio data dividing unit, first to Nth audio format converting units, and a audio data connecting unit. The audio data dividing unit creates first to Nth divided audio streams from an input audio stream, and adds the same frames as a predetermined number of frames from the head of the (i+1)th divided audio stream to the end of an i-th divided audio stream (i=1,2, to N−1). The first to Nth audio format converting units subject the first to Nth divided audio streams to audio format converting processing in parallel, so as to produce first to Nth converted audio streams. The audio data connecting unit discards the predetermined number of frames from the head of each of the second to Nth converted audio streams, and sequentially connects the first to Nth converted audio streams. | 05-17-2012 |
| 20120123788 | CODING METHOD, DECODING METHOD, AND DEVICE AND PROGRAM USING THE METHODS - A high-quality decoded signal is synthesized. A coding method of the present invention includes a local decoding coefficient searching step. The local decoding coefficient searching step includes a replication determining sub-step, a candidate replication shift signal sequence generating sub-step, a distance calculating sub-step, and a minimum distance shift amount finding sub-step. The replication determining sub-step determines, for each source signal sequence to be coded, whether or not a candidate replication shift signal sequence is to be generated from a decoded signal sequence and outputs a replication determination flag. If the replication determination flag indicates that a candidate replication shift signal sequence is to be generated, the candidate replication shift signal sequence generating sub-step generates a candidate replication shift signal sequence for each predetermined candidate signal shift amounts. The distance calculating sub-step calculates a parameter representing the distance between predetermined signal sequences. The minimum distance shift amount finding step obtains a signal shift amount that minimizes the distance. | 05-17-2012 |
| 20120130721 | DIGITAL MEDIA UNIVERSAL ELEMENTARY STREAM - Described techniques and tools include techniques and tools for mapping digital media data (e.g., audio, video, still images, and/or text, among others) in a given format to a transport or file container format useful for encoding the data on optical disks such as digital video disks (DVDs). A digital media universal elementary stream can be used to map digital media streams (e.g., an audio stream, video stream or an image) into any arbitrary transport or file container, including optical disk formats, and other transports, such as broadcast streams, wireless transmissions, etc. The information to decode any given frame of the digital media in the stream can be carried in each coded frame. A digital media universal elementary stream includes stream components called chunks. An implementation of a digital media universal elementary stream arranges data for a media stream in frames, the frames having one or more chunks. | 05-24-2012 |
| 20120130722 | MULTIPLE DESCRIPTION AUDIO CODING AND DECODING METHOD, APPARATUS, AND SYSTEM - Embodiments of the present invention provide a multiple description audio coding and decoding method, apparatus, and system. The audio coding method includes: dividing residual signals indicating current audio signal information into multiple frequency band parts having different frequencies; respectively coding the multiple frequency band parts by using multiple description coding (MDC) methods with different speech quality; and combining each of description signal parts that are generated after coding is performed by using different MDC methods to form multiple description bit streams of the residual signals. According to the present invention, multiple description coding and decoding methods with different speech quality are used for different frequency bands, which reduces the bit rate of multiple description coding and decoding, improves the effect of multiple description coding and decoding, and hence enhances the quality of audio transmission. | 05-24-2012 |
| 20120136669 | TRANSCODING METHOD, APPARATUS, DEVICE AND SYSTEM - A method, a device, and a system for transcoding between two embedded codecs are disclosed. The method includes: delaying a first encoded stream in input streams for integer number of frames, where the first encoded stream includes a stream of at least one extension layer in the input streams obtained after input signals are encoded by using a first codec; and using the first codec to decode other encoded streams in the input streams to obtain the first decoded signal; and performing delay aligning and adjusting to obtain an adjusted signal so as to reduce the transcoding complexity and enhance quality of the transcoded signals. | 05-31-2012 |
| 20120136670 | BANDWIDTH EXTENSION METHOD, BANDWIDTH EXTENSION APPARATUS, PROGRAM, INTEGRATED CIRCUIT, AND AUDIO DECODING APPARATUS - To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum. | 05-31-2012 |
| 20120143612 | METHOD AND APPARATUS FOR AUDIO COMMUNICATION OF INFORMATION - A system that incorporates teachings of the present disclosure may include, for example, a controller configured to obtain information associated with media content, to generate a first group of tones representative of the information associated with the media content, and to generate a media stream comprising the media content and the first group of tones; and a communication interface configured to transmit the media stream to a media device whereby the media device presents the media content and a sequence of tones, where the sequence of tones is generated based at least in part on the first group of tones, where the first group of tones comprises high frequency tones and low frequency tones, and where one of the high and low frequency tones represents a binary one and the other of the high and low frequency tones represents a binary zero. Other embodiments are disclosed. | 06-07-2012 |
| 20120143613 | APPARATUS FOR PROVIDING ONE OR MORE ADJUSTED PARAMETERS FOR A PROVISION OF AN UPMIX SIGNAL REPRESENTATION ON THE BASIS OF A DOWNMIX SIGNAL REPRESENTATION, AUDIO SIGNAL DECODER, AUDIO SIGNAL TRANSCODER, AUDIO SIGNAL ENCODER, AUDIO BITSTREAM, METHOD AND COMPUTER PROGRAM USING AN OBJECT-RELATED PARAMETRIC INFORMATION - An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information includes a parameter adjuster. The parameter adjuster is configured to receive one or more input parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on the one or more input parameters and the object-related parametric information, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for input parameters deviating from optimal parameters by more than a predetermined deviation. | 06-07-2012 |
| 20120143614 | ENCODING APPARATUS, ENCODING METHOD, DECODING APPARATUS, DECODING METHOD, AND PROGRAM - An encoding apparatus includes a time-frequency transform unit that performs a time-frequency transform on an audio signal, a normalization unit that normalizes a frequency spectral coefficient obtained by the time-frequency transform in order to generate encoded data of the audio signal, a level calculation unit that calculates a level of the audio signal, a scale factor changing unit that changes a concealment scale factor included in encoded concealment data obtained by performing, on the basis of the level of the audio signal, a time-frequency transform and normalization on a minute noise signal, the concealment scale factor being a scale factor relating to a coefficient used for the normalization, and an output unit that outputs the encoded data of the audio signal generated by the normalization unit or outputs, as encoded data of the audio signal, the encoded concealment data whose concealment scale factor has been changed. | 06-07-2012 |
| 20120158408 | Method And Apparatus For Reducing Rendering Latency For Audio Streaming Applications Using Internet Protocol Communications Networks - A method and apparatus for reducing rendering latency in a terminal device which receives audio data from a communication network such as, for example, Voice over Internet Protocol (VoIP) communications networks. Received packets are advantageously decoded “immediately” upon receipt, and the decoded data is placed directly in the rendering buffer at a location corresponding to the time appropriate for rendering, without using any intermediate buffer. Then, in accordance with the principles of the present invention and more particularly in accordance with certain illustrative embodiments thereof, packet loss concealment (PLC) routines are advantageously applied preemptively, without first determining whether or not any subsequent packets have or have not been received by any particular time. | 06-21-2012 |
| 20120158409 | Bandwidth Extension Encoder, Bandwidth Extension Decoder and Phase Vocoder - A bandwidth extension encoder for encoding an audio signal has a signal analyzer, a core encoder and a parameter calculator. The audio signal has a low frequency signal having a core frequency band and a high frequency signal having an upper frequency band. The signal analyzer is configured for analyzing the audio signal, the audio signal having a block of audio samples, the block having a specified length in time. The signal analyzer is furthermore configured for determining from a plurality of analysis windows an analysis window to be used for performing a bandwidth extension in a bandwidth extension decoder. The core encoder is configured for encoding the low frequency signal to acquire an encoded or frequency signal. The parameter calculator is configured for calculating bandwidth extension parameters from the high frequency signal. | 06-21-2012 |
| 20120158410 | DIGITAL AUDIO SIGNAL PROCESSING SYSTEM - A digital audio signal processing system is disclosed that enable flexible coexistence of signals having different formats in the same hardware architecture. The system comprises at least one input, at least one first format transformer, and at least one digital audio signal processor. The at least one input is arranged to receive at least a first digital audio signal having a first format comprising a first symbol resolution and a first symbol distribution. The at least one first format transformer is arranged to transform the first digital audio signal to a second digital audio signal having a second format comprising a second symbol resolution which is different from the first symbol resolution and a second symbol distribution which is different from the first symbol distribution based on at least a first parameter and a second parameter, wherein the first parameter is associated with a number of integer symbols of the second format and the second parameter is associated with a number of fractional symbols of the second format. The at least one digital audio signal processor is arranged to process the second digital audio signal to produce a third digital audio signal. Corresponding computer program product is also disclosed. | 06-21-2012 |
| 20120158411 | SIGNAL ENCODING APPARATUS AND METHOD, SIGNAL DECODING APPARATUS AND METHOD, PROGRAMS AND RECORDING MEDIUMS - An encoding apparatus that divides an input time series signal into a plurality of sub-bands and encodes a low frequency sub-band signal to generate encoded data of the low frequency sub-band signal. Concurrently, it compares the frequency amplitude peak of the new high frequency sub-band signal generated from the low frequency sub-band signal and the original high frequency sub-band signal and generates frequency amplitude peak information of the high frequency sub-band signal. It compares the gain of the new high frequency sub-band signal generated by using the low frequency sub-band signal and the original high frequency sub-band signal and generates gain information of the high frequency sub-band signal. Subsequently, the signal encoding apparatus multiplexes the encoded data of the low frequency sub-band signal, the frequency amplitude peak information of the high frequency sub-band signal and the gain information of the high frequency sub-band signal and outputs compressed data. | 06-21-2012 |
| 20120166205 | INSTRUCTION EXECUTION CIRCUIT - An instruction execution circuit includes: a memory circuit including a first memory element and a second memory element configured to require less power than the first memory element; a processor; and an address decoder configured to output an enable signal to either one, storing an instruction, of the first memory element and the second memory element when an address is outputted from the processor, the enable signal corresponding to a signal to output the instruction stored at the address, one portion of a program stored in the first memory element corresponding to a portion in which processing other than loop processing is described, the loop processing causing the processor to execute the specific instruction in a repetitive manner, the other portion of a program stored in the second memory element corresponding to a portion in which the loop processing is described. | 06-28-2012 |
| 20120173246 | VARIABLE ORDER SHORT-TERM PREDICTOR - The present invention provides a new recursive FIR filter scheme which supports a variable order short-term predictor, and uses a pipeline stall based on the radix-2 algorithm and an autocorrelation processing time for reducing the complexity of MPEG-4 ALS hardware implementation. | 07-05-2012 |
| 20120173247 | APPARATUS FOR ENCODING AND DECODING AN AUDIO SIGNAL USING A WEIGHTED LINEAR PREDICTIVE TRANSFORM, AND A METHOD FOR SAME - Disclosed is an apparatus for encoding and/or decoding an audio signal having a variable bit rate (VBR). A target bit rate is determined in accordance with characteristics of an audio signal, and a weighted linear predictive transform coding is performed in accordance with the determined target bit rate. | 07-05-2012 |
| 20120179474 | MOBILE TERMINAL AND METHOD FOR PROCESSING AUDIO DATA THEREOF - A mobile terminal and audio data processing method thereof are provided. The method includes preprocessing, at a control unit, audio data, sending the preprocessed audio data to an audio processing unit, decoding, at the audio processing unit, the preprocessed audio data, and outputting the decoded audio data. The method is capable of reducing power consumed when playing audio data. The method is advantageous when processing multichannel audio data and various sound effects. | 07-12-2012 |
| 20120179475 | REPRODUCING APPARATUS AND METHOD, AND RECORDING MEDIUM - A reproducing apparatus and method includes a reproducing unit to reproduce mainstream data and sub audio data separately added in the mainstream data, wherein the reproducing unit comprises a counter used in reproducing the sub audio data. Accordingly, it is possible to more naturally reproduce still image data, such as a browsable slide show, to which sub audio data is additionally included, thus preventing an interruption in reproduction of the sub audio data even during a forward or reverse play. | 07-12-2012 |
| 20120185255 | IMPROVED CODING/DECODING OF DIGITAL AUDIO SIGNALS - A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder. | 07-19-2012 |
| 20120185256 | ALLOCATION OF BITS IN AN ENHANCEMENT CODING/DECODING FOR IMPROVING A HIERARCHICAL CODING/DECODING OF DIGITAL AUDIO SIGNALS - A method of binary allocation in an enhancement coding/decoding for improving a hierarchical coding/decoding of digital audio signals, including a core coding/decoding in a first frequency band and a band extension coding/decoding in a second frequency band. For a predetermined number of bits to be allocated for the enhancement coding/decoding, a first number of bits is allocated to a coding/decoding for correcting the core coding/decoding in the first frequency band and according to a first mode of coding/decoding and a second number of bits is allocated to an enhancement coding/decoding for improving the extension coding/decoding in the second frequency band and according to a second mode of coding/decoding. Also provided are an allocation module implementing the method and a coder and decoder including this module. | 07-19-2012 |
| 20120185257 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 07-19-2012 |
| 20120185258 | DIGITAL AUDIO SIGNAL COMPRESSION METHOD AND APPARATUS - Compression of audio signal data is described herein. In various embodiments, the compression of each unit of the audio signal data includes the employment of a distribution substantially representative of a subblock of residual data of the unit of audio signal data, to reduce the amount of data having to be transmitted to transmit the unit of audio signal data to a recipient. | 07-19-2012 |
| 20120191462 | AUDIO SIGNAL PROCESSING DEVICE WITH ENHANCEMENT OF LOW-PITCH REGISTER OF AUDIO SIGNAL - An audio signal processing device is designed to enhance the low-pitch register of an audio signal by generating harmonics causing a missing fundamental effect with a light load of processing but without damaging an audio waveform. The audio signal processing device includes a filtering part (e.g. a band-pass filter configured of a high-pass filter and a low-pass filter) that extracts a low-pitch signal from an audio signal input thereto; a dynamic range compression part that compresses a dynamic range of the low-pitch signal by use of a time-variant gain relative to a peak of the low-pitch signal, which is detected via a peak hold operation using a predetermined time constant, thus producing a compressed signal; and an adder that adds the compressed signal to the audio signal so as to produce a processed audio signal including harmonics. | 07-26-2012 |
| 20120197648 | AUDIO ANNOTATION - Embodiments provide methods, apparatuses, systems, and articles of manufacture for annotating and receiving inaudible audio annotations associated with audio content. The inaudible audio annotations may be identified by inaudible marker tones. The inaudible audio annotations and the inaudible marker tones may be included in the source file of the audio content. | 08-02-2012 |
| 20120197649 | Audio Coding - A method for encoding an audio signal including: processing a selected subset of a lower series of samples forming a lower frequency spectral band of the audio signal and a higher series of samples forming a higher frequency spectral band of the audio signal to parametrically encode the higher series of samples forming the higher frequency spectral band by identifying a sub-series of the lower series of samples. | 08-02-2012 |
| 20120197650 | METADATA TIME MARKING INFORMATION FOR INDICATING A SECTION OF AN AUDIO OBJECT - The application relates to a method for encoding time marking information within audio data. According to the method, time marking information is encoded as audio metadata within the audio data. The time marking information indicates at least one section of an audio object encoded in the audio data. E.g. the time marking information may specify a start position and an end position of the section or only a start position. The at least one section may be a characteristic part of the audio object, which allows instant recognition by listening. The time marking information encoded in the audio data enables instantaneous browsing to a certain section of the audio object. The application further relates to a method for decoding the time marking information encoded in the audio data. | 08-02-2012 |
| 20120203560 | SYSTEMS AND METHODS FOR ENCODING CONTROL MESSAGES IN AN AUDIO BITSTREAM - An audio system including a first audio unit and a second audio unit coupled to the first audio unit through an audio bus. A first processor is coupled to the first audio unit. The first processor is configured to transmit bits comprising audio content to the second audio unit over the audio bus. The first processor is further configured to receive a control command selected from a plurality of control commands, and in response, interrupt the bits comprising audio content and send a preamble and a control message on the audio bus, wherein the control message corresponds to the control command. A second processor is coupled to the second audio unit. The second processor is configured to monitor the audio bus for a preamble, and if a preamble is detected, then process the control message and execute the corresponding control command. | 08-09-2012 |
| 20120203561 | DEVICES FOR ADAPTIVELY ENCODING AND DECODING A WATERMARKED SIGNAL - An electronic device configured for adaptively encoding a watermarked signal is described. The electronic device includes modeler circuitry that determines watermark data based on a first signal. The electronic device also includes coder circuitry coupled to the modeler circuitry. The coder circuitry determines a low priority portion of a second signal and embeds the watermark data into the low priority portion of the second signal to produce a watermarked second signal. | 08-09-2012 |
| 20120209614 | SHARED VIDEO-AUDIO PIPELINE - Techniques are disclosed that involve the processing of audio streams. For instance, a host processing platform may receive a content stream that includes an encoded audio stream. In turn, a graphics engine produces from it a decoded audio stream. This producing may involve the graphics engine performing various operations, such as an entropy decoding operation, an inverse quantization operation, and an inverse discrete cosine transform operation. In embodiments, the content stream may further include an encoded video stream. Thus the graphics engine may produce from it a decoded video stream. This audio and video decoding may be performed in parallel. | 08-16-2012 |
| 20120209615 | Efficient Multichannel Signal Processing by Selective Channel Decoding - An input signal conveying encoded information representing one or more audio channels is decoded by determining the configuration of channels represented by the encoded information, obtaining from the channel configuration a channel selection mask that specifies which of the one or more audio channels are to be decoded, extracting encoded information from the input signal, and decoding the extracted encoded information for those audio channels specified in the channel selection mask. | 08-16-2012 |
| 20120209616 | MULTIBAND COMPRESSOR - In a multiband compressor | 08-16-2012 |
| 20120215546 | Complexity Scalable Perceptual Tempo Estimation - The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit-stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity. | 08-23-2012 |
| 20120215547 | Threshold Crossing Detection - A signal processing apparatus comprises a signal path for a signal, the signal path comprising a signal processing stage. An auxiliary stage is coupled to an input of the signal processing stage for, in response to a signal in the signal path at the input of the signal processing stage, generating a control signal indicative of the time of a crossing of a first threshold by the signal in the signal path at an output of the signal processing stage by detecting a crossing by the signal of a second threshold established by the auxiliary stage. The second threshold is substantially equal to the first threshold. | 08-23-2012 |
| 20120221342 | DECODING APPARATUS AND DECODING METHOD - A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S | 08-30-2012 |
| 20120221343 | APPARATUS AND METHOD FOR ENCODING/DECODING A MULTICHANNEL SIGNAL - An apparatus for encoding/decoding a multichannel signal. The apparatus for encoding/decoding a multichannel signal processes phase parameters for phase information among a plurality of channels constituting the multichannel signal in consideration of the characteristics of the multichannel signal. The apparatus generates an encoded bit stream for the multichannel signal using the processed phase parameters and the mono signal extracted from the multichannel signal. | 08-30-2012 |
| 20120221344 | ENCODER APPARATUS, DECODER APPARATUS AND METHODS OF THESE - A coding apparatus is disclosed which can improve the quality of a decoded signal in a hierarchical coding (scalable coding) scheme in which a coding target band is selected in each hierarchy (layer). Coding apparatus ( | 08-30-2012 |
| 20120226504 | Method of distortion-free signal compression - An audio signal in which an audio signal is received as a stream of digital samples, each being a numerical value representing a sampled signal level. A first zero crossing point is identified and the received audio samples are stored until a second zero crossing point is identified, thereby storing a first half-wave of samples. The highest intensity sample is identified from the stored samples and this is compared against a predetermined threshold. All stored samples are scaled by an initial scaling factor so that the intensity of the highest intensity sample is not above this threshold. A second half-wave of samples is stored in which all samples of the second half-wave are below the threshold. All stored samples of the second half-wave are also scaled but by a modified scaling factor derived from a combination of the initial scaling factor and a decay factor. | 09-06-2012 |
| 20120226505 | HIERARCHICAL AUDIO CODING, DECODING METHOD AND SYSTEM - A hierarchical audio coding, decoding method and system are provided. The method includes dividing frequency domain coefficients of an audio signal after MDCT into a plurality of coding sub-bands, quantizing and coding amplitude envelope values of coding sub-bands; allocating bits to each coding sub-band of the core layer, quantizing and coding core layer frequency domain coefficients to obtain coded bits of core layer frequency domain coefficients; calculating the amplitude envelope value of each coding sub-band of the core layer residual signal; allocating bits to each coding sub-band of the extended layer, quantizing and coding the extended layer coding signal to obtain coded bits of the extended layer coding signal; multiplexing and packing amplitude value envelope coded bits of each coding sub-band composed by core layer and extended layer frequency domain coefficients, core layer frequency coefficients coded bits, and extended layer coding signal coded bits, then transmitting to the decoding end. | 09-06-2012 |
| 20120232908 | METHODS AND SYSTEMS FOR AVOIDING PARTIAL COLLAPSE IN MULTI-BLOCK AUDIO CODING - Embodiments are described of a multi-block coding scheme for an audio signal to prevent partial collapse conditions from causing pre-echo compression artifacts. An audio codec includes a segmentation component partitioning the audio signal into a plurality of tiles, wherein each tile comprises data from a particular segment of time and a particular set of frequencies of the audio signal; a band energy component determining an energy value for each tile corresponding to a signal component in a respective tile; an encoder flag tracking component marking a tile as not collapsed or collapsed based on the energy value in that tile; and a decoder flag tracking component filling all tiles marked as collapsed with pseudorandom noise at an estimated energy level. | 09-13-2012 |
| 20120232909 | METHOD AND SYSTEM FOR TWO-STEP SPREADING FOR TONAL ARTIFACT AVOIDANCE IN AUDIO CODING - Embodiments are directed to an audio coding scheme implemented in a codec that eliminates birdie artifacts generated by transform coding methods. A frequency coefficient spreading method invertibly rotates a spectrum of coefficient values based on a defined rotation angle, The rotated spectrum is then quantized, and the rotation operation is then reversed so that a previously sparse spectrum (i.e., one with few non-zero values) becomes one that has many non-zero values. The method arranges the coefficients for a particular partition into a linear array and computes a gain factor for the partition. A rotation angle of between 0 and π/4 for successive pairs of coefficients of the linear array based on the gain factor is then derived. One or more rotation operations are then applied to successive pairs of coefficients in the linear array using a specific rotation angle and a stride length for each rotation operation. | 09-13-2012 |
| 20120232910 | SYSTEM FOR DYNAMICALLY CREATING AND RENDERING AUDIO OBJECTS - Embodiments of systems and methods are described for providing backwards compatibility for legacy devices that are unable to natively render non-channel based audio objects. These systems and methods can also be beneficially used to produce a reduced set of audio objects for compatible object-based decoders with low computing resources. | 09-13-2012 |
| 20120232911 | OPTIMIZATION OF MP3 AUDIO ENCODING BY SCALE FACTORS AND GLOBAL QUANTIZATION STEP SIZE - 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. | 09-13-2012 |
| 20120239407 | SYSTEM AND METHOD FOR UTILIZING AUDIO ENCODING FOR MEASURING MEDIA EXPOSURE WITH ENVIRONMENTAL MASKING - An audio beacon system, apparatus and method for collecting information on a panelist's exposure to media. An audio beacon is configured as on-device encoding technology that is operative in a processing device (e.g., cell phone, PDA, PC) to enable the device to encode an environmental sound and transmit it for a predetermined period of time. The acoustically transmitted data is received and processed by a portable audience measurement device, such as Arbitron's Personal People Meter™ (“PPM”), or other specially equipped portable device to enable audience measurement systems to achieve higher levels of detail on panel member activity and greater association of measurement devices to their respective users. | 09-20-2012 |
| 20120239408 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, coding identification information indicating whether to apply a first coding scheme or a second coding scheme to a current frame; when the coding identification information indicates that the second coding scheme is applied to the current frame, receiving window type information indicating a particular window for the current frame, from among a plurality of windows; identifying that a current window is a long stop window based on the window type information, wherein the long stop window is followed by only long window of a following frame, wherein the long stop window includes a gentle long stop window and a steep long stop window; and, when the first coding scheme is applied to a previous frame, applying the gentle long stop window to the current frame, wherein: the gentle long stop window comprise an ascending line with first slope, the steep long stop window comprise an ascending line with second slope, and, the first slope is gentler than the second slope. | 09-20-2012 |
| 20120239409 | BIT-STREAM PROCESSING/TRANSMITTING AND/OR RECEIVING/PROCESSING METHOD, MEDIUM, AND APPARATUS - A method, medium, and apparatus hierarchically coding/decoding audio data, such as bit sliced arithmetic coding (BSAC), such that payloads of audio data and extension data can be grouped and interleaved according to priority so that some groups of the payloads are dropped, and the remainder of groups are transmitted. Therefore, extension data that is more important than a top layer of audio data, in terms of reproducing of original sounds, can be transmitted with priority. | 09-20-2012 |
| 20120245947 | MULTI-MODE AUDIO SIGNAL DECODER, MULTI-MODE AUDIO SIGNAL ENCODER, METHODS AND COMPUTER PROGRAM USING A LINEAR-PREDICTION-CODING BASED NOISE SHAPING - A multi-mode audio signal decoder has a spectral value determinator to obtain sets of decoded spectral coefficients for a plurality of portions of an audio content and a spectrum processor configured to apply a spectral shaping to a set of spectral coefficients in dependence on a set of linear-prediction-domain parameters for a portion of the audio content encoded in a linear-prediction mode, and in dependence on a set of scale factor parameters for a portion of the audio content encoded in a frequency-domain mode. The audio signal decoder has a frequency-domain-to-time-domain converter configured to obtain a time-domain audio representation on the basis of a spectrally-shaped set of decoded spectral coefficients for a portion of the audio content encoded in the linear-prediction mode and for a portion of the audio content encoded in the frequency domain mode. An audio signal encoder is also described. | 09-27-2012 |
| 20120253826 | TRANSMITTING DEVICE, TRANSMITTING METHOD, RECEIVING DEVICE, RECEIVING METHOD, PROGRAM, AND BROADCASTING SYSTEM - Disclosed herein is a transmitting device including an audio encoder and a transmitter. The audio encoder is configured to generate an encoded audio stream in which trigger information relating to control of an application program to be executed in conjunction with content in a receiving device is buried. The transmitter is configured to transmit the generated encoded audio stream to the receiving device. | 10-04-2012 |
| 20120253827 | ADVANCED ENCODING OF MUSIC FILES - Example embodiments allow for the creation, distribution, and use of flexible media formats. Example embodiments may allow individual content files to be rendered in multiple formats and versions. In addition, example embodiments may provide for granular rights management, which may allow users to access content files on a feature-by-feature basis. | 10-04-2012 |
| 20120259642 | AUDIO STREAM COMBINING APPARATUS, METHOD AND PROGRAM - A stream combining apparatus is provided, comprising an input unit that receives the input of group access units and group access units from two streams that are generated by overlap transform; a decoder that generates group frames by decoding the group access units and that generates group frames by decoding the group access units; and a combining unit that uses group frames and group frames as a frame of reference for the access units, that decodes the frames, that performs selective mixing to generate mixed frames, that encodes said mixed frames, that generates a prescribed number of group access units, and that joins two streams, using a prescribed number of group access units as a joint such that the access units adjacent to each other on the boundary between the two streams and a prescribed number of group access units are stitched so that the information for decoding the same common frames is distributed. | 10-11-2012 |
| 20120259643 | APPARATUS FOR PROVIDING AN UPMIX SIGNAL REPRESENTATION ON THE BASIS OF THE DOWNMIX SIGNAL REPRESENTATION, APPARATUS FOR PROVIDING A BITSTREAM REPRESENTING A MULTI-CHANNEL AUDIO SIGNAL, METHODS, COMPUTER PROGRAMS AND BITSTREAM REPRESENTING A MULTI-CHANNEL AUDIO SIGNAL USING A LINEAR COMBINATION PARAMETER - An apparatus for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information, which are included in a bitstream representation of an audio content, in independence on a user-specified rendering matrix, the apparatus has a distortion limiter configured to obtain a modified rendering matrix using a linear combination of a user-specified rendering matrix in a target rendering matrix in dependence on a linear combination parameter. The apparatus also has a signal processor configured to obtain the upmix signal representation on the basis of the downmix signal representation and the object-related parametric information using the modified rendering matrix. The apparatus is also configured to evaluate a bitstream element representing the linear combination parameter in order to obtain the linear combination parameter. | 10-11-2012 |
| 20120259644 | Audio-Encoding/Decoding Method and System of Lattice-Type Vector Quantizing - The audio coding method and system of lattice vector quantization is provided in the invention. The method comprises: dividing frequency domain coefficients of an audio signal for which a modified discrete cosine transform (MDCT) has been performed into a plurality of coding sub-bands, and quantizing and coding an amplitude envelope value of each coding sub-band to obtain coded bits of amplitude envelopes; performing bit allocation on each coding sub-band, and performing normalization, quantization and coding respectively on vectors in a low bit coding sub-band with pyramid lattice vector quantization and on vectors in a high bit coding sub-band with sphere lattice vector quantization to obtain coded bits of the frequency domain coefficients; multiplexing and packing the coded bits of the amplitude envelope and the coded bits of the frequency domain coefficients of each coding sub-band, then sending them to a decoding side. | 10-11-2012 |
| 20120259645 | 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. | 10-11-2012 |
| 20120265539 | CONTROL APPARATUS FOR AN ELECTRONIC DEVICE USING A BALANCED MICROPHONE CABLE - This invention enables a microphone to control the device having the microphone input. it includes both ideas related to the microphone as well as the device with the microphone input. It utilizes a standard two conductor shielded microphone cable and does not impact the audio from the microphone when the control operation is being performed. The microphone produces common-mode signals which are detected by the device with the microphone input. These signals provide control of the device with the microphone input. Likewise the device with the microphone input produces common-mode signals which are detected by the microphone. These signals provide control of the microphone visual display indicators. | 10-18-2012 |
| 20120265540 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AN AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A DETECTION OF A GROUP OF PREVIOUSLY-DECODED SPECTRAL VALUES - An audio decoder for providing a decoded audio information includes a arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state. The arithmetic decoder is configured to determine or modify the current context state in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder is configured to detect a group of a plurality of previously-decoded spectral values, which fulfill, individually or taken together, a predetermined condition regarding their magnitudes, and to determine the current context state in dependence on a result of the detection. | 10-18-2012 |
| 20120265541 | AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR PROVIDING AN ENCODED REPRESENTATION OF AN AUDIO CONTENT, METHOD FOR PROVIDING A DECODED REPRESENTATION OF AN AUDIO CONTENT AND COMPUTER PROGRAM FOR USE IN LOW DELAY APPLICATIONS - An audio signal encoder includes a transform-domain path which obtains spectral coefficients and noise-shaping information on the basis of a portion of the audio content, and which windows a time-domain representation of the audio content and applies a time-domain-to-frequency-domain conversion. The audio signal decoder includes a CELP path to obtain a code-excitation information and a LPD parameter information. A converter applies a predetermined asymmetric analysis window in both if a current portion is followed by a subsequent portion to be encoded in the transform-domain mode or in the CELP mode. Aliasing cancellation information is selectively provided in the later case. | 10-18-2012 |
| 20120265542 | OPTIMIZED PARAMETRIC STEREO DECODING - A method and decoder are provided for parametrically decoding a stereo digital audio signal. The method includes synthesizing the stereo signal, per frequency sub-band, on the basis of a decoded mono signal ({circumflex over (M)}[j]), arising from a downmix of the stereo signal and from spatial information parameters of the stereo signal, in such a way that the signals obtained have the following form: | 10-18-2012 |
| 20120265543 | MULTI-CHANNEL SIGNAL ENCODING AND DECODING METHOD, APPARATUS, AND SYSTEM - A multi-channel signal encoding method includes: determining a sum of channel level differences (CLDs) of a current frame of a multi-channel signal in a certain frequency band area; determining an average value of sums of channel level differences of at least two frames before the current frame in the certain frequency band area; according to the sum of channel level differences of the current frame of the multi-channel signal in the certain frequency band area, the average value of the sums of channel level differences of at least two frames before the current frame in the certain frequency band area, and a preset threshold, judging whether the channel level differences of the current frame are in a transient state or a non-transient state, and obtaining a judgment result; and according to the judgment result, performing quantization processing on the channel level differences of the current frame of the multi-channel signal. | 10-18-2012 |
| 20120271644 | AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL USING AN ALIASING-CANCELLATION - An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters. The transform domain path also includes a combiner configured to combine the time-domain representation of the audio content with an aliasing-cancellation synthesis signal to obtain an aliasing reduced time-domain signal. | 10-25-2012 |
| 20120278085 | METHOD AND A DECODER FOR ATTENUATION OF SIGNAL REGIONS RECONSTRUCTED WITH LOW ACCURACY - The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics. | 11-01-2012 |
| 20120278086 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AN AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A REGION-DEPENDENT ARITHMETIC CODING MAPPING RULE - An audio decoder for providing a decoded audio information includes an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using decoded spectral values. The arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state. The arithmetic decoder is configured to determine a numeric current context value describing the current context state in dependence on a plurality of previously decoded spectral values and also in dependence on whether a spectral value to be decoded is in a first predetermined frequency region or in a second predetermined frequency region. | 11-01-2012 |
| 20120278087 | MULTIBAND COMPRESSOR AND METHOD OF ADJUSTING THE SAME - Disclosed is a multiband compressor for applying dynamic compression to each of a plurality of bands of an audio signal that has been divided into frequency bands, then applying additive synthesis and outputting thereof; wherein power that is input to or output from each of the plurality of the bands is measured, and frequency bands are set on the basis of differences in power between each of the neighboring bands. Alternatively, power that is input to each of the plurality of bands is measured, a predetermined relative threshold is added to values for which a measured voltage has been smoothed, and a threshold to be given for dynamic compression processing for each of the plurality of bands is set. As a result, a multi-band compressor is provided for which an appropriate volume increase effect can be obtained regardless of what kind of signal is input. | 11-01-2012 |
| 20120278088 | Subband Block Based Harmonic Transposition - The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank ( | 11-01-2012 |
| 20120278089 | ERROR CONCEALMENT METHOD AND APPARATUS FOR AUDIO SIGNAL AND DECODING METHOD AND APPARATUS FOR AUDIO SIGNAL USING THE SAME - An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme. | 11-01-2012 |
| 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 |
| 20120284033 | METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS - The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. | 11-08-2012 |
| 20120284034 | METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS - The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. | 11-08-2012 |
| 20120290305 | Scalable Audio in a Multi-Point Environment - Use of a scalable audio codec to implement distributed mixing and/or sender bit rate regulation in a multipoint conference is disclosed. The scalable audio codec allows the audio signal from each endpoint to be split into one or more frequency bands and for the transform coefficients within such bands to be prioritized such that usable audio may be decoded from a subset of the entire signal. The subset may be created by omitting certain frequency bands and/or by omitting certain coefficients within the frequency bands. By providing various rules for each endpoint in a conference, the endpoint can determine the importance of its signal to the conference and can select an appropriate bit rate, thereby conserving bandwidth and/or processing power throughout the conference. | 11-15-2012 |
| 20120290306 | HYBRID CODED AUDIO DATA STREAMING APPARATUS AND METHOD - An audio coding system in which a plurality of quantization methods are selectable for application to components of a streamed audio signal to achieve a target frame size that is determined by comparing an achieved bit rate against a target bit rate. Based on the target frame size, the system calculates a bit allocation for signal components and compares the bit allocation to the dynamic range of the signal components. Depending on the outcome of the comparison, the system may select to quantize or not quantize a signal component. The system employs lossless coding techniques, but is capable of introducing lossy coding by quantization in order to meet the target bit rate. | 11-15-2012 |
| 20120290307 | BIT ALLOCATING, AUDIO ENCODING AND DECODING - A bit allocating method is provided that includes determining the allocated number of bits in decimal point units based on each frequency band so that a Signal-to-Noise Ratio (SNR) of a spectrum existing in a predetermined frequency band is maximized within a range of the allowable number of bits for a given frame; and adjusting the allocated number of bits based on each frequency band. | 11-15-2012 |
| 20120296656 | ADAPTIVE CONTROLLER FOR A CONFIGURABLE AUDIO CODING SYSTEM - An adaptive controller for a configurable audio coding system comprising a fuzzy logic controller modified to use reinforcement learning to create an intelligent control system. With no knowledge of the external system into which it is placed the audio coding system, under the control of the adaptive controller, is capable of adapting its coding configuration to achieve user set performance goals. | 11-22-2012 |
| 20120296657 | METHOD AND APPARATUS FOR MULTI-CHANNEL AUDIO PROCESSING USING SINGLE-CHANNEL COMPONENTS - Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels. | 11-22-2012 |
| 20120296658 | METHOD AND APPARATUS FOR REAL-TIME MULTIDIMENSIONAL ADAPTATION OF AN AUDIO CODING SYSTEM - An adaptive controller for a configurable audio coding system including a fuzzy logic controller modified to use reinforcement learning to create an intelligent control system. With no knowledge of the external system into which it is placed the audio coding system, under the control of the adaptive controller, is capable of adapting its coding configuration to achieve user set performance goals. | 11-22-2012 |
| 20120296659 | ENCODING DEVICE, DECODING DEVICE, SPECTRUM FLUCTUATION CALCULATION METHOD, AND SPECTRUM AMPLITUDE ADJUSTMENT METHOD - Disclosed is an encoding device whereby it is possible to improve the quality of an encoded signal, even when encoding music signals. In the encoding device, a Code-Excited Linear Prediction (CELP) encoder ( | 11-22-2012 |
| 20120303375 | AUDIO DECODING USING VARIABLE-LENGTH CODEBOOK APPLICATION RANGES - Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. At least one frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, and (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes. | 11-29-2012 |
| 20120310653 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, AND PROGRAM - A processing buffer unit stores an audio signal. A pitch calculation unit and a pitch cycle correction unit calculate a multiple of N as the number of samples in a pitch cycle of the audio signal, in which N is an integer equal to or more than 1. A processing control unit and a start-position movement amount correction unit sequentially determine, as a sample in a start position of a compression process in a time axis domain of the audio signal, a (multiple of N)-th sample from a start position immediately before the start position. An operation unit compresses samples in a predetermined number times the pitch cycle from the sample in the start position in a time axis domain, and sets the number of samples after the compression to be the multiple of N. The present technology, for example, may be applied to an audio signal processing apparatus. | 12-06-2012 |
| 20120310654 | System and Method for Non-destructively Normalizing Loudness of Audio Signals Within Portable Devices - Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed. | 12-06-2012 |
| 20120316885 | METHOD AND APPARATUS FOR ENCODING A SIGNAL - A method and apparatus for encoding a signal is provided herein. During operation a wideband signal that is to be encoded enters a filter bank. A highband signal and a lowband signal are output from the filter bank. Each signal is separately encoded. During the production of the highband signal, a downmixing operation is implemented after preprocessing, and prior to decimating. The downmixing operation greatly reduces system complexity. In fact, it will be observed that the highest sample rate in the prior-art implementation is 64 kHz whereas the sample rate in the system described above remains at 32 kHz or below. This represents a significant complexity saving, as do the reduced number of processing blocks. | 12-13-2012 |
| 20120316886 | SPARSE CODING USING OBJECT EXTTRACTION - The invention relates to a method and apparatus for efficient encoding of media signals including audio. A 2d sparse representation, or spikegram, of one frame of a digitized audio signal is generated using an overcomplete set of kernels. The spikegram is then mapped to a non-negative matrix, which is decomposed into a 3D component matrix containing hidden components and a 3D weight matrix using a two-dimensional non-negative matrix factorization. Elements of the 3D component and weight matrices are then adaptively quantized using integer programming to determine an optimal quantization scheme, and the quantized values are the optionally encoded using an arithmetic coder. | 12-13-2012 |
| 20120316887 | METHOD, APPARATUS, AND MEDIUM FOR BANDWIDTH EXTENSION ENCODING AND DECODING - Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality. | 12-13-2012 |
| 20120323582 | Hierarchical Audio Frequency Encoding and Decoding Method and System, Hierarchical Frequency Encoding and Decoding Method for Transient Signal - Hierarchical audio coding and decoding method and system and hierarchical audio coding and decoding method for transient signals are provided. In the present invention, by introducing a processing method for transient signal frames in the hierarchical audio coding and decoding methods, a segmented time-frequency transform is performed on the transient signal frames, and then the frequency-domain coefficients obtained by transformation are rearranged respectively within the core layer and within the extended layer, so as to perform the same subsequent coding processes, such as bit allocation, frequency-domain coefficient coding, etc., as those on the steady-state signal frames, thus enhancing the coding efficiency of the transient signal frames and improving the quality of the hierarchical audio coding and decoding. | 12-20-2012 |
| 20120323583 | COMMUNICATION TERMINAL AND COMMUNICATION METHOD - A communication terminal includes: a decoder which decodes an input bitstream received from another communication terminal, to generate an output audio signal and outputs the generated output audio signal to a speaker; an echo canceller which obtains an input audio signal representing sound captured by a microphone placed in a space to which the speaker outputs the sound, and removes, for respective subbands, an echo component included in the obtained input audio signal and corresponding to the output audio signal, to generate an audio signal for transmission; an encoder which codes the audio signal for transmission to generate an output bitstream and transmits the generated output bitstream to another communication terminal; and a control unit which controls, for the respective subbands, echo cancellation processing according to a reproduction band of at least one of the output audio signal and the audio signal for transmission. | 12-20-2012 |
| 20120323584 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique. | 12-20-2012 |
| 20120330670 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AN AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING AN ITERATIVE INTERVAL SIZE REDUCTION - An audio decoder has an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a numeric current context value describing a current context state. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates at least one table using an iterative interval size reduction to determine whether the numeric current context value is identical to a table context value described by an entry of the table or lies within an interval described by entries of the table, and derives a mapping rule index value describing a selected mapping table. | 12-27-2012 |
| 20130006644 | METHOD AND DEVICE FOR SPECTRAL BAND REPLICATION, AND METHOD AND SYSTEM FOR AUDIO DECODING - The present invention relates to a method and device for spectral band replication, and a method and system for audio decoding, and the method for spectral band replication comprises: A. searching for the position of a certain tone of an audio signal in MDCT frequency domain coefficients; B. according to the tone position, determining a spectral band replication period which is a bandwidth from a 0 frequency point to a frequency point of tone position, and a source frequency segment which is a frequency segment from a frequency point of the 0 frequency point shifting copyband_offset frequency points backwards to a frequency point of the frequency point of the tone position shifting the copyband_offset frequency points backwards, wherein said offset copyband_offset is greater than or equal to 0; and C. according to the spectral band replication period, carrying out spectral band replication on zero bit encoding subbands. | 01-03-2013 |
| 20130006645 | METHOD AND SYSTEM FOR AUDIO ENCODING AND DECODING AND METHOD FOR ESTIMATING NOISE LEVEL - The present invention relates to a method and system for audio encoding and decoding and a method for estimating a noise level, and the method for estimating a noise level in the present invention comprises: estimating a power spectrum of an audio signal to be encoded according to a frequency domain coefficient of the audio signal to be encoded; and estimating a noise level of a zero bit encoding subband audio signal according to the power spectrum obtained by calculating, and this noise level for controlling an energy proportion of noise filling to spectral band replication during decoding; wherein a zero bit encoding subband refers to an encoding subband of which allocated bit number is zero. The present invention can well reconstruct the uncoded frequency domain coefficients. | 01-03-2013 |
| 20130006646 | Transform Audio Codec and Methods for Encoding and Decoding a Time Segment of an Audio Signal - Methods and devices for efficient encoding/decoding of a time segment of an audio signal. Methods comprise deriving an indicator, z, of the position in a frequency scale of a residual vector associated with the time segment of the audio signal, and deriving a measure, Φ, related to the amount of structure of the residual vector. The methods further comprise determining whether a predefined criterion involving the measure Φ, the indicator z and a predefined threshold Θ, is fulfilled, which corresponds to estimating whether a change of sign of at least some of the non-zero coefficients of the residual vector would be audible after reconstruction of the audio signal time segment. The amplitude of the coefficients of the residual vector is encoded, and the signs of the coefficients of the residual vector are encoded only when it is determined that the criterion is fulfilled, and thus that a change of sign would be audible. | 01-03-2013 |
| 20130006647 | ENCODING DEVICE AND ENCODING METHOD, DECODING DEVICE AND DECODING METHOD, AND PROGRAM - The present invention relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program that reduce deterioration of sound quality due to encoding of audio signals. | 01-03-2013 |
| 20130013321 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - A method of processing an audio signal is disclosed. The present invention includes a method for processing an audio signal, comprising: receiving, by an audio processing apparatus, the spectral data including a current block, and substitution type information indicating whether to apply a shape prediction scheme to a current block; when the substitution type information indicates that the shape prediction scheme is applied to the current block, receiving lag information indicating an interval between spectral coefficients of the current block and the predictive shape vector of a current frame or a previous frame; obtaining spectral coefficients by substituting for spectral hole included in the current block using the predictive shape vector. | 01-10-2013 |
| 20130013322 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AND DECODING AN AUDIO INFORMATION, AND COMPUTER PROGRAM OBTAINING A CONTEXT SUB-REGION VALUE ON THE BASIS OF A NORM OF PREVIOUSLY DECODED SPECTRAL VALUES - An audio decoder has an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value which is determined in dependence on previously decoded spectral values. The arithmetic decoder obtains a plurality of context subregion values on the basis of previously decoded spectral values and derives a numeric current context value associated with one or more spectral values to be decoded in dependence on stored context subregion values. The arithmetic decoder computes the norm of a vector formed by a plurality of previously decoded spectral values in order to obtain a common context subregion value. An audio encoder uses a similar concept. | 01-10-2013 |
| 20130013323 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AND AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A MODIFICATION OF A NUMBER REPRESENTATION OF A NUMERIC PREVIOUS CONTEXT VALUE - An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value, and determines the numeric current context value in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder modifies a number representation of a numeric previous context value, describing a context state associated with one or more previously decoded spectral values, in dependence on a context subregion value, to acquire a number representation of a numeric current context value describing a context state associated with one or more spectral values to be decoded. An audio encoder uses a similar concept. | 01-10-2013 |
| 20130013324 | METHODS AND APPARATUS FOR CHARACTERIZING MEDIA - Methods and apparatus for characterizing media are described. A disclosed example apparatus includes a transformer, a decision metric processor, a signature determiner, and a processor to implement the transformer, the decision metric processor, and/or the signature determiner. The example transformer is to convert at least a portion of a block of audio into a frequency domain representation including a plurality of frequency components. The example decision metric processor is to: define a band of the frequency components; determine a difference in energy between a first convolution of a first complex vector with a first group of frequency bins in the band and a second convolution of a second complex vector with a second group of frequency bins in the band; and determine a decision metric for the band based on the difference. The example signature determiner is to determine a signature based on a value of the decision metric. | 01-10-2013 |
| 20130013325 | DECODING APPARATUS AND METHOD, ENCODING APPARATUS AND METHOD, AND PROGRAM - The present invention relates to a decoding apparatus, a decoding method, an encoding apparatus, an encoding method, and programs that can shorten the delay time caused by the band extension at the time of decoding, and restrain increases in resources on the decoding side. | 01-10-2013 |
| 20130018660 | AUDIO SIGNAL CODING AND DECODING METHOD AND DEVICE - Embodiments of the present invention provide an audio signal coding and decoding method and device. The coding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved. | 01-17-2013 |
| 20130024201 | ADAPTIVE TUNING OF THE PERCEPTUAL MODEL - Methods of encoding a signal using a perceptual model are described in which a signal to mask ratio parameter within the perceptual model is tuned. The signal to mask ratio parameter is tuned based on a function of the bitrate of the part of the signal which has already been encoded and the target bitrate for the encoding process. The tuned signal to 5 mask ratio parameter is used to compute a masking threshold for the signal which is then used to quantise the signal. | 01-24-2013 |
| 20130030817 | MDCT-Based Complex Prediction Stereo Coding - The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The method comprises applying independent bandwidth limits for the input channels. | 01-31-2013 |
| 20130030818 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD, ENCODER AND ENCODING METHOD, DECODER AND DECODING METHOD, AND PROGRAM - The present invention relates to a signal processing apparatus and a signal processing method, an encoder and an encoding method, a decoder and a decoding method, and a program capable of reproducing music signal having a better sound quality by expansion of frequency band. | 01-31-2013 |
| 20130030819 | AUDIO ENCODER, AUDIO DECODER AND RELATED METHODS FOR PROCESSING MULTI-CHANNEL AUDIO SIGNALS USING COMPLEX PREDICTION - An encoder, based on a combination of two audio channels, obtains a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded and written into a data stream together with the prediction information. A decoder generates decoded first and second channel signals using the prediction residual signal, the first combination signal and the prediction information. A real-to-imaginary transform may be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information. | 01-31-2013 |
| 20130030820 | METHOD, MEDIUM, AND SYSTEM SCALABLY ENCODING/DECODING AUDIO/SPEECH - A method, medium, and system scalably encoding/decoding audio/speech. The method includes splitting an input signal into a low frequency band signal that is lower than a predetermined frequency and a high frequency band signal that is higher than the predetermined frequency, scalably encoding the split low frequency band signal into a core layer and one or more extension layers and then decoding the encoded core layer and the encoded extension layers, generating an error signal by using the split low frequency band signal and a decoded signal of the encoded core layer and the encoded extension layers, and encoding the error signal and the high frequency band signal into a signal-to-noise ratio (SNR) enhancement layer and a bandwidth extension layer. | 01-31-2013 |
| 20130030821 | APPARATUS, SYSTEM AND METHOD FOR BUFFERING AUDIO DATA TO ALLOW LOW POWER STATES IN A PROCESSING SYSTEM DURING AUDIO PLAYBACK - An audio data stream from a processing system may be buffered to allow low power states in the processing system during audio playback. An audio buffer may be provided external to the processing system and between the processing system and an audio codec. The audio buffer may also shift to an alternate audio data interface mode when the processing system is in the low power state. Of course, many alternatives, variations, and modifications are possible without departing from this embodiment. | 01-31-2013 |
| 20130035943 | ENCODING DEVICE, DECODING DEVICE, ENCODING METHOD AND DECODING METHOD - Disclosed is an encoding device capable of improving decoded signal quality. A local search unit ( | 02-07-2013 |
| 20130041672 | METHOD AND ENCODER AND DECODER FOR SAMPLE-ACCURATE REPRESENTATION OF AN AUDIO SIGNAL - A method for providing information on the validity of encoded audio data is disclosed, the encoded audio data being a series of coded audio data units. Each coded audio data unit can include information on the valid audio data. The method includes: providing either information on a coded audio data level which describes the amount of data at the beginning of an audio data unit being invalid, or providing information on a coded audio data level which describes the amount of data at the end of an audio data unit being invalid, or providing information on a coded audio data level which describes both the amount of data at the beginning and the end of an audio data unit being invalid. A method for receiving encoded data including information on the validity of data and providing decoded output data is also disclosed. Furthermore, a corresponding encoder and a corresponding decoder are disclosed. | 02-14-2013 |
| 20130041673 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR GENERATING A WIDEBAND SIGNAL USING GUIDED BANDWIDTH EXTENSION AND BLIND BANDWIDTH EXTENSION - An apparatus, method and computer program for generating a wideband signal using a lowband input signal includes a processor for performing a guided bandwidth extension operation using transmitted parameters and a blind bandwidth extension operation only using derived parameters rather than transmitted parameters. To this end, the processor includes a parameter generator for generating the parameters for the blind bandwidth extension operation. | 02-14-2013 |
| 20130046545 | DIGITAL BROADCAST RECEPTION DEVICE - A digital broadcast reception device, provided with a first audio decoder for expanding and outputting a compressed digital audio signal included in the transport stream of a received digital broadcast; an encoder for encoding at least the digital audio signal outputted by the first audio decoder; and a second audio decoder for decoding the compressed digital audio signal outputted by the encoder. | 02-21-2013 |
| 20130046546 | APPARATUS AND METHOD FOR MODIFYING AN INPUT AUDIO SIGNAL - An apparatus for modifying an input audio signal has an excitation determiner, a storage device and a signal modifier. The excitation determiner determines a value of an excitation parameter of a subband of a plurality of subbands of the input audio signal based on an energy content of the subband. Further, the storage device stores a lookup table containing a plurality of spectral weighting factors. A spectral weighting factor of the plurality of spectral weighting factors is associated to a predefined value of the excitation parameter and a subband of the plurality of subbands. The storage device provides a spectral weighting factor corresponding to the determined value of the excitation parameter and corresponding to the subband, the value of the excitation parameter is determined for. Further, the signal modifier modifies a content of the subband of the audio signal, the value of the excitation parameter is determined for, based on the provided spectral weighting factor to provide a modified subband. | 02-21-2013 |
| 20130054251 | AUTOMATIC DETECTION OF AUDIO COMPRESSION PARAMETERS - For a media clip that includes audio content, a novel method for performing dynamic range compression of the audio content is presented. The method performs an analysis of the audio content. Based on the analysis of the audio content, the method generates a setting for an audio compressor that compresses the dynamic range of the audio content. The generated setting includes a set of audio compression parameters that include a noise gating threshold parameter (“noise gate”), a dynamic range compression threshold parameter (“threshold”), and a dynamic range compression ratio parameter (“ratio”). | 02-28-2013 |
| 20130054252 | Audio Processing Method and Apparatus - An audio processing method is disclosed. In the audio processing method, a modified discrete cosine transform (MDCT) algorithm is utilized to transform a present time domain audio signal into a spectrum audio signal. A spreading function (SF) coefficient of each partition domain of the spectrum audio signal is obtained by referencing an SF table. A masking partitioned energy threshold of each partition domain of the spectrum audio signal is calculated utilizing a logarithmic scale. An audio block type of each partition domain and an SMR of the spectrum audio signal are calculated. Subsequently, the spectrum audio signal is compressed into an audio bit stream according to the audio block type of each partition domain and the SMR. In addition, an audio signal processing apparatus is also disclosed in this invention. | 02-28-2013 |
| 20130054253 | AUDIO ENCODING DEVICE, AUDIO ENCODING METHOD, AND COMPUTER-READABLE RECORDING MEDIUM STORING AUDIO ENCODING COMPUTER PROGRAM - An audio encoding device includes a time-frequency converting unit that conducts time-frequency conversion of channel signals included in an audio signal having a plurality of channels in frame units having a certain length of time to convert the channel signals to respective frequency signals; a downmixing unit that generates a main signal representing a major component of a first channel and a second channel among the plurality of channels, and a residual signal that is a component orthogonal to the main signal; a weight determining unit that obtains a decoding value predicted and a decoding value predicted, obtains signal components affecting each other between the first channel and the second channel; a weighting unit that uses the weighting coefficient; a residual signal encoding unit that encodes the weighted residual signal the weighting coefficient; and a main signal encoding unit that encodes the main signal. | 02-28-2013 |
| 20130054254 | ENCODING METHOD, ENCODING APPARATUS, AND COMPUTER READABLE RECORDING MEDIUM - An encoding method executed by a computer, the method includes converting by the computer information about a transient included in a low-frequency component of an audio signal into information about a transient included in a high-frequency component of the audio signal, detecting, by the computer the transient of the high-frequency component of the audio signal based on the high-frequency component of the audio signal and on the information about the transient of the high-frequency component obtained by the converting; and encoding, by the computer the high-frequency component of the audio signal based on the transient detected by the detecting. | 02-28-2013 |
| 20130066638 | Echo Cancelling-Codec - Echo-cancellation is utilized in terminal devices such as speakerphones to compensate for acoustic echoes and interaction of the audio signal with the surrounding environment. An echo-cancelling codec incorporates encoding, decoding and acoustic echo-cancellation in a single device, enabling processing to be utilized that reduces processing and memory resources. The configuration enables processing information to also be shared between encoding, decoding and acoustic echo-cancellation functions to optimize operational characteristics. The acoustic echo cancelling codec interfaces between the amplitude signal domain, speaker and microphone, and an encoded data domain, a data interface, reducing component requirements required to provide echo-cancellation and coding functions. | 03-14-2013 |
| 20130066639 | SIGNAL PROCESSING METHOD, ENCODING APPARATUS THEREOF, AND DECODING APPARATUS THEREOF - A signal processing method performed by an encoding apparatus that down-mixes first through n channel signals to a mono-signal, an encoding apparatus, a decoding apparatus, and a decoding method are provided. The signal processing method includes: generating a spatial parameter between a reference channel signal that is from among the first through n channel signals, and residual channel signals from among the first through n channel signals except for the reference channel signal; and encoding and transmitting the spatial parameter to a decoding apparatus, whereby a down-mixed mono-signal may be exactly restored to original channel input signals. | 03-14-2013 |
| 20130066640 | AUDIO ENCODING/DECODING SCHEME HAVING A SWITCHABLE BYPASS - An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate. | 03-14-2013 |
| 20130066641 | Encoder Adaption in Teleconferencing System - The invention relates to a method and an arrangement for encoding of signals in teleconferencing. The method involves receiving ( | 03-14-2013 |
| 20130073295 | AUDIO CODEC WITH VIBRATOR SUPPORT - A dual channel audio coder decoder (codec) chip that has two output pins, which can be used to drive a pair of speakers in stereo mode, or a vibrator and a single speaker in mono mode. Each channel has its own DAC and audio power amplifier to receive an audio signal for driving a speaker. Each channel also has a variable signal generator to generate a vibrator signal for driving a vibrator. The DAC and variable signal generator outputs of each channel are input into a respective multiplexer. The multiplexer and the vibrator frequency are configured via an external digital communication interface. Other embodiments are also described. | 03-21-2013 |
| 20130073296 | AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHODS AND COMPUTER PROGRAM USING A SAMPLING RATE DEPENDENT TIME-WARP CONTOUR ENCODING - An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information. | 03-21-2013 |
| 20130073297 | METHODS AND DEVICES FOR PROVIDING AN ENCODED DIGITAL SIGNAL - In one embodiment, a method for providing an encoded digital signal is described comprising determining, for each data frame of a plurality of data frames of a digital signal, a plurality of pairs of an encoding data volume and an encoding quality, wherein each pair of an encoding data volume and an encoding quality specifies the encoding data volume required for achieving the encoding quality; determining for each data frame at least one or more interpolations between the plurality of determined pairs; determining a multi-frame relationship between encoding quality and encoding data volume required to encode the plurality of data frames at the encoding quality based on a combination of the at least one or more interpolations for the plurality of data frames; determining an encoding quality for the plurality of data frames based on the relationship; and providing at least one data frame of the plurality of data frames encoded at the determined encoding quality. | 03-21-2013 |
| 20130085762 | AUDIO ENCODING DEVICE - An audio encoding device capable of efficient encoding processing includes: a storage unit which stores audio data; a data acquisition controller which acquires the audio data from the storage unit; a transformation unit which processes an audio data signal outputted from the data acquisition unit for frequency transformation; a harmonic overtone generation/synthesizing unit which generates a harmonic based on a first output wave out of an output wave of the transformation unit and synthesizes the harmonic and a second output wave out of the output wave of the transformation unit, the second output wave being higher in frequency than the first output wave; and an encoder which subjects an output from the harmonic overtone generation/synthesizing unit to encoding processing. | 04-04-2013 |
| 20130085763 | CODEC DEVICES AND OPERATING AND DRIVING METHOD THEREOF - The present application discloses coding and decoding (CODEC) devices and operating and driving methods thereof. The CODEC device includes a first interface compatible with High Definition Audio (HDA) specification, a second interface compatible with Musical Instrument Digital Interface (MIDI) specification, and a converter. The converter is configured to convert a first MIDI command received from the first interface and output a corresponding first converted MIDI command via the second interface, and to convert a second MIDI command received from the second interface and output a corresponding second converted MIDI command via the first interface. | 04-04-2013 |
| 20130090933 | APPARATUS AND METHOD FOR PROCESSING AN INPUT AUDIO SIGNAL USING CASCADED FILTERBANKS - An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals. | 04-11-2013 |
| 20130090934 | APPARATUS AND METHOD FOR GENERATING A SYNTHESIS AUDIO SIGNAL AND FOR ENCODING AN AUDIO SIGNAL - An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal. The spectral domain patch generator is furthermore configured to select a first spectral domain patching algorithm from the plurality of patching algorithms for a first time portion and a second spectral domain patching algorithm from the plurality of patching algorithm for a second different time portion in accordance with the patching control signal to obtain the modified spectral representation. | 04-11-2013 |
| 20130096926 | AUDIO ADJUSTMENT SYSTEM - An audio adjustment system is provided that can output a user interface customized by the provider of the audio system instead of the electronic device manufacturer. Such an arrangement can save both field engineers and manufacturers a significant amount of time. Advantageously, in certain embodiments, such an audio adjustment system can be provided without knowledge of the electronic device's firmware. Instead, the audio adjustment system can communicate with the electronic device through an existing audio interface in the electronic device to enable a user to control audio enhancement parameters in the electronic device. For instance, the audio adjustment system can control the electronic device via an audio input jack on the electronic device. The electronic device can also include decoding features for decoding communications sent by the audio adjustment system. | 04-18-2013 |
| 20130096927 | AUDIO CODING DEVICE AND AUDIO CODING METHOD, AUDIO DECODING DEVICE AND AUDIO DECODING METHOD, AND PROGRAM - There is provided an audio coding device including a first windowing part that multiplies an audio signal by a first window function, a second windowing part that multiplies the audio signal by a second window function having a characteristic different from a characteristic of the first window function, a window selecting part that selects the first window function or the second window function as an optimum window function based on the audio signal multiplied by the first windowing part and the audio signal multiplied by the second windowing part, a coding part that codes a frequency spectrum of the audio signal multiplied by the optimum window function, and a transmitting part that transmits the frequency spectrum coded by the coding part and window function information representing the optimum window function. | 04-18-2013 |
| 20130096928 | METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL - The present invention relates to a method for processing an audio signal, comprising: determining bandwidth information indicating to which of a plurality of bands the current frame corresponds; determining information on the order corresponding to the present frame on the basis of the bandwidth information; performing a linear predictive analysis of the present frame to generate a first set linear predictive transform coefficient of a first order; performing a vector quantization on the first set linear predictive coefficient to generate a first index; performing a linear predictive analysis of the current frame to generate a second set linear predictive transform coefficient of a second order in accordance with the information on the order; and performing a vector quantization on a second set difference by using the first set index and the second set linear predictive transform coefficient, when the second set linear predictive coefficient is generated. | 04-18-2013 |
| 20130096929 | METHOD AND APPARATUS FOR SEARCHING IN A LAYERED HIERARCHICAL BIT STREAM FOLLOWED BY REPLAY, SAID BIT STREAM INCLUDING A BASE LAYER AND AT LEAST ONE ENHANCEMENT LAYER - A two-layer hierarchical audio bit stream can have a frame-based structure for the base layer bit stream and can be decoded independently from a higher layer and the decoding can start following every sync header. In the extension layer bit stream the frame structure may not be reflected on bit stream level. To facilitate seek operations with such highly compressed extension-layer data, the header of the extension layer bit stream comprises an FAT table with seek target positions. Because there are fewer entry points in the enhancement layer than sync headers in the base layer, a re-synchronisation and some base layer frames are required to start decoding of the enhancement layer and to generate the full audio quality. Three seeking ways of seeking are described, of which each one offers a different compromise between seeking accuracy, re-synchronisation latency and audio quality. | 04-18-2013 |
| 20130096930 | Multi-Resolution Switched Audio Encoding/Decoding Scheme - An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter. | 04-18-2013 |
| 20130103406 | CONTROL APPARATUS FOR AN ELECTRONIC DEVICE USING A BALANCED MICROPHONE CABLE - This invention enables a microphone to control the device having the microphone input. it includes both ideas related to the microphone as well as the device with the microphone input. It utilizes a standard two conductor shielded microphone cable and does not impact the audio from the microphone when the control operation is being performed. The microphone produces common-mode signals which are detected by the device with the microphone input. These signals provide control of the device with the microphone input. Likewise the device with the microphone input produces common-mode signals which are detected by the microphone. These signals provide control of the microphone visual display indicators. | 04-25-2013 |
| 20130103407 | METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL - The present invention relates to a method for processing an audio signal, comprising the following steps: performing a linear predictive analysis on the current frame of an audio signal so as to generate a first target vector, which is a target vector of a first stage, on the basis of a plurality of linear prediction transform coefficients; performing vector quantization on the first target vector so as to acquire a predetermined number of first temporary candidate code vectors of the first stage; calculating first temporary candidate errors, which are errors between the first temporary candidate code vectors and the first target vector; and determining a first number, which is the number of the first candidate code vectors, on the basis of the first temporary candidate errors, and acquiring first final candidate code vectors in the same amount as the first number. | 04-25-2013 |
| 20130103408 | Adaptive Linear Predictive Coding/Decoding - A method of coding/decoding of a digital audio signal comprising a succession of consecutive blocks of data, on the basis of a predictive filter. A modified predictive filter is used for the coding of at least one current block, the modified filter being constructed by the combination of: a rear filter calculated for a past block, preceding the current block, and enrichment parameters for the rear filter, which are determined as a function of the signal in the current block and comprising the coefficients of a modifying filter. | 04-25-2013 |
| 20130110521 | EXTRACTION AND ANALYSIS OF AUDIO FEATURE DATA | 05-02-2013 |
| 20130110522 | ENERGY LOSSLESS-ENCODING METHOD AND APPARATUS, AUDIO ENCODING METHOD AND APPARATUS, ENERGY LOSSLESS-DECODING METHOD AND APPARATUS, AND AUDIO DECODING METHOD AND APPARATUS | 05-02-2013 |
| 20130110523 | APPARTUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL | 05-02-2013 |
| 20130117028 | APPARATUS AND METHOD FOR CODING SIGNAL IN A COMMUNICATION SYSTEM - Disclosed are an apparatus for coding a signal in a communication system including: a coding unit configured to code voice and audio signals based on a code excited linear prediction (CELP) coding method; a residual signal calculation unit configured to calculate residual signals of the voice and audio signals; a frequency transform unit configured to transform the residual signal into a signal in a frequency domain; an energy calculation unit configured to use frequency coefficients of the residual signals to calculate frequency energy of the residual signals; an energy concentration calculation unit configured to calculate energy concentrations of each vector dimension of the residual signals from the frequency energy of the residual signals; and a vector dimension determination unit configured to compare the energy concentrations of each vector dimension to determine targeted vector dimensions of the residual signals. | 05-09-2013 |
| 20130117029 | SIGNAL CLASSIFICATION METHOD AND DEVICE, AND ENCODING AND DECODING METHODS AND DEVICES - Embodiments of the present invention provide a signal classification method and device, and encoding and decoding methods and devices. The encoding method includes: dividing a current frame into a low-frequency band signal and a high-frequency band signal; attenuating the high-frequency band signal or a to-be-encoded characteristic parameter of the high-frequency band signal according to an energy attenuation value of the low-frequency band signal, where the energy attenuation value indicates energy attenuation of the low-frequency band signal caused by encoding of the low-frequency band signal; and encoding the attenuated high-frequency band signal or the attenuated to-be-encoded characteristic parameter of the high-frequency band signal. The technical solutions according to the embodiments of the present invention can improve the effect of combining the low-frequency band signal and the high-frequency band signal at the decoder. | 05-09-2013 |
| 20130117030 | SIGNAL COMPRESSION METHOD AND APPARATUS - A signal compression method and apparatus are provided. The signal compression method includes: multiplying an input signal by a window function; calculating original autocorrelation coefficients of a windowed input signal; calculating a white-noise correction factor or a lag-window according to the original autocorrelation coefficients, and calculating modified autocorrelation coefficients according to the original autocorrelation coefficients, the white-noise correction factor and the lag-window; calculating linear prediction coefficients according to the modified autocorrelation coefficients; and outputting a coded bit stream according to the linear prediction coefficients. | 05-09-2013 |
| 20130117031 | AUDIO DATA ENCODING METHOD AND DEVICE - Provided is an audio data encoding method and device for use in Ogg/Vorbis encoding in portable multimedia players. The method comprises: receiving audio data requiring encoding ( | 05-09-2013 |
| 20130124214 | SIGNAL PROCESSING APPARATUS AND METHOD, AND PROGRAM - A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted. | 05-16-2013 |
| 20130124215 | CODER USING FORWARD ALIASING CANCELLATION - A codec supporting switching between time-domain aliasing cancellation transform coding mode and time-domain coding mode is made less liable to frame loss by adding a further syntax portion to the frames, depending on which the parser of the decoder may select between a first action of expecting the current frame to have, and thus reading forward aliasing cancellation data from the current frame and a second action of not-expecting the current frame to have, and thus not reading forward aliasing cancellation data from the current frame. In other words, while a bit of coding efficiency is lost due to the provision of the new syntax portion, it is merely the new syntax portion which provides for the ability to use the codec in case of a communication channel with frame loss. Without the new syntax portion, the decoder would not be capable of decoding any data stream portion after a loss and will crash in trying to resume parsing. Thus, in an error prone environment, the coding efficiency is prevented from vanishing by the introduction of the new syntax portion. | 05-16-2013 |
| 20130124216 | Method, Device and System for Signal Encoding and Decoding - A method, device, and system for signal encoding and decoding are disclosed. The method includes: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. According to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved. | 05-16-2013 |
| 20130132097 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting extension type identifier indicating whether extension area includes a residual signal from the side information; when the extension type identifier indicates that the extension area includes the residual signal, extracting control restriction information for residual using mode from the side information; receiving control information for controlling gain or panning of at least one object signal; estimating modified control information based on the control information and the control restriction information; obtaining at least one of enhanced object signal and one or more regular object signals from the downmix signal using the residual signal; and, generating an output signal using the modified control information and at least one of enhanced object signal and one or more regular object signal, wherein the control restriction information for residual using mode relates to a parameter indicating limiting degree of the control information in case of the residual using mode. | 05-23-2013 |
| 20130132098 | APPARATUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL INCLUDING INFORMATION BITSTREAM CONVERSION - Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals. | 05-23-2013 |
| 20130132099 | CODING DEVICE, DECODING DEVICE, AND METHODS THEREOF - Provided are a coding device, a decoding device, and methods thereof, with which it is possible to implement high sound quality coding and decoding in layered coding (scalable coding or embedded coding) wherein each layer comprises a plurality of bit rates (multi-rate). In the coding device ( | 05-23-2013 |
| 20130138445 | APPARATUS AND METHOD FOR DETERMINING BIT RATE FOR AUDIO CONTENT - An apparatus and a method for determining a bit rate of audio content, and more particularly, an audio content bit rate determining apparatus and a method capable of quickly and correctly identifying audio content compressed at a constant bit rate from among audio content compressed at a variable bit rate and a constant bit rate, are provided. The apparatus includes a first bit rate determiner for determining a bit rate type of audio content having frames with the same frame size by skipping a predetermined number of frames with respect to the audio content, and a second bit rate determiner for determining a bit rate type of audio content having frames with different frame sizes by skipping a predetermined number of frames with respect to the audio content. | 05-30-2013 |
| 20130138446 | AUDIO DECODER, AUDIO OBJECT ENCODER, METHOD FOR DECODING A MULTI-AUDIO-OBJECT SIGNAL, MULTI-AUDIO-OBJECT ENCODING METHOD, AND NON-TRANSITORY COMPUTER-READABLE MEDIUM THEREFOR - 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. | 05-30-2013 |
| 20130144630 | MULTI-CHANNEL AUDIO ENCODING AND DECODING - An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes. | 06-06-2013 |
| 20130144631 | AUDIO SIGNAL PROCESSING APPARATUS AND AUDIO SIGNAL PROCESSING METHOD - An audio signal processing apparatus that processes a bit stream generated by coding an audio signal on a frame-by-frame basis, the bit stream including, for each frame, coded data representing the audio signal, additional data and attribute information, the audio signal processing apparatus including a decoding unit configured to decode the coded data to generate a decoded signal, a processing unit configured to process the decoded signal, a detection unit configured to detect whether or not there has been a change in the attribute information, and a storage unit, wherein the processing unit is configured to, when the change is not detected, process the decoded signal by using at least two pieces of additional data stored, and when the change is detected, process the decoded signal by using only either additional data before detection of the change or additional data after detection of the change. | 06-06-2013 |
| 20130151260 | APPARATUS AND METHOD FOR AUDIO ENCODING - A method and apparatus provides for encoding an audio signal. A bit rate value is received. A set of energy thresholds based on the bit rate value is selected. The set of energy thresholds is one of a plurality of sets of energy thresholds. The energy thresholds of each set of energy thresholds correspond on a one-to-one basis with a set of sub-bands of the audio signal. The audio signal is received. The energy of each sub-band of the set of sub-bands is determined. A highest frequency sub-band that has an energy exceeding the corresponding threshold is determined. A selected bandwidth of the audio signal is encoded. The selected bandwidth includes only those frequencies of the audio signal that are in the highest frequency sub-band that has an energy exceeding the corresponding threshold, as well as the lower frequencies of the audio signal that are above a high-pass cut-off frequency. | 06-13-2013 |
| 20130151261 | Analog Signal Transfer System, Variable Compressor, and Variable Expander - An analog signal transfer system includes a transmission apparatus including a variable compressor that variably compresses input signals exponentially according to the amplitudes of the input signals; and a reception apparatus including a variable expander that variably expands the compressed signals exponentially according to the amplitudes of the compressed signals. | 06-13-2013 |
| 20130151262 | RESAMPLING OUTPUT SIGNALS OF QMF BASED AUDIO CODECS - An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes an analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting. | 06-13-2013 |
| 20130151263 | METHOD AND DEVICE FOR PROCESSING AUDIO SIGNALS - The present invention provides a method for processing audio signals, and the method comprises the steps of: receiving input audio signals corresponding to a plurality of spectral coefficients; obtaining location information that indicates a location of a particular spectral coefficient among said spectral coefficients, on the basis of energy of said input signals: generating a shape vector by using said location information and said spectral coefficients; determining a codebook index by searching for a codebook corresponding to said shape vector; and transmitting said codebook index and said location information, wherein said shape vector is generated by using a part which is selected from said spectral coefficients, and said selected part is selected on the basis of said location information. | 06-13-2013 |
| 20130159004 | Seamless Playback of Successive Multimedia Files - The present document relates to methods and systems for encoding and decoding multimedia files. In particular, the present document relates to methods and systems for encoding and decoding a plurality of audio tracks for seamless playback of the plurality of audio tracks. A method for encoding an audio signal comprising a first and a directly following second audio track for seamless and individual playback of the first and second audio tracks is described. The first and second audio tracks comprise a first and second plurality of audio frames, respectively. The method comprises jointly encoding the audio signal using a frame based audio encoder, thereby yielding a continuous sequence of encoded frames; extracting a first plurality of encoded frames from the continuous sequence of encoded frames; extracting a second plurality of encoded frames from the continuous sequence of encoded frames; appending one or more rear extension frames to an end of the first plurality of encoded frames; and appending one or more front extension frames to the beginning of the second plurality of encoded frames. | 06-20-2013 |
| 20130159005 | AUDIO DECODING DEVICE, AUDIO DECODING METHOD, AUDIO DECODING PROGRAM, AUDIO ENCODING DEVICE, AUDIO ENCODING METHOD, AND AUDIO ENCODING PROGRAM - In an audio decoding device of an embodiment, a plurality of decoding units execute different audio decoding schemes, respectively, to generate audio signals from coded sequences. An extraction unit extracts long-term encoding scheme information from a stream. The stream has a plurality of frames each including a coded sequence of an audio signal. The long-term encoding scheme information is a unit information for multiple frames and indicates that a common audio encoding scheme was used to generate coded sequences of the multiple frames. According to the extracted long-term encoding scheme information, a selection unit selects, from the plurality of decoding units, a decoding unit to be used commonly to decode the coded sequences of the multiple frames. | 06-20-2013 |
| 20130159006 | Object-based Audio-Visual Terminal and Bitstream Structure - As information to be processed at an object-based video or audio-visual (AV) terminal, an object-oriented bitstream includes objects, composition information, and scene demarcation information. Such bitstream structure allows on-line editing, e.g. cut and paste, insertion/deletion, grouping, and special effects. In the interest of ease of editing, AV objects and their composition information are transmitted or accessed on separate logical channels (LCs). Objects which have a lifetime in the decoder beyond their initial presentation time are cached for reuse until a selected expiration time. The system includes a de-multiplexer, a controller which controls the operation of the AV terminal, input buffers, AV objects decoders, buffers for decoded data, a composer, a display, and an object cache. | 06-20-2013 |
| 20130166306 | PULSE LOCATION SEARCH DEVICE, CODEBOOK SEARCH DEVICE, AND METHODS THEREFOR - A pulse location search device ( | 06-27-2013 |
| 20130166307 | Efficient Implementation of Phase Shift Filtering for Decorrelation and Other Applications in an Audio Coding System - An analysis/synthesis system uses existing analysis and synthesis filterbanks in an audio coding system to implement a phase shift filter that requires very little if any additional processing. One implementation using a single processing path can obtain a phase shift of either zero or ninety degrees. Another implementation that uses two processing paths can obtain a phase shift of essentially any desired angle. | 06-27-2013 |
| 20130166308 | ENCODER APPARATUS AND ENCODING METHOD - Provided is an encoder apparatus that can suppress the quality degradation of encoding processes. An ultimate selection candidate limiting unit ( | 06-27-2013 |
| 20130173271 | METHOD AND SYSTEM FOR REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES AND GENERAL PURPOSE AUDIO CODEC - A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms. | 07-04-2013 |
| 20130173272 | METHOD AND SYSTEM FOR REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES AND GENERAL PURPOSE AUDIO CODEC - A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms. | 07-04-2013 |
| 20130173273 | APPARATUS FOR DECODING A SIGNAL COMPRISING TRANSIENTS USING A COMBINING UNIT AND A MIXER - An apparatus for generating a decorrelated signal including a transient separator, a transient decorrelator, a second decorrelator, a combining unit and a mixer, wherein the transient separator is adapted to separate an input signal into a first signal component and into a second signal component such that the first signal component includes transient signal portions of the input signal and such that the second signal component includes non-transient signal portions of the input signal. The combining unit and the mixer are arranged so that a decorrelated signal from a combination unit is fed into the mixer as an input signal. | 07-04-2013 |
| 20130173274 | APPARATUS FOR GENERATING A DECORRELATED SIGNAL USING TRANSMITTED PHASE INFORMATION - An apparatus for generating a decorrelated signal having a receiving unit for receiving phase information, a transient separator, a transient decorrelator, a second decorrelator and a combining unit, wherein the transient separator is adapted to separate an input signal into a first signal component and into a second signal component such that the first signal component has transient signal portions of the input signal and such that the second signal component has non-transient signal portions of the input signal. The transient decorrelator is adapted to apply the phase information received by the receiving unit to a transient signal component. | 07-04-2013 |
| 20130173275 | AUDIO ENCODING DEVICE AND AUDIO DECODING DEVICE - Provided is an audio encoding device that can suppress degradation of audio quality. Spectral coefficients of synthesized signal from CELP core layer are utilized to fulfill spectral gaps in error signal spectrum coefficients from a transform coding layer. By both spectral coefficients, decoded signal spectral coefficients are generated. The decoded signal spectral coefficients and the input signal spectral coefficients are divided into a plurality of sub bands. In each sub band, the energy of the input signal spectral coefficient corresponding to a zero decoded error signal spectral coefficient is calculated, and the energy of the decoded signal spectral coefficient corresponding to the zero decoding error signal spectral coefficient is calculated, and their energy ratio is calculated and is quantized and transmitted. | 07-04-2013 |
| 20130179175 | Method and System for Encoding Audio Data with Adaptive Low Frequency Compensation - A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method. | 07-11-2013 |
| 20130185082 | AUDIO ENCODER, METHOD FOR PROVIDING OUTPUT SIGNAL, BANDWIDTH EXTENSION DECODER, AND METHOD FOR PROVIDING BANDWIDTH EXTENDED AUDIO SIGNAL - An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter fulfils a predefined criterion. | 07-18-2013 |
| 20130185083 | AUDIO ENCODING APPARATUS - There is provided an audio encoding apparatus that can avoid that audio data becomes irreproducible after fast-forward play. A quantization unit quantizes and buffers audio data into a buffer unit. A stream generating unit puts buffered audio data in a frame where there is a header related to the audio data in a stream and/or in one or plural frames preceding that frame. As for a predetermined frame, the stream generating unit puts in a data field of the frame the whole of an audio data piece related to a header included in that frame and puts audio sample data following that audio sample in a remaining part of the data field. As for a frame not a predetermined one, it puts in a data field of the frame an audio data piece related to a header included in that frame and/or audio data pieces following that audio data piece. | 07-18-2013 |
| 20130185084 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR BIT ALLOCATION FOR REDUNDANT TRANSMISSION - Compressibility-based reallocation of initial bit allocations for frames of an audio signal is described. Applications to redundancy-based retransmission of critical frames (e.g., for fixed-bit-rate modes of speech codec operation) are also described. | 07-18-2013 |
| 20130185085 | Audio Signal Encoding Method, Audio Signal Decoding Method, Encoding Device, Decoding Device, Audio Signal Processing System, Audio Signal Encoding Program, and Audio Signal Decoding Program - When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized. | 07-18-2013 |
| 20130191133 | APPARATUS FOR AUDIO DATA PROCESSING AND METHOD THEREFOR - An apparatus for audio data processing and a method therefor are provided. The apparatus includes a processing unit and an audio decoder. The processing unit receives an audio data stream, and the audio data stream includes a first frame header that complies with a communication protocol and an audio data encoded in an audio compression format. The processing unit parses the audio data stream to split the first frame header and the audio data, generates at least one frame information according to the first frame header, and acquires a second frame header according to the frame information. Here, the second frame header complies with an international audio and video coding standard. The audio decoder is coupled to the processing unit, receives the second frame header and the audio data from the processing unit, and decompresses the audio data according to the second frame header. | 07-25-2013 |
| 20130191134 | METHOD AND APPARATUS FOR DECODING AN AUDIO SIGNAL USING A SHAPING FUNCTION - The present invention relates to a method and apparatus for decoding an audio signal using a shaping function. According to one embodiment of the present invention, the method for decoding an audio signal comprises the following steps: taking frame data of the audio signal as an input; restoring a fixed codebook of the frame data using a random function; calculating a shaping function using an adaptive codebook of the frame data; shaping the restored fixed codebook using the shaping function; and synthesizing the audio signal from the frame data using the shaped fixed codebook and adaptive codebook. According to the present invention, the fixed codebook may be restored using the shaping function calculated on the basis of the adaptive codebook upon the occurrence of frame data loss, thus emphasizing a pitch period and reducing the influence of the fixed codebook between the pitch periods so as to reduce the degradation in the quality of the synthesized signal. | 07-25-2013 |
| 20130197918 | TRANSFERRING DATA VIA AUDIO LINK - Transferring data via audio link is described. In an example a short sequence of data can be transferred between two devices by encoding the sequence of data as an audio sequence. For example, the audio sequence may be a sequence of tones which vary in dependence on the encoded data. The sequence of data may be encoded by a first device and transmitted using a loudspeaker associated with the first device. At least one mobile communications device can be used to capture the audio sequence, for example using a microphone, and to decode the sequence, retrieving the data encoded therein. In some examples the encoded data may comprise a shortened URL or other information which can be used to control one or more aspects of the capture device. | 08-01-2013 |
| 20130197919 | "METHOD AND DEVICE FOR DETERMINING A NUMBER OF BITS FOR ENCODING AN AUDIO SIGNAL" - A method for determining a number of bits for encoding an audio signal comprising a core audio signal portion and a residual audio signal portion is described that comprises selecting, from the residual audio signal portion, a reference residual audio signal portion and at least one candidate residual audio signal portion; comparing the reference residual audio signal portion with the candidate residual audio signal portion; and determining the number of bits for encoding the audio signal depending on the result of the comparison. | 08-01-2013 |
| 20130197920 | DATA TRANSFER - Circuitry for transferring multiple digital data streams, e.g. digital audio data, over a single communications link such as a single wire. A pulse-length-modulator is responsive to a plurality of data streams to generate a series of data pulses with a single data pulse having a rising and falling edge in each of a plurality of transfer periods defined by a first clock signal. The timing of the rising and falling edge of each data pulse is dependent on a combination of the then current data samples from the plurality of data streams. The duration and position of the data pulse in the transfer window in effect defines a data symbol encoding the data. An interface receives the stream of data pulses, and data extraction circuitry samples the data pulse to determine which of the possible data symbols the pulse represents and determines a data value for at least one received data stream. | 08-01-2013 |
| 20130204630 | Controlling a Noise-Shaping Feedback Loop in a Digital Audio Signal Encoder - A method and apparatus are provided for controlling the shaping of encoding noise during the ADPCM encoding of a digital audio input signal. The noise-shaping is carried out through the use of feedback that comprises filtering noise. The method includes the following steps: obtaining a parameter for indicating a high spectral dynamic range of the signal, the parameter indicating a risk of instability of the feedback; detecting a risk of instability by comparing the indication parameter to at least one predetermined threshold; limiting the feedback in the event that a risk of instability is detected; and gradually reactivating the feedback over a predetermined number of frames subsequent to the current frame for which the feedback is limited. Also provided is an encoder with feedback, including a control module implementing the control method as described. | 08-08-2013 |
| 20130204631 | COMPRESSED SAMPLING AUDIO APPARATUS - Apparatus comprising at least one processor and at least one memory including computer code, the at least one memory and the computer code configured to with the at least one processor cause the apparatus to at least perform: transforming an audio signal into a sparse domain signal, the sparse domain signal representing the audio signal; transforming the sparse domain signal into a measurement domain signal; determining a sampling pattern dependent on the measurement domain signal; and measuring the measurement domain signal dependent on the sampling pattern. | 08-08-2013 |
| 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 |
| 20130218576 | AUDIO SIGNAL CODING DEVICE AND AUDIO SIGNAL CODING METHOD - An audio signal coding device divides a frequency spectrum obtained from an input digital signal to a plurality of bands, scales and quantizes divided frequency spectra based on a scalefactor of each of the bands and a common scale which is common to the plurality of bands, and codes quantized frequency spectra. The audio signal coding device includes a band number determination unit configured to calculate a number of coding bands for coding the quantized frequency spectra, and a common scale estimation unit configured to estimate the common scale in accordance with the number of coding bands. | 08-22-2013 |
| 20130218577 | Method and Device For Noise Filling - A method for perceptual spectral decoding comprises decoding of spectral coefficients recovered from a binary flux into decoded spectral coefficients of an initial set of spectral coefficients. The initial set of spectral coefficients are spectrum filled. The spectrum filling comprises noise filling of spectral holes by setting spectral coefficients in the initial set of spectral coefficients not being decoded from the binary flux equal to elements derived from the decoded spectral coefficients. The set of reconstructed spectral coefficients of a frequency domain formed by the spectrum filling is converted into an audio signal of a time domain. A perceptual spectral decoder comprises a noise filler, operating according to the method for perceptual spectral decoding. | 08-22-2013 |
| 20130218578 | System and Method for Mixed Codebook Excitation for Speech Coding - In accordance with an embodiment, a method of encoding an audio/speech signal includes determining a mixed codebook vector based on an incoming audio/speech signal, where the mixed codebook vector includes a sum of a first codebook entry from a first codebook and a second codebook entry from a second codebook. The method further includes generating an encoded audio signal based on the determined mixed codebook vector, and transmitting a coded excitation index of the determined mixed codebook vector. | 08-22-2013 |
| 20130226594 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AND AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING AN OPTIMIZED HASH TABLE - An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically encoded representation thereof, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code representing a spectral value, or a most significant bit-plane thereof, in a decoded form, in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. It evaluates a hash table, entries of which define both significant state values amongst the numeric context values and boundaries of intervals of numeric context values, in order to select the mapping rule, wherein the hash table ari_hash_m is defined as given in FIGS. | 08-29-2013 |
| 20130226595 | METHOD AND DEVICE FOR ENCODING A HIGH FREQUENCY SIGNAL, AND METHOD AND DEVICE FOR DECODING A HIGH FREQUENCY SIGNAL - A method and a device for encoding a high frequency signal, and a method and a device for decoding a high frequency signal are provided, which relate to encoding and decoding technology. The method for encoding a high frequency signal includes: determining a signal type of a high frequency signal of a current frame; smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded, if the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal; and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame. | 08-29-2013 |
| 20130226596 | APPARATUS AND METHOD FOR LEVEL ESTIMATION OF CODED AUDIO FRAMES IN A BIT STREAM DOMAIN - An apparatus for level estimation of an encoded audio signal is provided. The apparatus has a codebook determinator for determining a codebook from a plurality of codebooks as an identified codebook. The audio signal has been encoded by employing the identified codebook. Moreover, the apparatus has an estimation unit configured for deriving a level value associated with the identified codebook as a derived level value and for estimating a level estimate of the audio signal using the derived level value. | 08-29-2013 |
| 20130226597 | Methods for Improving High Frequency Reconstruction - The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit. | 08-29-2013 |
| 20130226598 | AUDIO ENCODER OR DECODER APPARATUS - An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform: determining from an audio signal at least a first part and a second part; encoding the first part of the audio signal with a first encoder for generating a first encoded audio signal; encoding the second part of the audio signal with a second encoder configured to generate a second encoded audio signal comprising for a first section of the second part an indicator to at least part of the first part of the audio signal; and determining the first section of the second part of the audio signal such that the first encoded audio signal and second encoded audio signal is within a defined encoding efficiency parameter. | 08-29-2013 |
| 20130231939 | COMPUTATIONALLY EFFICIENT AUDIO CODER - The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal. The audio coder further compares the computed short-block characteristics to a set of threshold values to detect presence of an attack in each of the short-blocks and changes the long-block frame length of one or more short-blocks upon detecting the attack in the respective one or more short-blocks. | 09-05-2013 |
| 20130231940 | PARAMETER DECODING APPARATUS AND PARAMETER DECODING METHOD - A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and an auto-regressive predictor produces a predicted parameter by multiplying a predictive coefficient with a past decoded parameter. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue. | 09-05-2013 |
| 20130238343 | 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. | 09-12-2013 |
| 20130238344 | COMPUTATIONALLY EFFICIENT AUDIO CODER - The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal. The audio coder further compares the computed short-block characteristics to a set of threshold values to detect presence of an attack in each of the short-blocks and changes the long-block frame length of one or more short-blocks upon detecting the attack in the respective one or more short-blocks. | 09-12-2013 |
| 20130238345 | PARTIALLY COMPLEX MODULATED FILTER BANK - An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal. | 09-12-2013 |
| 20130238346 | LOW COMPLEXITY TARGET VECTOR IDENTIFICATION - It is inter alia disclosed to identify one or more target vectors from a plurality of candidate vectors, each candidate vector having sorted elements and being associated with a respective class of one or more code vectors of a codebook and at least one of the candidate vectors being associated with a respective class of two or more code vectors that comprise the respective candidate vector and at least one code vector obtainable from the respective candidate vector by one of permutation and signed permutation, the target vectors having, among all candidate vectors of the plurality of candidate vectors, smallest distances towards a at least sorted representation of an input vector. The identifying comprises checking, for a candidate vector of the plurality of candidate vectors, at least based on a distance between the candidate vector and a reference vector and on a distance between the reference vector and the at least sorted representation of the input vector, if a distance between the at least sorted representation of the input vector and the candidate vector is larger than a distance between the at least sorted representation of the input vector and the reference vector. The identifying further comprises computing, for the candidate vector, the distance between the at least sorted representation of the input vector and the candidate vector only if the checking yields a negative result. | 09-12-2013 |
| 20130246073 | CODING DEVICE, CODING METHOD, DECODING DEVICE, DECODING METHOD, AND STORAGE MEDIUM - For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length. | 09-19-2013 |
| 20130246074 | Low-Complexity Spectral Analysis/Synthesis Using Selectable Time Resolution - The signal processing is based on the concept of using a time-domain aliased frame as a basis for time segmentation and spectral analysis, performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall “segmented” time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame. | 09-19-2013 |
| 20130246075 | CODING APPARATUS, DECODING APPARATUS, CODING METHOD AND DECODING METHOD - There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit and a gain codebook select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit expresses the gain of the sub-band not selected, by mutually interpolating the selected gains. | 09-19-2013 |
| 20130246076 | CODING OF STRINGS - It is inter alia disclosed to apply a function on a string in accordance with a rule of a set of rules, the string comprising first and second representatives, wherein the function ensures that the string comprises at least one of a predetermined representative at an end of the string after the function has been applied; and to determine a representation of the string, wherein said representation comprises at least one encoded representative, each of said at least one encoded representative being associated with at least one representative of the string, wherein said representation does not comprise an encoded representative being associated with at least one of the at least one predetermined representative at the end of the string. | 09-19-2013 |
| 20130246077 | ADAPTIVE PROCESSING WITH MULTIPLE MEDIA PROCESSING NODES - Techniques for adaptive processing of media data based on separate data specifying a state of the media data are provided. A device in a media processing chain may determine whether a type of media processing has already been performed on an input version of media data. If so, the device may adapt its processing of the media data to disable performing the type of media processing. If not, the device performs the type of media processing. The device may create a state of the media data specifying the type of media processing. The device may communicate the state of the media data and an output version of the media data to a recipient device in the media processing chain, for the purpose of supporting the recipient device's adaptive processing of the media data. | 09-19-2013 |
| 20130253938 | Audio Encoding Using Adaptive Codebook Application Ranges - A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal. | 09-26-2013 |
| 20130253939 | AUDIO ENCODING DEVICE, METHOD AND PROGRAM, AND AUDIO DECODING DEVICE, METHOD AND PROGRAM - An audio packet error concealment system includes an encoding unit for encoding an audio signal consisting of a plurality of frames, and an auxiliary information encoding unit for estimating and encoding auxiliary information about a temporal change of power of the audio signal. The auxiliary information is used in packet loss concealment in decoding of the audio signal. The auxiliary information about the temporal change of power may contain a parameter that functionally approximates a plurality of powers of subframes shorter than one frame, or may contain information about a vector obtained by vector quantization of a plurality of powers of subframes shorter than one frame. | 09-26-2013 |
| 20130262128 | SYSTEM AND METHOD FOR METHOD FOR IMPROVING SPEECH INTELLIGIBILITY OF VOICE CALLS USING COMMON SPEECH CODECS - System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal. | 10-03-2013 |
| 20130262129 | METHOD AND APPARATUS FOR AUDIO ENCODING FOR NOISE REDUCTION - A method and apparatus for audio signal encoding for noise reduction are provided. The method includes: receiving an audio signal and performing modified discrete cosine transformation (MDCT) on the audio signal to convert the audio signal into a long block or a short block; reducing noise included in the audio signal in accordance with the long block or the short block; and performing advanced audio coding (AAC) on the long block or the short block in which noise is reduced. | 10-03-2013 |
| 20130262130 | STEREO PARAMETRIC CODING/DECODING FOR CHANNELS IN PHASE OPPOSITION - A method and apparatus for the parametric encoding of a stereo digital-audio signal. The method includes encoding a mono signal produced by downmixing applied to the stereo signal and encoding spatialisation information of the stereo signal. Downmixing includes determining, for a predetermined set of frequency sub-bands, a phase difference between two stereo channels; obtaining an intermediate channel by rotating a first predetermined channel of the stereo signal through an angle obtained by reducing the phase difference; determining the phase of the mono signal from the phase of the signal that is the sum of the intermediate channel and the second stereo signal, and from a phase difference between, on the one hand, the signal that is the sum of the intermediate channel and the second channel and, on the other hand, the second channel of the stereo signal. Also provided are a decoding method, an encoder and a decoder. | 10-03-2013 |
| 20130268277 | WIRELESS TRANSACTION COMMUNICATION APPARATUS AND METHOD - The invention can be a simple method for data transfer from one electronic device to another. In this embodiment, a sender can upload data to a server using an out-of-band connection while broadcasting an identification signal over one or several mediums, such as acoustic and/or radio (Ultrasound, Bluetooth, infrared, etc. . . . ). In the case that a connection to the server can be established, the receiver will detect the identification signal, decode it, and request the information from the server. The receiver can then send an authorization for a transaction through the server via an out-of-band connection or directly to the sender via one of the primary communication mediums, at which point the transaction is complete. | 10-10-2013 |
| 20130268278 | Audio Communication System, An Audio Transmitter and An Audio Receiver - An audio communication system, an audio transmitter and an audio receiver are described herein. In one aspect, the audio transmitter includes: An analog to digital converter configured to convert an analog audio signal into a digital audio signal; a digital compressor configured to compress the dynamic range of said digital audio signal; a digital modulator configured to modulate the compressed digital audio signal into a wireless communication signal; and the analog to digital converter, the digital compressor and the digital modulator are integrated in one integrated circuit. The described devices improve the compression and expansion device linearity, resulting in better match for the compressor at the transmitter and the expandor at the receiver. | 10-10-2013 |
| 20130268279 | METHODS AND APPARATUS FOR PERFORMING VARIABLE BLOCK LENGTH WATERMARKING OF MEDIA - Methods and apparatus for performing variable block length watermarking of media are disclosed. An example method to encode auxiliary data in audio data comprises selecting a frequency based on a code, selecting a block size based on the code, a combination of the block size and the frequency to represent of the code, encoding the code in an audio stream according to the block size and the frequency, and transmitting the audio stream including the encoded code. | 10-10-2013 |
| 20130268280 | APPARATUS AND METHOD FOR GEOMETRY-BASED SPATIAL AUDIO CODING - An apparatus for generating at least one audio output signal based on an audio data stream having audio data relating to one or more sound sources is provided. The apparatus has a receiver for receiving the audio data stream having the audio data. The audio data has one or more pressure values for each one of the sound sources. Furthermore, the audio data has one or more position values indicating a position of one of the sound sources for each one of the sound sources. Moreover, the apparatus has a synthesis module for generating the at least one audio output signal based on at least one of the one or more pressure values of the audio data of the audio data stream and based on at least one of the one or more position values of the audio data of the audio data stream. | 10-10-2013 |
| 20130268281 | Apparatus and Method for Decomposing an Input Signal Using a Pre-Calculated Reference Curve - An apparatus for decomposing a signal having an number of at least three channels includes an analyzer for analyzing a similarity between two channels of an analysis signal related to the signal having at least two analysis channels, wherein the analyzer is configured for using a pre-calculated frequency dependent similarity curve as a reference curve to determine the analysis result. The signal processor processes the analysis signal or a signal derived from the analysis signal or a signal, from which the analysis signal is derived using the analysis result to obtain a decomposed signal. | 10-10-2013 |
| 20130275140 | METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNALS AT LOW COMPLEXITY - An audio encoding apparatus is provided that includes a transform unit to transform an audio signal in the time domain into an audio spectrum in the frequency domain; a bit allocation unit to determine the number of allocated bits by using spectral energy in predetermined frequency band units for the audio spectrum; and an encoding unit to determine the number of unit magnitude pulses for factorial pulse coding based on the number of allocated bits for the audio spectrum and to perform factorial pulse coding in the frequency band units for the audio spectrum by using the determined number of unit magnitude pulses. | 10-17-2013 |
| 20130275141 | PREPROCESSING METHOD, PREPROCESSING APPARATUS AND CODING DEVICE - The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals. | 10-17-2013 |
| 20130275142 | SIGNAL PROCESSING DEVICE, METHOD, AND PROGRAM - The present technology relates to a signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal. | 10-17-2013 |
| 20130282382 | Audio Encoder and Decoder - The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit. | 10-24-2013 |
| 20130282383 | Audio Encoder and Decoder - The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit. | 10-24-2013 |
| 20130282384 | Apparatus and Method for Encoding a Multi-Channel Audio Signal - An encoding apparatus comprises a frame processor ( | 10-24-2013 |
| 20130282385 | NICAM Decoder with Output Resampler - A NICAM audio signal re-sampler may include a non-linear interpolator configured to interpolate in a non-linear manner between sequential digital samples that are based on a stream of demodulated NICAM audio samples. A phase differential calculator may be included that compares phase information at different resolutions. | 10-24-2013 |
| 20130282386 | MULTI-CHANNEL ENCODING AND/OR DECODING - A method comprising: receiving input signals for multiple channels; and parameterizing the received input signals into parameters defining multiple different object spectra and defining a distribution of the multiple different object spectra in the multiple channels. | 10-24-2013 |
| 20130282387 | FILTERING IN THE TRANSFORMED DOMAIN - A method for processing a signal in the form of consecutive sample blocks, the method comprising filtering in a transformed domain of sub-bands, and particularly equalization processing, applied to a current block in the transformed domain, and filtering-adjustment processing that is applied in the transformed domain to at least one block adjacent to the current block. | 10-24-2013 |
| 20130282388 | SONG TRANSITION EFFECTS FOR BROWSING - In one aspect, a method of providing directive transitions between audio signals comprises associating a first/second browsing direction (A | 10-24-2013 |
| 20130290003 | METHOD AND APPARATUS FOR ENCODING AND DECODING HIGH FREQUENCY FOR BANDWIDTH EXTENSION - Disclosed are a method and apparatus for encoding and decoding a high frequency for bandwidth extension. The method includes: estimating a weight; and generating a high frequency excitation signal by applying the weight between random noise and a decoded low frequency spectrum. | 10-31-2013 |
| 20130297322 | ERROR CONCEALMENT METHOD AND APPARATUS FOR AUDIO SIGNAL AND DECODING METHOD AND APPARATUS FOR AUDIO SIGNAL USING THE SAME - An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme. | 11-07-2013 |
| 20130304480 | ENCODING AND DECODING OF SLOT POSITIONS OF EVENTS IN AN AUDIO SIGNAL FRAME - An apparatus for decoding, an apparatus for encoding, a method for decoding and a method for encoding positions of slots having events in an audio signal frame and respective computer programs and encoded signals, wherein the apparatus for decoding has: an analysing unit for analysing a frame slots number indicating the total of slots of the audio signal frame, an event slots number indicating the number of slots having the events of the audio signal frame, and an event state number, and a generating unit for generating an indication of a plurality of positions of slots having the events in the audio signal frame using the frame slots number, the event slots number and the event state number. | 11-14-2013 |
| 20130304481 | Determining the Inter-Channel Time Difference of a Multi-Channel Audio Signal - There is provided a method and device for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A set of local maxima of a cross-correlation function involving at least two different channels of the multi-channel audio signal is determined (S | 11-14-2013 |
| 20130311192 | ENCODING METHOD, ENCODER, PERIODIC FEATURE AMOUNT DETERMINATION METHOD, PERIODIC FEATURE AMOUNT DETERMINATION APPARATUS, PROGRAM AND RECORDING MEDIUM - An encoding technique encoding a sound signal at a low bit rate with reduced processing. The technique includes: an interval determination determining an interval T between samples corresponding to periodicity of an audio signal or an integer multiple of a fundamental frequency of the audio signal from a set S of candidates for the interval T; and a side information generating encoding the determined interval T to obtain side information. The interval determining determines the interval T from a set S of Y candidates (Y| 11-21-2013 | |
| 20130317829 | Audio Decoding Method and Associated Apparatus - An audio decoding method is provided. In the audio decoding method, a synchronization word and a corresponding packet header are inserted at the beginning of each packet data. A position of the packet data is confirmed according to the synchronization word, and the packet data is then decoded according to information in the packet header. Accordingly, when an error occurs during the decoding process, the decoding process skips to a next packet data for decoding to avoid noise. In addition, a packet header can be directly accessed in the situation of a fast-forward operation to obtain decoding information of the packet data to perform audio decoding. | 11-28-2013 |
| 20130317830 | THREE-DIMENSIONAL SOUND COMPRESSION AND OVER-THE-AIR TRANSMISSION DURING A CALL - A method for encoding three dimensional audio by a wireless communication device is disclosed. The wireless communication device detects an indication of a plurality of localizable audio sources. The wireless communication device also records a plurality of audio signals associated with the plurality of localizable audio sources. The wireless communication device also encodes the plurality of audio signals. | 11-28-2013 |
| 20130317831 | BANDWIDTH EXPANSION METHOD AND APPARATUS - A bandwidth expansion method and apparatus are disclosed, where the method includes: estimating a bandwidth of at least one decoded frame of a whole-band signal, so as to obtain an estimated bandwidth, where the estimated bandwidth corresponds to a whole-band signal that a decoded lower-band signal needs to be extended into; performing first predictive decoding on a part of the lower-band signal in a band above an effective bandwidth of the lower-band signal and below the estimated bandwidth, so as to obtain the part of the lower-band signal above the effective bandwidth of the lower-band signal and below the estimated bandwidth; and performing second predictive decoding on a part of the lower-band signal in a band above the estimated bandwidth, so as to obtain the part of the lower-band signal above the estimated bandwidth. | 11-28-2013 |
| 20130317832 | AUDIO FRAME TIMING CORRECTION METHOD AND WIRELESS DEVICE - An audio frame timing correction method and a wireless device are provided. A controller generates a reference clock for audio coding/decoding such that the reference clock runs fast and moved forward within an audio data sampling interval with the remaining time becoming a margin of the interval. An audio codec decodes demodulated data based on the reference clock, and codes an audio signal based on the reference clock. A demodulator detects wireless frame deviation and determines an adjustment timing whereat the wireless frame symbol timing and the audio frame timing are corrected based on the deviation and the margin. Upon the adjustment timing, the controller synchronizes audio sampling timing with the wireless frame symbol timing. | 11-28-2013 |
| 20130317833 | Methods and Systems for Generating Filter Coefficients and Configuring Filters - Methods for generating a palette of feedback (IIR) filter coefficient sets and using the palette to configure (e.g., adaptively update) a prediction filter which includes a feedback filter, and a system for performing any of the methods. Examples of the system include an encoder, including a prediction filter and configured to encode data indicative of a waveform signal (e.g., samples of an audio signal), and a decoder. In some embodiments, the prediction filter is included in an encoder operable to generate (and assert to a decoder) encoded data including filter coefficient data indicative of the selected IIR coefficient set with which the prediction filter was configured during generation of the encoded data. In some embodiments, the timing with which adaptive updating of prediction filter configuration occurs or is allowed to occur is constrained (e.g., to optimize efficiency of prediction encoding). | 11-28-2013 |
| 20130325486 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 12-05-2013 |
| 20130325487 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 12-05-2013 |
| 20130332174 | AUDIO CODEC SUPPORTING TIME-DOMAIN AND FREQUENCY-DOMAIN CODING MODES - An audio codec supporting both, time-domain and frequency-domain coding modes, having low-delay and an increased coding efficiency in terms of iterate/distortion ratio, is obtained by configuring the audio encoder such that same operates in different operating modes such that if the active operative mode is a first operating mode, a mode dependent set of available frame coding modes is disjoined to a first subset of time-domain coding modes, and overlaps with a second subset of frequency-domain coding modes, whereas if the active operating mode is a second operating mode, the mode dependent set of available frame coding modes overlaps with both subsets, i.e. the subset of time-domain coding modes as well as the subset of frequency-domain coding modes. | 12-12-2013 |
| 20130332175 | AUDIO CODEC USING NOISE SYNTHESIS DURING INACTIVE PHASES - A parametric background noise estimate is continuously updated during an active or non-silence phase so that the noise generation may immediately be started with upon the entrance of an inactive phase following the active phase. In accordance with another aspect, a spectral domain is very efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching. | 12-12-2013 |
| 20130332176 | NOISE GENERATION IN AUDIO CODECS - The spectral domain is efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching. | 12-12-2013 |
| 20130332177 | APPARATUS AND METHOD FOR CODING A PORTION OF AN AUDIO SIGNAL USING A TRANSIENT DETECTION AND A QUALITY RESULT - An apparatus for coding a portion of an audio signal to obtain an encoded audio signal for the portion of the audio signal includes a transient detector for detecting whether a transient signal is located in the portion of the audio signal to obtain a transient detection result, an encoder stage for performing first and second encoding algorithms on the audio signal, the first and second encoding algorithms having differing first and second characteristics, respectively, a processor for determining which encoding algorithm results in an encoded audio signal being a better approximation to the portion of the audio signal with respect to the other encoding algorithm to obtain a quality result, and a controller for determining whether the encoded audio signal for the portion of the audio signal is to be generated by either the first or the second encoding algorithm based on the transient-detection and quality results. | 12-12-2013 |
| 20130339034 | SYSTEM AND METHOD FOR EFFICIENTLY TRANSLATING MEDIA FILES BETWEEN FORMATS USING A UNIVERSAL REPRESENTATION - An apparatus and method are described for reading a file into a universal representation and translating from that universal representation into various file formats. For example, a method according to one embodiment comprises: reading compressed audio data from a first audio file, the first audio file comprising audio data compressed using a first compression algorithm and bookkeeping data having a first format, the bookkeeping data specifying a location of the compressed audio data within the first audio file; and generating a universal representation of the first audio file without decompressing and recompressing the audio data, the universal representation having bookkeeping data of a second format specifying the location of compressed audio data within the universal representation. | 12-19-2013 |
| 20130339035 | AUTOMATIC CONVERSION OF SPEECH INTO SONG, RAP, OR OTHER AUDIBLE EXPRESSION HAVING TARGET METER OR RHYTHM - Captured vocals may be automatically transformed using advanced digital signal processing techniques that provide captivating applications, and even purpose-built devices, in which mere novice user-musicians may generate, audibly render and share musical performances. In some cases, the automated transformations allow spoken vocals to be segmented, arranged, temporally aligned with a target rhythm, meter or accompanying backing tracks and pitch corrected in accord with a score or note sequence. Speech-to-song music applications are one such example. In some cases, spoken vocals may be transformed in accord with musical genres such as rap using automated segmentation and temporal alignment techniques, often without pitch correction. Such applications, which may employ different signal processing and different automated transformations, may nonetheless be understood as speech-to-rap variations on the theme. | 12-19-2013 |
| 20130339036 | ENCODING AND DECODING OF PULSE POSITIONS OF TRACKS OF AN AUDIO SIGNAL - An apparatus for decoding an encoded audio signal is provided. The apparatus includes a pulse information decoder and a signal decoder. The pulse information decoder is adapted to decode a plurality of pulse positions, wherein each one of the pulse positions indicates a position of one of the pulses of the track, wherein the pulse information decoder is configured to decode the plurality of pulse positions by using a track positions number, a total pulses number, and one state number. The signal decoder is adapted to decode the encoded audio signal by generating a synthesized audio signal using the plurality of pulse positions and a plurality of predictive filter coefficients. | 12-19-2013 |
| 20130339037 | Spectral Translation/Folding in the Subband Domain - The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications. | 12-19-2013 |
| 20130339038 | Post-Quantization Gain Correction in Audio Coding - A gain adjustment apparatus ( | 12-19-2013 |
| 20130346087 | Filling of Non-Coded Sub-Vectors in Transform Coded Audio Signals - A spectrum filler for filling non-coded residual sub-vectors of a transform coded audio signal includes a sub-vector compressor ( | 12-26-2013 |
| 20130346088 | AUDIO CODING METHOD AND APPARATUS - An audio coding method and apparatus are disclosed, where the method includes: dividing an input audio signal into a low-band signal and a high-band signal; identifying types of the low-band signal and the high-band signal; adaptively allocating a total input rate of the audio signal to the low-band signal and the high-band signal according to different coding modes corresponding to the low-band signal and the high-band signal; and coding the low-band signal through a coding mode corresponding to the low-band signal according to the low-band rate, and coding the high-band signal through a coding mode corresponding to the high-band signal according to the high-band rate. In embodiments of the present application, when the low-band signal and the high-band signal are coded, coding rates are adaptively adjusted according to different types of the signals, thereby improving overall audio coding performance. | 12-26-2013 |
| 20140006035 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD | 01-02-2014 |
| 20140006036 | METHOD AND APPARATUS FOR CODING AND DECODING | 01-02-2014 |
| 20140006037 | ENCODING DEVICE, ENCODING METHOD, AND PROGRAM | 01-02-2014 |
| 20140012588 | REDUCED COMPLEXITY TRANSFORM FOR A LOW-FREQUENCY-EFFECTS CHANNEL - The computational resources that are needed to apply a transform-based filterbank to a limited-bandwidth audio signals are reduced by performing an integrated process of combining real-valued input data into complex-valued data and applying a short transform to the complex-valued data, applying a bank of very short transforms to the output of the integrated process, and deriving a sequence of real-valued output data from the outputs of the bank of very short transforms. | 01-09-2014 |
| 20140012589 | METHOD AND APPARATUS TO ENCODE AND DECODE AN AUDIO/SPEECH SIGNAL - A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed. | 01-09-2014 |
| 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 |
| 20140019143 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 01-16-2014 |
| 20140019144 | ENCODING DEVICE, DECODING DEVICE, AND METHOD THEREOF FOR SECIFYING A BAND OF A GREAT ERROR - Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band. | 01-16-2014 |
| 20140019145 | ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM, AND RECORDING MEDIUM - In encoding, a frequency-domain sample sequence derived from an acoustic signal is divided by a weighted envelope and is then divided by a gain, the result obtained is quantized, and each sample is variable-length encoded. The error between the sample before quantization and the sample after quantization is quantized with information saved in this variable-length encoding. This quantization is performed under a rule that specifies, according to the number of saved bits, samples whose errors are to be quantized. In decoding, variable-length codes in an input sequence of codes are decoded to obtain a frequency-domain sample sequence; an error signal is further decoded under a rule that depends on the number of bits of the variable-length codes; and from the obtained sample sequence, the original sample sequence is obtained according to supplementary information. | 01-16-2014 |
| 20140019146 | FRAME ELEMENT POSITIONING IN FRAMES OF A BITSTREAM REPRESENTING AUDIO CONTENT - A better compromise between a too high bitstream and decoding overhead on the one hand and flexibility of frame element positioning on the other hand is achieved by arranging that each of the sequence of frames of the bitstream has a sequence of N frame elements and, on the other hand, the bitstream has a configuration block having a field indicating the number of elements N and a type indication syntax portion indicating, for each element position of the sequence of N element positions, an element type out of a plurality of element types with, in the sequences of N frame elements of the frames, each frame element being of the element type indicated, by the type indication portion, for the respective element position at which the respective frame element is positioned within the sequence of N frame elements of the respective frame in the bitstream. | 01-16-2014 |
| 20140019147 | DISTRIBUTED CALL SERVER SUPPORTING COMMUNICATION SESSIONS IN A COMMUNICATION SYSTEM AND METHOD - An apparatus, method, and computer program manage communication sessions that include a plurality of portions. Different processors handle each portion of a communication session. The apparatus, method, and computer program transfer the communication session from one of the processors to another of the processors during the different portions of the communication session. | 01-16-2014 |
| 20140025386 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR AUDIO OBJECT CLUSTERING - Systems, methods, and apparatus for grouping audio objects into clusters are described. | 01-23-2014 |
| 20140025387 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 01-23-2014 |
| 20140025388 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 01-23-2014 |
| 20140025389 | AUTOMATIC CONFIGURATION OF METADATA FOR USE IN MIXING AUDIO PROGRAMS FROM TWO ENCODED BITSTREAMS - An audio coding system uses mixing metadata to control the attenuation of a main audio program that is subsequently mixed with an associated audio program. The value of the attenuation is calculated by analyzing the estimated loudness of the main and associated audio programs. | 01-23-2014 |
| 20140032225 | COMPUTING DEVICE AND SIGNAL ENHANCEMENT METHOD - A computing device provides a resonance algorithm to process digital signal data according to a principle of physical resonance. The resonance algorithm determines a division length n of digital signal data according to a frequency f1 of an audio signal to be detected and a sampling frequency f2, which is used for sampling the digital signal data by a coder. | 01-30-2014 |
| 20140032226 | METHOD AND APPARATUS FOR PROCESSING AUDIO DATA - A method and apparatus for processing audio data are provided. When an encoded audio bitstream sampled at a sampling frequency is received, a resampling ratio for processing the encoded audio bitstream is computed. If the the resampling ratio is within the resampling threshold range, then the encoded audio bitstream is processed in frequency domain and a desired number of audio samples per frame are outputted according to the resampling ratio. The encoded audio bitstream is processed in frequency domain using sample rate converter integrated into a filter bank of an audio decoder. If the resampling ratio is outside the resampling threshold range, then the encoded audio bitstream is processed in time domain and a desired number of audio samples per frame are outputted according to the resampling ratio. | 01-30-2014 |
| 20140032227 | BIT ERROR MANAGEMENT METHODS FOR WIRELESS AUDIO COMMUNICATION CHANNELS - Systems and methods are described for managing bit errors present in an encoded bit stream representative of a portion of an audio signal, wherein the encoded bit stream is received via a channel in a wireless communications system. The channel may comprise, for example, a Synchronous Connection-Oriented (SCO) channel or an Extended SCO (eSCO) channel in a Bluetooth® wireless communications system. | 01-30-2014 |
| 20140039901 | VOICE-CODED IN-BAND DATA FOR INTERACTIVE CALLS - A voice-coded in-band communication device monitors a voice-coded channel to detect data to present to a user. During operation, the communication device can detect a data-encoding signal from the voice-coded channel, such that the voice-coded channel can carry an audio signal that includes a voice signal and the data-encoding signal. The device decodes the data-encoding signal to detect a data element. The data element can include information that is to be presented to a local user, a request from a remote device for information about the local user, or information that the system can use to establish a peer-to-peer connection with the remote device over a separate data channel. The device can also generate a filtered audio signal to present to the user by removing the detected data-encoding signal from the voice-coded channel, and then reproduces the filtered audio signal for the user. | 02-06-2014 |
| 20140039902 | DATA COMPRESSION APPARATUS, COMPUTER-READABLE STORAGE MEDIUM HAVING STORED THEREIN DATA COMPRESSION PROGRAM, DATA COMPRESSION SYSTEM, DATA COMPRESSION METHOD, DATA DECOMPRESSION APPARATUS, DATA COMPRESSION/DECOMPRESSION APPARATUS, AND DATA STRUCTURE OF COMPRESSED DATA - A data compression/decompression apparatus, for example, acquires sampling data obtained by sampling an audio signal with a predetermined period, and converts the sampling data into frequency domain data. The data compression/decompression apparatus divides a data sequence of the converted frequency domain data into a plurality of blocks such that the number of pieces of data included in each block is variable, and compresses each block. | 02-06-2014 |
| 20140039903 | Audio Watermarking - A system, including a processor to define opportunities for encoding a watermark into an audio stream having sections, each section, when represented in the frequency domain, including a signal of amplitude against frequency, the processor being operative to, for each one of the sections, identify a fundamental frequency, f being the frequency with the largest amplitude of the signal in the one section, the fundamental frequency f defining harmonic frequencies, each harmonic frequency being at a frequency f/2n or 2fn, n being a positive integer, and define the one section as an opportunity for encoding at least part of the watermark if the amplitude of the signal of the one section is less than a value v for all frequencies in one or more different frequency ranges, each of the different frequency ranges being centered around different ones of the harmonic frequencies. Related apparatus and methods are also described. | 02-06-2014 |
| 20140046670 | AUDIO ENCODING METHOD AND APPARATUS, AUDIO DECODING METHOD AND APPARATUS, AND MULTIMEDIA DEVICE EMPLOYING THE SAME - Provided is a method of encoding an audio signal. A method of encoding an audio signal includes generating a modified signal of a time domain to compensate a frequency resolution in frame units, analysis-windowing the modified signal of the time domain by using a window type which is designed to have an overlapping section less than 50%, and generating transform coefficients of a frequency domain by transforming the analysis-windowed signal of the time domain. | 02-13-2014 |
| 20140046671 | AUDIO SIGNAL PROCESSING CIRCUIT FOR REDUCING ZERO CROSSING DISTORTION AND METHOD THEREOF - An audio signal processing circuit includes an encoding circuit, a first audio conversion circuit, and a second audio conversion circuit. The encoding circuit receives pulse coded modulation signals and generates a first audio signal and a second audio signal accordingly. The first audio conversion circuit generates a first pulse width modulation (PWM) signal according to consecutive values of the first audio signal for configuring a first power stage circuit. The second audio conversion circuit generates a second PWM signal according to consecutive values of the second audio signal for configuring a second power stage circuit. The pulse width of the first PWM signal is configured to be substantially equal to the pulse width of the second PWM signal, and the pulse edges of the first PWM signal and the second PWM signal are configured to be separated by a predetermined time interval to mute the audio signal processing circuit. | 02-13-2014 |
| 20140046672 | Signal Classification Method and Device, and Encoding and Decoding Methods and Devices - Embodiments of the present invention provide a signal classification method and device, and encoding and decoding methods and devices. The encoding method includes dividing a current frame into a low-frequency band signal and a high-frequency band signal, attenuating the high-frequency band signal or a to-be-encoded characteristic parameter of the high-frequency band signal according to an energy attenuation value of the low-frequency band signal, where the energy attenuation value indicates energy attenuation of the low-frequency band signal caused by encoding of the low-frequency band signal, and encoding the attenuated high-frequency band signal or the attenuated to-be-encoded characteristic parameter of the high-frequency band signal. The technical solutions according to the embodiments of the present invention can improve the effect of combining the low-frequency band signal and the high-frequency band signal at the decoder. | 02-13-2014 |
| 20140046673 | Multistage IIR Filter and Parallelized Filtering of Data with Same - In some embodiments, a multistage filter whose biquad filter stages are combined with latency between the stages, a system (e.g., an audio encoder or decoder) including such a filter, and methods for multistage biquad filtering. In typical embodiments, all biquad filter stages of the filter are operable independently to perform fully parallelized processing of data. In some embodiments, the inventive multistage filter includes a buffer memory, at least two biquad filter stages, and a controller coupled and configured to assert a single stream of instructions to the filter stages. Typically, the multistage filter is configured to perform multistage filtering of a block of input samples in a single processing loop with iteration over a sample index but without iteration over a biquadratic filter stage index. | 02-13-2014 |
| 20140052454 | METHOD FOR DETERMINING FORMAT OF LINEAR PULSE-CODE MODULATION DATA - A method for determining a data format of linear pulse-code modulation is provided. The method includes steps of reading a plurality of bytes of the linear pulse-code modulation data; obtaining a data property by performing a predetermined calculation on the plurality of bytes; and determining the data format of the linear pulse-code modulation data according to the data property. | 02-20-2014 |
| 20140052455 | METHOD, MEDIUM, AND APPARATUS ENCODING AND/OR DECODING MULTICHANNEL AUDIO SIGNALS - A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end. | 02-20-2014 |
| 20140058735 | Artificial Neural Network Based System for Classification of the Emotional Content of Digital Music - A system for classification of the emotional content of music is provided. An encoder receives a digital audio recording of a piece of music, and encodes it using musical notes and associated amplitudes. The artificial neural network is configured to take a plurality of encoded time slices and provide output indicative of the emotional content of the music. | 02-27-2014 |
| 20140058736 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD AND COMPUTER PROGRAM PRODUCT - According to an embodiment, a signal processing apparatus includes an estimation unit and an updating unit. The estimation unit is configured to estimate an auxiliary variable of a target section including first and second sections of input signals by using an approximating auxiliary function for approximating an auxiliary function having an auxiliary variable as an argument. The auxiliary function is determined according to an objective function that outputs a function value that is smaller as a statistical independence of separated signals into which input signals in time-series are separated by a demixing matrix is higher. The estimation unit is configured to estimate a value of the auxiliary variable of the target section based on the estimated auxiliary variable. The updating unit is configured to update the demixing matrix such that a function value of the approximating auxiliary function is minimized. | 02-27-2014 |
| 20140058737 | HYBRID SOUND SIGNAL DECODER, HYBRID SOUND SIGNAL ENCODER, SOUND SIGNAL DECODING METHOD, AND SOUND SIGNAL ENCODING METHOD - A hybrid sound signal decoder decodes a bitstream including audio frames encoded by an audio encoding process using a low delay filter bank and speech frames encoded by a speech encoding process using linear prediction coefficients. When a current frame to be decoded is an ith frame which is an initial speech frame after switching from an audio frame to a speech frame, the hybrid sound signal decoder generates sub-frames which are a signal corresponding to an i−1th frame before being encoded, using a sub-frame which is a signal generated using a signal of the i−1th frame before being encoded, the signal of the i−1th frame being obtained by decoding the ith frame. | 02-27-2014 |
| 20140067404 | INTENSITY STEREO CODING IN ADVANCED AUDIO CODING - A system and method for selectively applying Intensity Stereo coding to an audio signal is described. The system and method make decisions on whether to apply Intensity Stereo coding to each scale factor band of the audio signal based on (1) the number of bits necessary to encode each scale factor band using Intensity Stereo coding, (2) spatial distortions generated by using Intensity Stereo coding with each scale factor band, and (3) switching distortions for each scale factor band resulting from switching Intensity Stereo coding on or off in relation to a previous scale factor band. | 03-06-2014 |
| 20140067405 | ADAPTIVE AUDIO CODEC SELECTION DURING A COMMUNICATION SESSION - A method for adaptive audio codec selection during a communication session is disclosed. The method can include negotiating a set of audio codecs for use during the communication session. The method can further include defining multiple audio tiers. Each audio tier can be associated with a network condition and can define an audio codec from the set of audio codecs for use in the associated network condition. The method can also include using a first audio codec during the wireless communication session. The method can additionally include determining a changed network condition selecting a second audio codec by determining the audio tier corresponding to the changed network condition. The method can further include, in response to the changed network condition, switching from the first audio codec to a second audio codec that is defined by an audio tier having an associated network condition corresponding to the changed network condition. | 03-06-2014 |
| 20140074484 | ENCODER AND DECODER DRIVER DEVELOPMENT TECHNIQUES - A codec architecture including an audio wave driver and a coded topology driver. The audio wave driver is communicatively coupled to an audio engine and an analog audio codec. The coded topology driver is communicatively coupled to the audio wave driver by a set of interfaces that enables streamlined code implementation, improved operation efficiency and power savings, while allowing vendors to supply differentiating functionality outside of the basic requirements of the operating system. | 03-13-2014 |
| 20140074485 | EFFICIENT AND SCALABLE PARAMETRIC STEREO CODING FOR LOW BITRATE AUDIO CODING APPLICATIONS - The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described. | 03-13-2014 |
| 20140074486 | APPARATUS AND METHOD FOR AUDIO ENCODING AND DECODING EMPLOYING SINUSOIDAL SUBSTITUTION - An apparatus for generating an audio output signal based on an encoded audio signal spectrum has a processing unit, a pseudo coefficients determiner, a spectrum modification unit, a spectrum-time conversion unit, a controllable oscillator and a mixer. The pseudo coefficients determiner is configured to determine pseudo coefficients of the decoded audio signal spectrum. The spectrum modification unit is configured to set the pseudo coefficients to a predefined value to acquire a modified audio signal spectrum. The spectrum-time conversion unit is configured to convert the modified audio signal spectrum to a time-domain. The controllable oscillator is configured to generate a time-domain oscillator signal and is controlled by the spectral location and the spectral value of at least one of the pseudo coefficients. The mixer is configured to mix the time-domain conversion signal and the time-domain oscillator signal. | 03-13-2014 |
| 20140074487 | APPARATUS AND METHOD FOR RESTORING MULTI-CHANNEL AUDIO SIGNAL USING HE-AAC DECODER AND MPEG SURROUND DECODER - Provided is a method for controlling synchronizing downmix signals and MPEG surround side information signals by controlling a delay according to the kind of downmix audio signals in an MPEG surround decoder. When multi-channel audio signals are restored using an HE-AAC decoder and a low-power MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. Anther delay unit delays spatial parameters to compensate for a delay caused in QMF and Nyquist banks when time-domain downmix signals are used. Also, when multi-channel audio signals are restored using an HE-AAC decoder and a high-quality MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. | 03-13-2014 |
| 20140074488 | ENCODING OF STEREOPHONIC SIGNALS - A stereophonic signal is converted into a mid channel signal and a side channel signal. Noise is added to the side channel signal. The amount of noise is selected depending on masking thresholds for at least two channels of the stereophonic signal. The mid channel signal and the modified side channel signal are quantized for transmission. Alternatively or in addition, a set of quantization parameter for the quantization of the side channel signal is selected depending on the masking thresholds. | 03-13-2014 |
| 20140074489 | SOUND SIGNAL HYBRID ENCODER, SOUND SIGNAL HYBRID DECODER, SOUND SIGNAL ENCODING METHOD, AND SOUND SIGNAL DECODING METHOD - A sound signal hybrid encoder includes: a signal analysis unit which determines a scheme for encoding a frame included in a sound signal; an LFD encoder which encodes a frame to generate an LFD frame; an LP encoder which encodes a frame to generate an LP frame; a switching unit which switches between the encoders according to a result of the determination by the signal analysis unit; and an AC signal generation unit which generates an AC signal according to a scheme selected from among schemes, outputs the generated AC signal, and also outputs an AC flag indicating the selected scheme. | 03-13-2014 |
| 20140081645 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AN AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A DETECTION OF A GROUP OF PREVIOUSLY-DECODED SPECTRAL VALUES - An audio decoder for providing a decoded audio information includes a arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state. The arithmetic decoder is configured to determine or modify the current context state in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder is configured to detect a group of a plurality of previously-decoded spectral values, which fulfill, individually or taken together, a predetermined condition regarding their magnitudes, and to determine the current context state in dependence on a result of the detection. | 03-20-2014 |
| 20140081646 | Method and a Decoder for Attenuation of Signal Regions Reconstructed with Low Accuracy - The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics. | 03-20-2014 |
| 20140088973 | METHOD AND APPARATUS FOR ENCODING AN AUDIO SIGNAL - A hybrid speech encoder detects changes from music-like sounds to speech-like sounds. When the encoder detects music-like sounds (e.g., music), it operates in a first mode, in which it employs a frequency domain coder. When the encoder detects speech-like sounds (e.g., human speech), it operates in a second mode, and employs a time domain or waveform coder. When a switch occurs, the encoder backfills a gap in the signal with a portion of the signal occurring after the gap. | 03-27-2014 |
| 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 |
| 20140088975 | Method for Controlling a Computing Device over Existing Broadcast Media Acoustic Channels - The present invention provides a method for broadcasting a control message across an existing broadcast transport medium such as TV, Radio, Webcasts, Smart Phones/Tablets, Audio Buttons/Toys and PA/audio systems via a common acoustic channel with normal traditional broadcasts over those mediums to control computing devices, especially smart mobile devices such as smart phones or tablets. The present invention also involves a method for encoding the control message onto an acoustic wave composed of active sinusoidal frequency components selected out of monitored possible frequencies according to the concept of mathematical combinations. This encoding method is aimed at using a minimum amount of transmitted energy to get a maximum amount of data with as little reflective noise as possible. | 03-27-2014 |
| 20140088976 | AUDIO DECODING METHOD AND APPARATUS - An audio decoding method and apparatus are disclosed. The audio decoding method includes: receiving data packets; when data packet loss is detected and audio data of an audio frame corresponding to M channels in N channels is lost, if audio data of other channels than the M channels in the N channels, which belongs to the same audio frame as the lost audio data in the audio frame, is not lost, decoding the un-lost audio data; extracting a signal characteristic parameter of the data obtained after decoding; determining whether a correlation exists between a first channel and a second channel; and if the correlation exists, performing packet loss concealment processing on the lost audio data of the audio frame corresponding to the first channel according to the second channel. The audio decoding method and apparatus can effectively improve the effect of packet loss concealment processing in audio decoding. | 03-27-2014 |
| 20140088977 | METHOD AND APPARATUS FOR ENCODING AND DECODING STEREO SIGNAL AND MULTI-CHANNEL SIGNAL - Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal. | 03-27-2014 |
| 20140088978 | FORENSIC DETECTION OF PARAMETRIC AUDIO CODING SCHEMES - The present document relates to audio forensics, notably the blind detection of traces of parametric audio encoding/decoding. In particular, the present document relates to the detection of parametric frequency extension audio coding, such as spectral band replication (SBR) or spectral extension (SPX), from uncompressed waveforms such as PCM (pulse code modulation) encoded waveforms. A method for detecting frequency extension coding history in a time domain audio signal is described. The method may comprise transforming the time domain audio signal into a frequency domain, thereby generating a plurality of subband signals in a corresponding plurality of subbands comprising low and high frequency subbands; determining a degree of relationship between subband signals in the low frequency subbands and subband signals in the high frequency subbands; wherein the degree of relationship is determined based on the plurality of subband signals; and determining frequency extension coding history if the degree of relationship is greater than a relationship threshold. | 03-27-2014 |
| 20140095178 | 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. | 04-03-2014 |
| 20140095179 | 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. | 04-03-2014 |
| 20140100855 | Audio Signal Transient Detection - Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment. | 04-10-2014 |
| 20140100856 | APPARATUS AND METHOD FOR CODING AND DECODING MULTI OBJECT AUDIO SIGNAL WITH MULTI CHANNEL - Provided are an apparatus and method for coding and decoding a multi object audio signal with multi channel. The apparatus includes: a multi channel encoding means for down-mixing an audio signal including a plurality of channels, generating a spatial cue for the audio signal including the plurality of channels, and generating first rendering information including the generated spatial cue; and a multi object encoding unit for down-mixing an audio signal including a plurality of objects, which includes the down-mixed signal from the multi channel encoding unit, generating a spatial cue for the audio signal including the plurality of objects, and generating second rendering information including the generated spatial cue, wherein the multichannel encoding unit generates a spatial cue for the audio signal including the plurality of objects regardless of a Coder-DECoder (CODEC) scheme the limits the multi channel encoding unit. | 04-10-2014 |
| 20140108020 | MULTI-MODE AUDIO RECOGNITION AND AUXILIARY DATA ENCODING AND DECODING - Audio signal processing enhances audio watermark embedding and detecting processes. Audio signal processes include audio classification and adapting watermark embedding and detecting based on classification. Advances in audio watermark design include adaptive watermark signal structure data protocols, perceptual models, and insertion methods. Perceptual and robustness evaluation is integrated into audio watermark embedding to optimize audio quality relative the original signal, and to optimize robustness or data capacity. These methods are applied to audio segments in audio embedder and detector configurations to support real time operation. Feature extraction and matching are also used to adapt audio watermark embedding and detecting. | 04-17-2014 |
| 20140108021 | 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. | 04-17-2014 |
| 20140114666 | ORCHESTRATED ENCODING AND DECODING - Orchestrated encoding schemes facilitate encoding and decoding of data in content signals at several points in the distribution path of content items. Orchestrated encoding adheres to a set of encoding rules that enables multiple watermarks and corresponding applications to co-exist, avoids collisions among watermarks, and simplifies metadata and routing database infrastructure. | 04-24-2014 |
| 20140114667 | Transform Audio Codec and Methods for Encoding and Decoding a Time Segment of an Audio Signal - Methods and devices for efficient encoding/decoding of a time segment of an audio signal. The methods comprise deriving an indicator, z, of the position in a frequency scale of a residual vector associated with the time segment of the audio signal, and deriving a measure, Φ, related to the amount of structure of the residual vector. The methods further comprise determining whether a predefined criterion involving the measure Φ, the indicator z and a predefined threshold Θ, is fulfilled, which corresponds to estimating whether a change of sign of at least some of the non-zero coefficients of the residual vector would be audible after reconstruction of the audio signal time segment. The respective amplitude of the coefficients of the residual vector is encoded, and the signs of the coefficients of the residual vector are encoded only when it is determined that the criterion is fulfilled, and thus that a change of sign would be audible. | 04-24-2014 |
| 20140114668 | APPARATUS, SYSTEM AND METHOD FOR BUFFERING AUDIO DATA TO ALLOW LOW POWER STATES IN A PROCESSING SYSTEM DURING AUDIO PLAYBACK - An audio data stream from a processing system may be buffered to allow low power states in the processing system during audio playback. An audio buffer may be provided external to the processing system and between the processing system and an audio codec. The audio buffer may also shift to an alternate audio data interface mode when the processing system is in the low power state. Of course, many alternatives, variations, and modifications are possible without departing from this embodiment. | 04-24-2014 |
| 20140114669 | METHODS, APPARATUS AND ARTICLES OF MANUFACTURE TO PERFORM AUDIO WATERMARK DECODING - Example methods, apparatus and articles of manufacture to perform audio watermark decoding are disclosed. A disclosed example method includes receiving an audio signal including an audience measurement code embedded therein using a first plurality of frequency components, sampling the audio signal, transforming the sampled audio signal into a first frequency domain representation, determining whether the code is detectable in the first plurality of frequency components of the first frequency domain representation, and when the code is not detected in the first plurality of frequency components, examining a second plurality of frequency components of a second frequency domain representation to determine whether the code is detected, the second plurality of frequency components being offset from the first plurality of frequency components by a first offset, the first offset corresponding to a sampling frequency mismatch. | 04-24-2014 |
| 20140114670 | Adaptive Audio Signal Coding - Example embodiments described herein generally provide for adaptive audio signal coding of low-frequency and high-frequency audio signals. More specifically, audio signals are categorized into high-frequency audio signals and low-frequency audio signals. Then, based on a set coding and/or characteristics of the low-frequency audio signals, the low-frequency coding manner is selected. Similarly, but in addition to, a bandwidth extension mode to code the high-frequency audio signals is selected according to the low-frequency coding manner and/or characteristics of the audio signals. | 04-24-2014 |
| 20140122093 | Method for Carrying Out an Audio Conference, Audio Conference Device, and Method for Switching Between Encoders - A method and an audio conference device for carrying out an audio conference are disclosed, whereby classification information associated with a respective audio date flow is recorded for supplied audio data flows. According to a result of an evaluation of the classification information, the audio data flows are associated with at least three groups which are homogeneous with regard to the results. The individual audio data flows are processed uniformly in each group in terms of the signals thereof, and said audio data flows processed in this way are superimposed in order to form audio conference data flows to be transmitted to the communication terminals. | 05-01-2014 |
| 20140129236 | SYSTEM AND METHOD FOR LINEAR FREQUENCY TRANSLATION, FREQUENCY COMPRESSION AND USER SELECTABLE RESPONSE TIME - A method and system has been developed and demonstrated which provides real-time frequency translation, frequency compression, and user selectable response time for non-deterministic signals. This method and system provides for the real-time separation and isolation of theoretically an infinite amount of frequencies present in an incoming non-deterministic signal. The bandwidth of the filter for the separated frequencies is user selectable and provides varying rise times for the individual frequencies. The linear frequency shifting property of the algorithm creates bandwidth compression opportunities while signals are present in a channel for transmission. | 05-08-2014 |
| 20140142955 | Encoding Digital Media for Fast Start on Digital Media Players - Systems, methods and computer program products are disclosed for encoding digital media for fast start on digital media players. In some implementations, a set of frames at the beginning of a digital media file or stream are encoded at a first bitrate (e.g., a constant low bitrate), and subsequent frames of the digital file or stream are encoded at a second, bitrate that may be higher than the first bitrate. | 05-22-2014 |
| 20140142956 | Transform Coding of Speech and Audio Signals - In a method of perceptual transform coding of audio signals in a telecommunication system, performing the steps of determining transform coefficients representative of a time to frequency transformation of a time segmented input audio signal; determining a spectrum of perceptual sub-bands for said input audio signal based on said determined transform coefficients; determining masking thresholds for each said sub-band based on said determined spectrum; computing scale factors for each said sub-band based on said determined masking thresholds, and finally adapting said computed scale factors for each said sub-band to prevent energy loss for perceptually relevant sub-bands. | 05-22-2014 |
| 20140142957 | FRAME ERROR CONCEALMENT METHOD AND APPARATUS, AND AUDIO DECODING METHOD AND APPARATUS - Disclosed are a frame error concealment method and apparatus and an audio decoding method and apparatus. The frame error concealment (FEC) method includes: selecting an FEC mode based on at least one of a state of at least one frame and a phase matching flag, with regard to a time domain signal generated after time-frequency inverse transform processing; and performing corresponding time domain error concealment processing on the current frame based on the selected FEC mode, wherein the current frame is an error frame or the current frame is a normal frame when the previous frame is an error frame. | 05-22-2014 |
| 20140142958 | MULTI-MODE AUDIO RECOGNITION AND AUXILIARY DATA ENCODING AND DECODING - Audio signal processing enhances audio watermark embedding and detecting processes. Audio signal processes include audio classification and adapting watermark embedding and detecting based on classification. Advances in audio watermark design include adaptive watermark signal structure data protocols, perceptual models, and insertion methods. Perceptual and robustness evaluation is integrated into audio watermark embedding to optimize audio quality relative the original signal, and to optimize robustness or data capacity. These methods are applied to audio segments in audio embedder and detector configurations to support real time operation. Feature extraction and matching are also used to adapt audio watermark embedding and detecting. | 05-22-2014 |
| 20140142959 | RECONSTRUCTION OF A HIGH-FREQUENCY RANGE IN LOW-BITRATE AUDIO CODING USING PREDICTIVE PATTERN ANALYSIS - A predictive pattern high-frequency reconstruction system and method that finds patterns in high-frequency components of an audio signal, encodes the audio signal into an encoded bitstream along with pattern information, and then uses the patterns to reconstruct the high-frequency components during decoding. The high-frequency components can be reconstructed using the pattern information alone. Embodiments of the system and method map normalized subband signals of the audio signal to a scaled representation of a time-frequency grid containing multiple tiles and perform statistical analysis on each tile to estimate subband parameters and determine whether a pattern exists. If a pattern does exist, it can be encoded in the encoded bitstream, transmitted, and used to reconstruct the high-frequency components at the decoder. A direct search technique and a fast Fourier transform (FFT) technique may be used to perform the statistical analysis. | 05-22-2014 |
| 20140149123 | Method and Device for Storing Audio Data - A method for storing audio data is disclosed, including: recording basic information of a versatile audio data storage file into the versatile audio data storage file; storing Versatile Audio Codec (VAC) frame data into the versatile audio data storage file sequentially; recording payload information of the versatile audio data storage file into the versatile audio data storage file; and recording index information of VAC frames stored in the versatile audio data storage file into the versatile audio data storage file. A device for storing the audio data is also disclosed, including: a basic information record module, a VAC frame data storage module, a payload information record module and an index information record module. The file generated with this method is simple and is easy to read and access, which can be applied to various applications of the versatile audio frequently. | 05-29-2014 |
| 20140149124 | 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. | 05-29-2014 |
| 20140149125 | 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. | 05-29-2014 |
| 20140149126 | SYSTEM FOR PERCEIVED ENHANCEMENT AND RESTORATION OF COMPRESSED AUDIO SIGNALS - A system for processing compressed audio includes a signal enhancer module configured to generate one or more signal treatments. The one or more signal treatments may be generated by the signal enhancer module based on analysis of the incoming audio signal. Alternatively, or in addition, characteristics of the incoming audio signal may be provided to the signal enhancer module for use in generating the one or more signal treatments. The one or more signal treatments may be added to the audio signals. | 05-29-2014 |
| 20140149127 | GENERATION OF A MODIFIED DIGITAL MEDIA FILE BASED ON AN ENCODING OF A DIGITAL MEDIA FILE WITH A DECODABLE DATA SUCH THAT THE DECODABLE DATA IS INDISTINGUISHABLE THROUGH A HUMAN EAR FROM A PRIMARY AUDIO STREAM - Disclosed are a method, a device and a system of generation of a modified digital media file based on a encoding of a digital media file with a decodable data such that the decodable data is indistinguishable through a human ear from a primary audio stream. In one embodiment, a method of an audio encoding system includes validating a user of the audio encoding system as a publisher, associating a response action to a message of the user, the response action is at least one of a call-back action, a web-form action, and a resource-page redirect action using a processor and a memory, generating a unique identifier through a hash function applied to the response action associated with the message of the user, encoding a digital media file associated with the message with a decodable data using the unique identifier such that the decodable data is indistinguishable from a primary audio stream through a human ear, and generating a modified digital media file associated with the digital media file based on the encoding of the message with the decodable data such that the decodable data is indistinguishable from a primary audio stream through the human ear. | 05-29-2014 |
| 20140156284 | AUDIO-ENCODING METHOD AND APPARATUS, AUDIO-DECODING METHOD AND APPARATUS, RECODING MEDIUM THEREOF, AND MULTIMEDIA DEVICE EMPLOYING SAME - Provided is an audio encoding method. The audio encoding method includes: acquiring envelopes based on a predetermined sub-band for an audio spectrum; quantizing the envelopes based on the predetermined sub-band; and obtaining a difference value between quantized envelopes for adjacent sub-bands and lossless encoding a difference value of a current sub-band by using a difference value of a previous sub-band as a context. Accordingly, the number of bits required to encode envelope information of an audio spectrum may be reduced in a limited bit range, thereby increasing the number of bits required to encode an actual spectral component. | 06-05-2014 |
| 20140156285 | METHOD AND APPARATUS FOR QUANTISATION INDEX MODULATION FOR WATERMARKING AN INPUT SIGNAL - With quantisation index modulation QIM it is possible to achieve a very high data rate, and the capacity of the watermark transmission is mostly independent of the characteristics of the original audio signal, but the audio quality suffers from degradation with each watermark embedding-and-removal step. In order to avoid degradation of the audio quality, the inventive audio signal watermarking uses specific quantiser curves in time domain and in particular in frequency domain for embedding the watermark message into the audio signal, whereby the processing is almost perfectly reversible. Furthermore, it has embedded a power constraint in order to guarantee that the modifications of the audio signal due to the watermark embedding are inaudible. | 06-05-2014 |
| 20140156286 | APPARATUS AND METHOD OF ENCODING AND DECODING SIGNALS - A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals. | 06-05-2014 |
| 20140156287 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique. | 06-05-2014 |
| 20140156288 | APPARATUS AND METHOD FOR SYNCHRONIZING MULTICHANNEL EXTENSION DATA WITH AN AUDIO SIGNAL AND FOR PROCESSING THE AUDIO SIGNAL - For synchronizing multichannel extension data with an audio signal, wherein the audio signal includes block division information and the multichannel extension data include reference audio signal fingerprint information, the block division information in the audio signal is detected by means of a block detector. Thereupon, block division of the audio signal is performed by a fingerprint calculator according to the block division information in order to obtain a sequence of test audio signal fingerprints. In addition to that, a sequence of reference audio signal fingerprints is extracted from the reference audio signal fingerprint information of the multichannel extension data. Both sequences of fingerprints are correlated in order to obtain a correlation result, by which a compensator is controlled in order to reduce or eliminate a time offset between the multichannel extension data and the audio signal. | 06-05-2014 |
| 20140156289 | DECODING DEVICE, DECODING METHOD, ENCODING DEVICE, ENCODING METHOD, AND PROGRAM - The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. | 06-05-2014 |
| 20140163998 | PROCESSING IN THE ENCODED DOMAIN OF AN AUDIO SIGNAL ENCODED BY ADPCM CODING - A method for processing an encoded audio signal in a binary stream by MICDA predictive coding. The method includes the following steps: determining a signal assessed from quantification indices of the binary stream; determining unencoded parameters representative of the audio signal from the assessed signal; and processing the encoded audio signal using the determined parameters. Also provided is a device implementing the method. | 06-12-2014 |
| 20140163999 | METHOD OF ENCODING AND DECODING AUDIO SIGNAL AND APPARATUS FOR ENCODING AND DECODING AUDIO SIGNAL - Exemplary embodiments may provide a method of encoding an audio signal. The method includes: segmenting the audio signal into a plurality of frames, wherein each of the frames includes M samples and M is a natural number greater than one; applying a first window, a second window, and at least one third window to the frames, wherein a length of the second window is longer than a length of the first window, and a length of the third window is longer than the length of the first window and shorter than the length of the second window; time-frequency transforming the frames to which the first window, the second window, and the at least one third window have been applied; and generating a bitstream including the time-frequency transformed frames. | 06-12-2014 |
| 20140164000 | DEVICE AND METHOD FOR GENERATING AND DECODING A SIDE CHANNEL SIGNAL TRANSMITTED WITH A MAIN CHANNEL SIGNAL - For generating a signal to be transmitted original information is encoded into a main channel and a side channel, wherein the side channel is more robust against channel influences than the main channel. On the receiver side, when the receive quality is above a threshold, which is necessitated to execute a successful decoding of the main channel, the main channel is reproduced. If the receive quality falls below this threshold, however, the side channel is reproduced which may have less bits than the main channel and which is a correspondingly lower quality representation of the original information than the main channel. | 06-12-2014 |
| 20140164001 | Method for Inter-Channel Difference Estimation and Spatial Audio Coding Device - Methods and devices for a low complex inter-channel phase difference estimation are provided. A method for the estimation of inter-channel phase differences (IPDs), comprises applying a transformation from a time domain to a frequency domain to a plurality of audio channel signals, calculating a plurality of IPD values for the IPDs between at least one of the plurality of audio channel signals and a reference audio channel signal over a predetermined frequency range, each IPD value being calculated over a portion of the predetermined frequency range, calculating, for each of the plurality of IPD values, a weighted IPD value by multiplying each of the plurality of IPD values with a corresponding frequency-dependent weighting factor, and calculating an IPD range value for the predetermined frequency range by adding the plurality of weighted IPD values. | 06-12-2014 |
| 20140172433 | ENCODING DEVICE, ENCODING METHOD, AND PROGRAM - This technology relates to an encoding device, an encoding method, and a program capable of improving audio quality and more efficiently encoding audio. A first high-frequency encoding circuit encodes a high-frequency range based on a low-frequency subband signal and a high-frequency subband signal and obtains a high-frequency code amount. A low-frequency encoding circuit encodes a low-frequency signal with a code amount determined by the high-frequency code amount and a low-frequency decoding circuit decodes the encoded low-frequency signal. A subband dividing circuit divides a decoded low-frequency signal obtained by decoding into decoded low-frequency subband signals of a plurality of subbands and a second high-frequency encoding circuit generates a high-frequency code string such that a code amount of the high-frequency code string for obtaining a high-frequency component is not larger than the high-frequency code amount based on the decoded low-frequency subband signals and the high-frequency subband signals. The present invention is applicable to the encoding device. | 06-19-2014 |
| 20140172434 | COMPRESSED DOMAIN ENCODING APPARATUS AND METHODS FOR USE WITH MEDIA SIGNALS - Apparatus, methods, and articles of manufacture for encoding a compressed media stream are disclosed. Example method of watermarking a digital media signal disclosed herein include copying compressed audio packets associated with an audio stream included in a transport stream of the digital media signal into respective frames of compressed audio data to be watermarked to include media identification information. Such example methods can also include determining whether a composition of the transport stream has changed during copying of the compressed audio packets into the respective frames of the compressed audio data. Such example methods can further include, if the composition of the transport stream has changed, writing the frames of the compressed audio data to an output stream corresponding to the digital media signal without applying a watermark to the frames of the compressed audio data. | 06-19-2014 |
| 20140172435 | Direction of Arrival Estimation Using Watermarked Audio Signals and Microphone Arrays - An apparatus for providing direction information based on a reproduced audio signal with an embedded watermark includes a signal processor, which is adapted to process at least two received watermarked audio signals recorded by at least two audio receivers at different spatial positions. The signal processor is adapted to process the received watermarked audio signals to obtain a receiver-specific information for each received watermarked audio signal. The receiver-specific information depends on the embedded watermarks embedded in the received watermarked audio signals. Moreover, the apparatus includes a direction information provider for providing direction information based on the receiver-specific information for each received watermarked audio signal. | 06-19-2014 |
| 20140188487 | METHOD AND SYSTEM FOR ROBUST AUDIO HASHING - Method and system for channel-invariant robust audio hashing, the method comprising:
| 07-03-2014 |
| 20140188488 | Reduced Complexity Converter SNR Calculation - An audio encoder configured to encode an audio signal to generate a bitstream having E-AC-3 format, including by determining a first control parameter indicative of an allocation of available mantissa bits for quantized audio content of the signal. The encoder is configured to perform transcoding simulation to determine a second control parameter in a manner based at least in part on statistical analysis of results of E-AC-3 bit allocation processing of audio data assuming a first target data rate, and of AC-3 bit allocation processing of the data assuming a second target data rate, and to include the second control parameter in the bitstream for use by a converter to convert the bitstream into a second to bitstream having AC-3 format at the second target data rate. Other aspects are converters configured to perform transcoding on a bitstream using such a second control parameter, and methods performed by any embodiment of the inventive encoder or converter. | 07-03-2014 |
| 20140188489 | Method and Apparatus for Frame-Based Buffer Control in a Communication System - A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The decoder buffer level limits are specified in terms of a maximum number of encoded frames (or duration). The transmitter can predict the number of encoded frames, F | 07-03-2014 |
| 20140188490 | METHOD AND SYSTEM FOR LOSSLESS VALUE-LOCATION ENCODING - A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L−1 remaining possible distinct data values, wherein each of the L−1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L−1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L−1 remaining possible distinct data values in the N samples, and encoding the L−1 coding vectors. | 07-03-2014 |
| 20140195253 | Audio Signal Encoder - An apparatus comprising: a coding rate determiner configured to determine a first coding bitrate for at least one first frame audio signal multi-channel parameter and a second coding bitrate for at least one second frame audio signal multi-channel parameter, wherein the combined first and second coding bitrate is less than a bitrate limit; a channel analyser configured to determine for a first frame the at least one first frame audio signal multi-channel parameter and configured to determine for a second frame the at least one second frame audio signal multi-channel parameter; a multi-channel parameter determiner configured to generate an encoded first frame audio signal multi-channel parameter within the first coding bitrate from the at least one first frame audio signal multi-channel parameter and configured to generate an encoded at least one second frame audio signal parameter within the second coding bitrate from the at least one second frame audio signal multi-channel parameter; and a multiplexer configured to combine the encoded at least one first frame audio signal multi-channel parameter and the encoded at least one second frame audio signal multi-channel parameter. | 07-10-2014 |
| 20140200899 | ENCODING DEVICE AND ENCODING METHOD, DECODING DEVICE AND DECODING METHOD, AND PROGRAM - The present technology relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program, configured to obtain a high quality audio with less encoding amount. A number-of-sections determining feature amount calculating circuit calculates a number-of-sections determining feature amount for determining the number of divisions to divide a process target section into continuous frame sections each including a frame for which the same estimation coefficient is selected, based on sub-band signals of a plurality of sub-bands constituting an input signal. A quasi-high frequency sub-band power difference calculating circuit determines the number of continuous frame sections in the process target section based on the number-of-sections determining feature amount, selects an estimation coefficient for obtaining a high frequency component of the input signal by estimation for each continuous frame section, and generates data including a coefficient index for obtaining the estimation coefficient. A high frequency encoding circuit encodes the obtained data, and generates high frequency encoded data. The present technology can be applied to an encoding device. | 07-17-2014 |
| 20140200900 | ENCODING DEVICE AND METHOD, DECODING DEVICE AND METHOD, AND PROGRAM - The present technology relates to an encoding device and method, a decoding device and method, and a program, which enable improvement of audio quality. | 07-17-2014 |
| 20140200901 | ENCODING DEVICE, DECODING DEVICE, ENCODING METHOD AND DECODING METHOD - By copying to a high-frequency band portion (extension band) a low-frequency band portion in which peaking has been set to a sufficiently low state, this encoding device is capable of preventing generation of a spectrum with overly high peaking in the high-frequency band portion, and of generating a high-quality extension band spectrum. This device comprises: a maximum value search unit which searches, in each of multiple sub-bands obtained by dividing the low-frequency band portion of an audio signal and/or music signal below a prescribed frequency, for the maximum value of the amplitude of a first spectrum obtained by decoding first encoded data, which is encoded data in the low-frequency band portion; and an amplitude normalization unit which obtains a normalized spectrum by normalizing, at the maximum values of the amplitude of each sub-band, the first spectrum contained in each sub-band. | 07-17-2014 |
| 20140207473 | REARRANGEMENT AND RATE ALLOCATION FOR COMPRESSING MULTICHANNEL AUDIO - Provided are methods and systems for rearranging a multichannel audio signal into sub-signals and allocating bit rates among them, such that compressing the sub-signals with a set of audio codecs at the allocated bit rates yields an optimal fidelity with respect to the original multichannel audio signal. Rearranging the multichannel audio signal into sub-signals and assigning each sub-signal a bit rate may be optimized according to a criterion. Existing audio codecs may be used to quantize the sub-signals at the assigned bit rates and the compressed sub-signals may be combined into the original format according to the manner in which the original multichannel audio signal is rearranged. | 07-24-2014 |
| 20140207474 | AUDIO DECELERATION - An audio receiving system includes logic configured to reduce the accumulation of delays caused by the late arrival of audio packets. This logic is configured to accelerate or decelerate presentation of a resulting audio stream in response to the detection of late packets. The acceleration is discontinued once the effects of the late packets have been compensated for. The audio receiving system is typically applied to applications in which lag is undesirable. These can include web conferencing, telepresence, and online video games. | 07-24-2014 |
| 20140214431 | SAMPLE RATE SCALABLE LOSSLESS AUDIO CODING - A transmitter in an audio coding system generates an encoded audio signal that conveys a losslessly encoded representation of an audio signal at a first sample rate and losslessly encoded representations of related audio information at other sample rates. A companion receiver with limited computational resources can generate a high-quality output audio signal at a desired sample rate by losslessly decoding the encoded representation of the audio signal and possibly other portions of the encoded audio signal as needed to obtain an output signal at one of the other sample rates. | 07-31-2014 |
| 20140214432 | DECODING DEVICE, DECODING METHOD, ENCODING DEVICE, ENCODING METHOD, AND PROGRAM - The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. | 07-31-2014 |
| 20140214433 | DECODING DEVICE, DECODING METHOD, ENCODING DEVICE, ENCODING METHOD, AND PROGRAM - The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. | 07-31-2014 |
| 20140214434 | Method for processing sound data and circuit therefor - A sound data processing apparatus includes a central processing unit for controlling predetermined processing in the apparatus, a rewritable RAM, a decoder performing the decoding processing for sound data, and an interface unit for being fitted with an external memory. The sound data processing apparatus reads a driver from the external memory mounted in the interface unit and stores the read driver into the RAM, and reads the sound data from the external memory with the driver and processes the read sound data. As a result, the wastefully using of the memory capacity of the memory mounted in the sound data processing apparatus is reduced. | 07-31-2014 |
| 20140222438 | ENCODING AND DECODING AN AUDIO WATERMARK - Embodiments of the invention are directed to systems, methods and computer program products for providing targeted location-based communications. An exemplary apparatus is configured to receive an encoded signal, decode the encoded signal such that embedded data is retrieved, send the embedded data a remote server; and receive a message based at least partially on sending the embedded data. Another exemplary apparatus is configured to provide the encoded signal by receiving data input, receiving a host signal, embedding the data input within the host signal such that an encoded signal is generated, and transmitting the encoded signal. A third exemplary apparatus is configured to provide the targeted communications by storing one or more messages associated with an entity, receiving data, selecting at least one of the one or more messages based at least partially on the data received, and sending the at least one of the one or more messages selected. | 08-07-2014 |
| 20140222439 | Apparatus and Method for Encoding/Decoding Signal - An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes skipping extension information included in an input bitstream, extracting a three-dimensional (3D) down-mix signal and spatial information from the input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using a down-mix signal obtained by the removal and the spatial information. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment. | 08-07-2014 |
| 20140222440 | METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL - A method for decoding an audio signal, receiving a downmix signal having at least one independent object and a background object downmixed therein receiving object information and enhanced object information, wherein the object information includes at least one of level information and correlation information between the independent object and the background object, wherein the enhanced object information includes a residual signal extracting the at least one independent object and the background object from the downmix signal using the object information and the enhanced object information receiving mix information from a user, the mix information being usable to control gain or panning of the independent object or the background object generating downmix processing information using at least one of the object information and enhanced object information processing at least one independent object and the background object using at least one of the downmix processing information and the mix information. | 08-07-2014 |
| 20140222441 | APPARATUS FOR GENERATING A DECORRELATED SIGNAL USING TRANSMITTED PHASE INFORMATION - An apparatus for generating a decorrelated signal having a receiving unit for receiving phase information, a transient separator, a transient decorrelator, a second decorrelator and a combining unit, wherein the transient separator is adapted to separate an input signal into a first signal component and into a second signal component such that the first signal component has transient signal portions of the input signal and such that the second signal component has non-transient signal portions of the input signal. The transient decorrelator is adapted to apply the phase information received by the receiving unit to a transient signal component. | 08-07-2014 |
| 20140236603 | AUDIO CODING DEVICE AND METHOD - An audio coding device that performs predictive coding on a third-channel signal included in a plurality of channels in an audio signal according to a first-channel signal and a second-channel signal, which are included in the plurality of channels, and to a plurality of channel prediction coefficients included in a coding book, the device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, selecting channel prediction coefficients corresponding to the first-channel signal and the second-channel signal so that an error, which is determined by a difference between the third-channel signal before predictive coding and the third-channel signal after predictive coding, is minimized; and controlling the first-channel signal or the second-channel signal so that the error is further reduced. | 08-21-2014 |
| 20140236604 | APPARATUS AND METHOD FOR GENERATING A LEVEL PARAMETER AND APPARATUS AND METHOD FOR GENERATING A MULTI-CHANNEL REPRESENTATION - A parameter representation of a multi-channel signal having several original channels includes a parameter set, which, when used together with at least one down-mix channel allows a multi-channel reconstruction. An additional level parameter is calculated such that an energy of the at least one downmix channel weighted by the level parameter is equal to a sum of energies of the original channels. The additional level parameter is transmitted to a multi-channel reconstructor together with the parameter set or together with a down-mix channel. An apparatus for generating a multi-channel representation uses the level parameter to correct the energy of the at least one transmitted down-mix channel before entering the down-mix signal into an upmixer or within the up-mixing process. | 08-21-2014 |
| 20140236605 | AUDIO ENCODER, AUDIO DECODER, METHODS FOR ENCODING AND DECODING AN AUDIO SIGNAL, AND A COMPUTER PROGRAM - An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. | 08-21-2014 |
| 20140244274 | ENCODING DEVICE AND ENCODING METHOD - An encoding device is disclosed in which frequency domain converters ( | 08-28-2014 |
| 20140249827 | SPECIFYING SPHERICAL HARMONIC AND/OR HIGHER ORDER AMBISONICS COEFFICIENTS IN BITSTREAMS - In general, techniques are described for specifying spherical harmonic coefficients in a bitstream. A device comprising one or more processors may perform the techniques. The processors may be configured to identify, from the bitstream, a plurality of hierarchical elements describing a sound field that are included in the bitstream. The processors may further be configured to parse the bitstream to determine the identified plurality of hierarchical elements. | 09-04-2014 |
| 20140249828 | Audio Encoding/Decoding based on an Efficient Representation of Auto-Regressive Coefficients - Described is an encoder ( | 09-04-2014 |
| 20140257822 | 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. | 09-11-2014 |
| 20140257823 | Real-Time Scheduling Method with Reduced Input/Output Latency and Improved Tolerance for Variable Processing Time - A method and apparatus for processing encoded audio data that operates on batches of data having a predetermined time block size. An input/output memory buffer provides a delay from input to corresponding output of 2+x time blocks where x is a predetermined constant and 0| 09-11-2014 | |
| 20140257824 | APPARATUS AND A METHOD FOR ENCODING AN INPUT SIGNAL - An apparatus and a method for encoding an input signal applied to the apparatus comprising: a transient detector adapted to detect whether the applied input signal comprises a transient; at least two transient signal encoders adapted to encode the applied input signal if a transient is detected by the transient detector; and a selection unit adapted to select a transient signal encoder among the transient signal encoders according to at least one predetermined selection criterion. | 09-11-2014 |
| 20140257825 | ENCODING APPARATUS AND ENCODING METHOD - Provided is an encoding apparatus. A threshold value calculating unit ( | 09-11-2014 |
| 20140257826 | METHOD AND APPARATUS FOR AUDIO CODING USING CONTEXT DEPENDENT INFORMATION - A method comprising determining a first geographical location; processing contextual dependent data to generate a priori information indicative of the quality of a communication channel associated with the first geographical location, wherein the contextual dependent data comprises at least one communication channel quality measure relating to the first geographical location; determining a codec function signal dependent on the a priori information indicative of the quality of the communication channel associated with the first geographical location; and encoding an audio signal according to the determined codec function signal. | 09-11-2014 |
| 20140257827 | GENERATION OF A HIGH BAND EXTENSION OF A BANDWIDTH EXTENDED AUDIO SIGNAL - An audio decoder configured to generate a high band extension of an audio signal from an envelope and an excitation. The audio decoder includes a control arrangement configured to jointly control envelope shape and excitation noisiness with a common control parameter (f). | 09-11-2014 |
| 20140278446 | DEVICE AND METHOD FOR DATA EMBEDDING AND DEVICE AND METHOD FOR DATA EXTRACTION - A data embedding device includes a storage unit configured to store a code book that includes a plurality of prediction parameters; a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, extracting a plurality of candidates, of which a prediction error in prediction coding, the prediction coding being based on signals of other two channels, of a signal of one channel among signals of a plurality of channels is within a predetermined range, of a prediction parameter from the code book and extracting the number of candidates of the prediction parameter, the candidates being extracted; converting at least part of data that is an embedding object into a number base based on the number of candidates; and selecting a prediction parameter, the prediction parameter being a result of the prediction coding, from the candidates. | 09-18-2014 |
| 20140278447 | DIGITAL WATERMARK DETECTION DEVICE AND DIGITAL WATERMARK DETECTION METHOD, AS WELL AS TAMPERING DETECTION DEVICE USING DIGITAL WATERMARK AND TAMPERING DETECTION METHOD USING DIGITAL WATERMARK - A digital watermark detection device includes a first chirp z-transform unit ( | 09-18-2014 |
| 20140288940 | METHOD AND SYSTEM FOR GENERATING AN AUDIO METADATA QUALITY SCORE - A method including the steps of assessing at least two metadata parameters associated with an audio bitstream (e.g., an encoded Dolby Digital (AC-3), Dolby Digital Plus, or Dolby E bitstream), determining individual metadata parameter quality values, each of the individual metadata parameter quality values indicative of quality (e.g., correctness) of a different one of the at least two metadata parameters, and generating data indicative of a metadata score, where the metadata score is a value determined by a combination (e.g., a linear combination or other weighted combination) of the individual metadata parameter quality values. The metadata score is indicative of overall quality (e.g., correctness) of the at least two metadata parameters. Another aspect is a system (e.g., a test device or measurement device, or another test or measurement product, or a processor) configured (e.g., programmed) to perform any embodiment of the method. | 09-25-2014 |
| 20140297290 | APPARATUS AND METHOD FOR DECODING AUDIO DATA - An apparatus and method for decoding audio data. The apparatus for decoding the audio data may perform block data unpacking by preferring a channel order to a block order from a bitstream, and perform dithering through preferring a block order to a channel order. Complexity in decoding may be reduced through integrating bitstream searching and the bock data unpacking, and a dithering error may be prevented through processing the block data unpacking and the dithering separately. | 10-02-2014 |
| 20140297291 | METADATA DRIVEN DYNAMIC RANGE CONTROL - A system for encoding and applying Dynamic Range Control/Compression (DRC) gain values to a piece of sound program content is described. In particular, a set of DRC gain values representing a DRC gain curve for the piece of content may be divided into frames corresponding to frames of the piece of content. A set of fields may be included with an audio signal representing the piece of content. The additional fields may represent the DRC gain values using linear or spline interpolation. The additional fields may include 1) an initial gain value for each DRC frame, 2) a set of slope values at particular points in the DRC curve, 3) a set of time delta values for each consecutive pair of slope values, and/or 4) one or more gain delta values representing changes of DRC gain values in the DRC gain curve between points of the slope values. | 10-02-2014 |
| 20140297292 | SYSTEM AND METHOD FOR INCREASING TRANSMISSION BANDWIDTH EFFICIENCY ("EBT2") - Systems and methods for increasing transmission bandwidth efficiency by the analysis and synthesis of the ultimate components of transmitted content are presented. To implement such a system, a dictionary or database of elemental codewords can be generated from a set of audio clips. Using such a database, a given arbitrary song or other audio file can be expressed as a series of such codewords, where each given codeword in the series is a compressed audio packet that can be used as is, or, for example, can be tagged to be modified to better match the corresponding portion of the original audio file. Each codeword in the database has an index number or unique identifier. For a relatively small number of bits used in a unique ID, e.g. 27-30, several hundreds of millions of codewords can be uniquely identified. By providing the database of codewords to receivers of a broadcast or content delivery system in advance, instead of broadcasting or streaming the actual compressed audio signal, all that need be transmitted is the series of identifiers along with any modification instructions to the identified codewords. After reception, intelligence on the receiver having access to a locally stored copy of the dictionary can reconstruct the original audio clip by accessing the codewords via the received IDs, modify them as instructed by the modification instructions, further modify the codewords either individually or in groups using the audio profile of the original audio file (also sent by the encoder) and play back a generated sequence of phase corrected codewords and modified codewords as instructed. In exemplary embodiments of the present invention, such modification can extend into neighboring codewords, and can utilize either or both (i) cross correlation based time alignment and (ii) phase continuity between harmonics, to achieve higher fidelity to the original audio clip. | 10-02-2014 |
| 20140297293 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR AVOIDING CLIPPING ARTEFACTS - An audio encoding apparatus includes an encoder for encoding a time segment of an input audio signal to be encoded to obtain a corresponding encoded signal segment. The audio encoding apparatus further includes a decoder for decoding the encoded signal segment to obtain a re-decoded signal segment. A clipping detector is provided for analyzing the re-decoded signal segment with respect to at least one of an actual signal clipping or an perceptible signal clipping and for generating a corresponding clipping alert. The encoder is further configured to again encode the time segment of the audio signal with at least one modified encoding parameter resulting in a reduced clipping probability in response to the clipping alert. | 10-02-2014 |
| 20140297294 | Methods and Apparatuses for Encoding and Decoding Object-Based Audio Signals - An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal and object-based side information, the downmix signal comprising at least two downmix channel signals; extracting gain information from the object-based side information and generating modification information for modifying the downmix channel signals on a channel-by-channel basis based on the gain information; and modifying the downmix channel signals by applying the modification information to the downmix channel signals. | 10-02-2014 |
| 20140297295 | Cross Product Enhanced Harmonic Transposition - The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal. | 10-02-2014 |
| 20140297296 | AUDIO OBJECT ENCODING AND DECODING - An audio object encoder comprises a receiver ( | 10-02-2014 |
| 20140303984 | LAYERED AUDIO CODING AND TRANSMISSION - Embodiments of systems and methods are described for generating layered audio such that computing devices can request a variable amount of data based on criteria such as their available bandwidth, device capability, or user selection. A base layer and one or more enhancement layers that incrementally enhance the previous layers may be generated. A computing device may retrieve the base layer and/or one or more enhancement layers, adjusting, in real-time or near real-time, which layers are retrieved based on fluctuations in the available bandwidth among other possible criteria. | 10-09-2014 |
| 20140303985 | METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS - Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method generating a third downmix signal by combining a first downmix signal extracted from a first audio signal and a second downmix signal extracted from a second audio signal; generating third object-based side information by combining first object-based side information extracted from the first audio signal and second object-based side information extracted from the second audio signal; converting the third object-based side information into channel-based side information; and generating a multi-channel audio signal using the third downmix signal and the channel-based side information. | 10-09-2014 |
| 20140310006 | METHOD TO GENERATE AUDIO FINGERPRINTS - It is characterised in that it comprises:
| 10-16-2014 |
| 20140310007 | METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO SIGNAL USING ADAPTIVE SINUSOIDAL CODING - A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently. | 10-16-2014 |
| 20140310008 | METHOD OF MANAGING A JITTER BUFFER, AND JITTER BUFFER USING SAME - The present invention relates to a method of managing a jitter buffer and a jitter buffer using same. The method of managing a jitter buffer includes the steps of: receiving audio information frames; and adjusting a jitter buffer on the basis of the received audio information frames, wherein the adjusting step of the jitter buffer includes compensation of an audio signal, and the compensation of the audio signal can be performed for each sub frame of the audio information frames. | 10-16-2014 |
| 20140310009 | SIGNAL CODEC DEVICE AND METHOD IN COMMUNICATION SYSTEM - The present invention relates to a codec device and method for encoding/decoding voice and audio signals in a communication system, wherein: a fixed codebook excited signal is generated by using a pulse index for a voice signal; a first adaptive codebook excited signal is generated by using a pitch index for the voice signal; a fixed codebook signal is generated by multiplying the fixed codebook excited signal by a fixed codebook gain; a first adaptive codebook signal is generated by multiplying the first adaptive codebook excited signal by a first adaptive codebook gain; and a synthesized filter excited signal is generated by adding the fixed codebook signal and the first adaptive codebook signal. | 10-16-2014 |
| 20140310010 | APPARATUS FOR ENCODING AND APPARATUS FOR DECODING SUPPORTING SCALABLE MULTICHANNEL AUDIO SIGNAL, AND METHOD FOR APPARATUSES PERFORMING SAME - An encoding apparatus and a decoding apparatus supporting a scalable multichannel audio signal, and methods performed by the apparatuses art provided. When compressing and decompressing a multichannel audio signal to compress and reproduce high quality 3-dimensional (3D) audio, the apparatuses and the methods in integrated form of (1) a sound quality scalability function for providing various qualities of audio adaptively to a transmission environment, terminal performance, and a listening environment (2) a channel scalability function for providing multichannel signals of various formats adaptively to the transmission environment, the terminal performance, and a reproduction environment of a terminal, such as speaker arrangement, and (3) an object scalability function for independently controlling a particular audio object to maximize a 3D sound field effect. | 10-16-2014 |
| 20140310011 | Enhanced Chroma Extraction from an Audio Codec - The present document relates to methods and systems for music information retrieval (MIR). In particular, the present document relates to methods and systems for extracting a chroma vector from an audio signal. A method ( | 10-16-2014 |
| 20140316788 | QUALITY IMPROVEMENT TECHNIQUES IN AN AUDIO ENCODER - An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits. | 10-23-2014 |
| 20140316789 | SYSTEMS AND METHODS FOR IMPLEMENTING CROSS-FADING, INTERSTITIALS AND OTHER EFFECTS DOWNSTREAM - Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element. The next elements(s) can, for example, be stored in an input buffer, then decoded and stored in a decoded sample buffer, all prior to the required presentation time of the multiple element effect. At the requisite time, a client device component can access the respective samples of the decoded audio clips as it performs the cross-fade, mix or other effect. Such exemplary embodiments use a single decoder and thus do not require synchronized simultaneous decodes. | 10-23-2014 |
| 20140324440 | Detection of an Audio Signal Transient Using First and Second Maximum Norms - Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment. | 10-30-2014 |
| 20140324441 | METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION - A method for determining mantissa bit allocation of audio data values of frequency domain audio data to be encoded. The allocation method includes a step of determining masking values for the audio data values, including by performing adaptive low frequency compensation on the audio data of each frequency band of a set of low frequency bands of the audio data. The adaptive low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set of low frequency bands has prominent tonal content; and performing low frequency compensation on the audio data in each frequency band in the set of low frequency bands having prominent tonal content as indicated by the compensation control data, but not performing low frequency compensation on the audio data in any other frequency band in the set of low frequency bands. | 10-30-2014 |
| 20140337038 | METHOD, APPLICATION, AND DEVICE FOR AUDIO SIGNAL TRANSMISSION - The current invention discloses methods, applications, and devices for audio transmission from a mobile terminal. After receiving an audio signal transmission request from a user, the mobile terminal may initiate a recording session to record audio signals into audio frames. During the recording session, the terminal may adjust the audio codecs used for encoding the audio frames based on the workload and the performance of the terminal. By measuring and evaluating the encoding time, the terminal may change between using a floating-point AMR audio codec and a fixed-point AMR audio codec. The encoded audio frames are transmitted to a remote server. The current invention provides a flexible and efficient approach for audio signal encoding and transmission, balancing signal integrity and encoding speed at the same time. | 11-13-2014 |
| 20140337039 | Frame Loss Compensation Method And Apparatus For Voice Frame Signal - A frame loss compensation method and apparatus for audio signals are disclosed. The method includes: when a first frame immediately following a correctly received frame is lost, judging a frame type of the first lost frame, and when the first lost frame is a non-multi-harmonic frame, calculating MDCT coefficients of the first lost frame by using MDCT coefficients of one or more frames prior to the first lost frame; obtaining an initially compensated signal of the first lost frame according to the MDCT coefficients of the first lost frame; and performing a first class of waveform adjustment on the initially compensated signal of the first lost frame and taking an adjusted time-domain signal as a time-domain signal of the first lost frame. The apparatus includes a frame type judgment module, an MDCT coefficient acquisition module, an initial compensation signal acquisition module and an adjustment module. | 11-13-2014 |
| 20140343953 | MULTI-MODE AUDIO CODEC AND CELP CODING ADAPTED THEREFORE - In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal. | 11-20-2014 |
| 20140343954 | BINAURAL MULTI-CHANNEL DECODER IN THE CONTEXT OF NON-ENERGY-CONSERVING UPMIX RULES - A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained. | 11-20-2014 |
| 20140350944 | ENCODING AND REPRODUCTION OF THREE DIMENSIONAL AUDIO SOUNDTRACKS - The present invention provides a novel end-to-end solution for creating, encoding, transmitting, decoding and reproducing spatial audio soundtracks. The provided soundtrack encoding format is compatible with legacy surround-sound encoding formats, so that soundtracks encoded in the new format may be decoded and reproduced on legacy playback equipment with no loss of quality compared to legacy formats. | 11-27-2014 |
| 20140358554 | AUDIO ENCODING METHOD AND SYSTEM FOR GENERATING A UNIFIED BITSTREAM DECODABLE BY DECODERS IMPLEMENTING DIFFERENT DECODING PROTOCOLS - In a class of embodiments, an audio encoding system (typically, a perceptual encoding system that is configured to generate a single (“unified”) bitstream that is compatible with (i.e., decodable by) a first decoder configured to decode audio data encoded in accordance with a first encoding protocol (e.g., the multichannel Dolby Digital Plus, or DD+, protocol) and a second decoder configured to decode audio data encoded in accordance with a second encoding protocol (e.g., the stereo AAC, HE AAC v1, or HE AAC v2 protocol). The unified bitstream can include both encoded data (e.g., bursts of data) decodable by the first decoder (and ignored by the second decoder) and encoded data (e.g., other bursts of data) decodable by the second decoder (and ignored by the first decoder). In effect, the second encoding format is hidden within the unified bitstream when the bitstream is decoded by the first decoder, and the first encoding format is hidden within the unified bitstream when the bitstream is decoded by the second decoder. The format of the unified bitstream generated in accordance with the invention may eliminate the need for transcoding elements throughout an entire media chain and/or ecosystem. Other aspects of the invention are an encoding method performed by any embodiment of the inventive encoder, a decoding method performed by any embodiment of the inventive decoder, and a computer readable medium (e.g., disc) which stores code for implementing any embodiment of the inventive method. | 12-04-2014 |
| 20140358555 | Periodic Ambient Waveform Analysis for Enhanced Social Functions - In particular embodiments, one or more computer-readable non-transitory storage media embody software that is operable when executed to receive an audio waveform fingerprint and a client-determined location from a client device. The received audio waveform fingerprint may be compared to a database of stored audio waveform fingerprints, each stored audio waveform fingerprint associated with an object in an object database. One or more matching audio waveform fingerprints may be found from a comparison set of audio waveform fingerprints obtained from the audio waveform fingerprint database. Location information associated with a location of the client device may be determined, and the location information may be sent to the client device. The client device may be operable to update the client-determined location based at least in part on the location information. | 12-04-2014 |
| 20140358556 | AUDIO DECOMPRESS PROGRAM - A method of decompressing audio comprises normalizing the over compressed audio at a percentage of 48. Next the audio is converted from 16 to 32 bit to allow the filter a better resolution source to process. The next step is processing the converted ausion with a Chebyshev filter, preferably with a range of 4 Hz to 22050 Hz (bandpass mode) at the order of 18 (@36 dB estimated. After the filter, the audio is converted back to 16 bit with the dither setting shown in the block. Now, the end result is a newly repaired piece of audio with the dynamic and harmonic content restored. | 12-04-2014 |
| 20140358557 | PERFORMING POSITIONAL ANALYSIS TO CODE SPHERICAL HARMONIC COEFFICIENTS - In general, techniques are described for performing a positional analysis to code audio data. Typically, this audio data comprises a hierarchical representation of a soundfield and may include, as one example, spherical harmonic coefficients (which may also be referred to as higher-order ambisonic coefficients). An audio compression device that includes one or more processors may perform the techniques. The processors may be configured to allocate bits to one or more portions of the audio data, at least in part by performing positional analysis on the audio data. | 12-04-2014 |
| 20140358558 | IDENTIFYING SOURCES FROM WHICH HIGHER ORDER AMBISONIC AUDIO DATA IS GENERATED - In general, techniques are described for obtaining an indication of whether spherical harmonic coefficients are representative of a synthetic audio object. In accordance with the techniques, a device comprising one or more processors may be configured to obtain an indication of whether spherical harmonic coefficients representative of a sound field are generated from a synthetic audio object. | 12-04-2014 |
| 20140358559 | COMPENSATING FOR ERROR IN DECOMPOSED REPRESENTATIONS OF SOUND FIELDS - In general, techniques are described for compensating for error in decomposed representations of sound fields. In accordance with the techniques, a device comprising one or more processors may be configured to quantize one or more first vectors representative of one or more components of a sound field, and compensate for error introduced due to the quantization of the one or more first vectors in one or more second vectors that are also representative of the same one or more components of the sound field. | 12-04-2014 |
| 20140358560 | PERFORMING ORDER REDUCTION WITH RESPECT TO HIGHER ORDER AMBISONIC COEFFICIENTS - In general, techniques are described for performing order reduction with respect to a plurality of spherical harmonic coefficients. In accordance with the techniques, a device comprising one or more processors may be configured to perform, based on a target bitrate, order reduction with respect to a plurality of spherical harmonic coefficients or decompositions thereof to generate reduced spherical harmonic coefficients or the reduced decompositions thereof, wherein the plurality of spherical harmonic coefficients represent a sound field. | 12-04-2014 |
| 20140358561 | IDENTIFYING CODEBOOKS TO USE WHEN CODING SPATIAL COMPONENTS OF A SOUND FIELD - In general, techniques are described for identifying a codebook to be used when compressing spatial components of a sound field. A device comprising one or more processors may be configured to perform the techniques. The one or more processors may be configured to identify a Huffman codebook to use when compressing a spatial component of a plurality of spatial components based on an order of the spatial component relative to remaining ones of the plurality of spatial components, the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients. | 12-04-2014 |
| 20140358562 | QUANTIZATION STEP SIZES FOR COMPRESSION OF SPATIAL COMPONENTS OF A SOUND FIELD - In general, techniques are described for determining quantization step sizes for compression of spatial components of a sound field. A device comprising one or more processors may be configured to perform the techniques. In other words, the one or more processors may be configured to determine a quantization step size to be used when compressing a spatial component of a sound field, where the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients. | 12-04-2014 |
| 20140358563 | COMPRESSION OF DECOMPOSED REPRESENTATIONS OF A SOUND FIELD - In general, techniques are described for compressing decomposed representations of a sound field. A device comprising one or more processors may be configured to perform the techniques. The one or more processors may be configured to obtain a bitstream comprising a compressed version of a spatial component of a sound field, the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients. | 12-04-2014 |
| 20140358564 | INTERPOLATION FOR DECOMPOSED REPRESENTATIONS OF A SOUND FIELD - In general, techniques are described for performing an interpolation with respect to decomposed versions of a sound field. A device comprising one or more processors may be configured to perform the techniques. The processors may be configured to obtain decomposed interpolated spherical harmonic coefficients for a time segment by, at least in part, performing an interpolation with respect to a first decomposition of a first plurality of spherical harmonic coefficients and a second decomposition of a second plurality of spherical harmonic coefficients. | 12-04-2014 |
| 20140358565 | COMPRESSION OF DECOMPOSED REPRESENTATIONS OF A SOUND FIELD - In general, techniques are described for obtaining decomposed versions of spherical harmonic coefficients. A device comprising one or more processors may be configured to perform the techniques, whereby the processors may be configured to obtain, from a bitstream, at least one of one or more vectors decomposed from spherical harmonic coefficients that were recombined with background spherical harmonic coefficients, wherein the spherical harmonic coefficients describe a sound field, and wherein the background spherical harmonic coefficients described one or more background components of the same sound field. | 12-04-2014 |
| 20140358566 | METHODS AND DEVICES FOR AUDIO PROCESSING - An audio processing method for use in a server, includes: receiving an audio file uploaded from a terminal that has downloaded a first accompaniment music file of a song from the server, the audio file being generated by the terminal by encoding collected audio information relating to singing a portion of the song and the first accompaniment music file; and marking an unmarked, audio mixing portion in the received audio file as a portion that has been sung, to generate a second accompaniment music file of the song. | 12-04-2014 |
| 20140358567 | SPATIAL AUDIO RENDERING AND ENCODING - An encoder ( | 12-04-2014 |
| 20140365230 | METHOD AND APPARATUS FOR MAXIMIZING A LIMITED SET OF IDENTIFIERS FOR AUDIO WATERMARKING - A method and apparatus for encoding identifiers into a media item, the method including: embedding a sequence of identifiers equally and in order over a duration of a content of the media item; and storing a mapping between the embedded identifiers and the media item; wherein the sequence of identifiers is created by: (a) generating a subsequence of 2M unique identifiers, M being an integer greater than one; (b) shifting every other identifiers in the generated subsequence to the right by two positions in the subsequence and with the identifier at the end cycling around to the corresponding position at the beginning to generate a new subsequence; (c) repeating step (b) up to M−1 times; and (d) concatenating the generated subsequences to create the sequence of identifiers. | 12-11-2014 |
| 20140365231 | UPSAMPLING USING OVERSAMPLED SBR - An encoder (250) comprises a core encoder (252) for encoding a low frequency component of the audio signal at the signal sampling rate (fs_in) and a spectral band replication-referred to as SBR-encoding unit (153, 254) for determining a plurality of SBR parameters. A plurality of the SBR parameters is determined such that a high frequency component of the audio signal can be approximated based on the low frequency component of the audio signal and the plurality of SBR parameters. A multiplexer (155) is adapted to generate an overall bitstream comprising the core encoded bitstream, the plurality of SBR parameters and an indication of one or more SBR encoder settings applied by the SBR encoder (153, 254); wherein the generated overall bitstream does not indicate that the core encoded bitstream has been determined by encoding the low frequency component at the signal sampling rate (fs_in). | 12-11-2014 |
| 20140372130 | DEVICE AND METHOD FOR ENCODING AND DECODING MULTICHANNEL SIGNAL - Provided is an apparatus and method for converting a | 12-18-2014 |
| 20140372131 | PHASE COHERENCE CONTROL FOR HARMONIC SIGNALS IN PERCEPTUAL AUDIO CODECS - A decoder for decoding an encoded audio signal to obtain a phase-adjusted audio signal is provided. The decoder has a decoding unit and a phase adjustment unit. The decoding unit is adapted to decode the encoded audio signal to obtain a decoded audio signal. The phase adjustment unit is adapted to adjust the decoded audio signal to obtain the phase-adjusted audio signal. The phase adjustment unit is configured to receive control information depending on a vertical phase coherence of the encoded audio signal. Moreover, the phase adjustment unit is adapted to adjust the decoded audio signal based on the control information. | 12-18-2014 |
| 20140379355 | MULTIBAND COMPRESSOR - In a multiband compressor | 12-25-2014 |
| 20150012281 | SYSTEM AND METHOD FOR CONTROLLING AUDIO DATA PROCESSING - An audio accelerator includes a decoder to decode first and second sets of data blocks, a processor to process the first and second sets of decoded data blocks, a storage area to store the first and second sets of processed data blocks, and a controller to generate interrupt signals for controlling operation of the decoder. The controller may control a rate at which data blocks are to be decoded by the decoder to reduce a time gap between outputting adjacent ones of the data blocks from the first and second sets in the storage area. | 01-08-2015 |
| 20150012282 | Processing Multichannel Audio Signals - In order to enable audio applications for a mobile device that utilize more than one audio input, a peripheral audio device is provided for encoding multichannel audio signals into a reduced number of channels. The peripheral audio device receives audio signals from an audio input/output device, generates at least one output audio signal by combining the received audio signals, and transmits the at least one generated output audio signal to the mobile device. The number of received audio signals is greater than the number of generated output audio signals. | 01-08-2015 |
| 20150019230 | Method and Apparatus for Sending Information Codes with Audio Signals and Obtaining Information Codes - The invention relates to a method and apparatus for sending an information code via an audio signal and retrieving an information code. The method of sending an information code via an audio signal includes the steps of: modulating the information code and a sync header into the audio signal in a second band; and transmitting the modulated audio signal in the second band directly, or superimposing and then transmitting the modulated audio signal in the second band with an audio signal in a first band, wherein the first band is different from the second band. | 01-15-2015 |
| 20150025894 | METHOD FOR ENCODING AND DECODING OF MULTI CHANNEL AUDIO SIGNAL, ENCODER AND DECODER - Provided are a method of encoding and decoding a multichannel audio signal, and an encoder and a decoder to perform the method. The present invention may perform encoding into consideration of a size of bit to be allocated based on a feature of audio signal for each channel with respect to an audio signal having a plurality of channels, thereby enhancing an encoding efficiency of the multichannel audio signal. | 01-22-2015 |
| 20150025895 | Audio Encoder with Parallel Architecture - The present document relates to methods and systems for audio encoding. In particular, the present document relates to methods and systems for fast audio encoding using a parallel system architecture. A frame-based audio encoder ( | 01-22-2015 |
| 20150025896 | Enabling Sampling Rate Diversity In A Voice Communication System - An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams. | 01-22-2015 |
| 20150025897 | 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. | 01-22-2015 |
| 20150032461 | Subband Block Based Harmonic Transposition - The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank ( | 01-29-2015 |
| 20150039320 | Systems Apparatus and Methods for Encoding/Decoding Persistent Universal Media Codes to Encoded Audio - Apparatus, system and method for encoding and/or decoding persistent universal media identification (ID) codes embedded in audio. For encoding, a persistent identifier code is generated or received from a registry database, where the code includes data for uniquely identifying a media object. Audio code components including frequency characteristics are generated to represent symbols of the persistent identifier code and the audio code components are psychoacoustically embedded into an audio portion of the media object to include the persistent identifier code within one or more of a plurality of encoding layers. Such embedded audio may be subsequently decoded by transforming the audio data into a frequency domain and processing the transformed audio data to detect the persistent identifier code. | 02-05-2015 |
| 20150039321 | Apparatus, System and Method for Reading Codes From Digital Audio on a Processing Device - Apparatus, system and method for reading ancillary code embedded into digital audio, where a processing device executes a decoder application that includes a decoder application interface that is communicatively coupled to a media player application within one or more frameworks of the processing device. As digital audio is received and sampled, the decoder application is configured to transform the digital audio from a time domain to a frequency domain and to process frequency characteristics to determine the presence of ancillary audio codes. | 02-05-2015 |
| 20150039322 | APPARATUS, SYSTEM AND METHOD FOR MERGING CODE LAYERS FOR AUDIO ENCODING AND DECODING - Apparatus, system and method for encoding and decoding ancillary code for digital audio, where multiple encoding layers are merged. The merging allows a greater number of ancillary codes to be embedded into the encoding space, and further introduces efficiencies in the encoding process. | 02-05-2015 |
| 20150039323 | ENCODING AND DECODING SYSTEM, DECODING APPARATUS, ENCODING APPARATUS, ENCODING AND DECODING METHOD - An encoding and decoding system includes: a characteristic determining unit which determines whether a sound signal is a speech signal or an audio signal; an encoder which encodes the sound signal into an encoded signal, based on a determination by the characteristic determining unit; a transmitting unit which transmits the encoded signal; a receiving unit which receives the encoded signal; a decoder which decodes the encoded signal; and a packet loss detecting unit which detects a loss of data of the encoded signal and transmits a notification indicating the loss of the data to the characteristic determining unit. Upon receiving the notification indicating the loss of the data, the characteristic determining unit causes the encoder to encode the sound signal portion into a signal portion composed of independently decodable frames. | 02-05-2015 |
| 20150046171 | Transform Encoding/Decoding of Harmonic Audio Signals - An encoder ( | 02-12-2015 |
| 20150046172 | ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM AND RECORDING MEDIUM - A frequency-domain sample interval corresponding to a time-domain pitch period L corresponding to a time-domain pitch period code of an audio signal in a given time period is obtained as a converted interval T | 02-12-2015 |
| 20150051914 | DYNAMIC DECODING OF COMMUNICATION BETWEEN CARD READER AND PORTABLE DEVICE - The proposed technology generally relates the field of data transmission, in particular it relates to decoding an encoded data signal received at an audio interface of a portable electronic device, wherein the encoded data signal is encoded with an encoding scheme having an adjustable encoder clock frequency. The proposed method comprises pre-processing the received encoded data signal; scanning the received encoded data signal for a known start sequence and when a known start sequence is successfully detected then calculating an actual frequency based on the detected start sequence; interpreting, a data block succeeding the start sequence using the assessed actual frequency; and assessing whether to request adjustment of the adjustable encoder clock frequency based on the scanning and/or the interpretation. The proposed technology relates to a method performed in a portable communications device well as a corresponding device and computer program. | 02-19-2015 |
| 20150058024 | WIRELESS CELLULAR TELEPHONE WITH AUDIO CODEC - A wireless cellular telephone with an audio codec for converting digital audio signals to analog audio signals. The audio codec comprises two digital audio bus interfaces for coupling to respective digital audio buses, and a digital-only signal path between the two digital audio bus interfaces, such that no analog processing of the audio signals occurs in the digital-only signal path. | 02-26-2015 |
| 20150058025 | Oversampling in a Combined Transposer Filterbank - The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank ( | 02-26-2015 |
| 20150066517 | RESOLUTION-INDEPENDENT DITHER SAMPLE INSERTION FOR AUDIO TRANSMISSIONS - Methods, systems, and apparatuses are provided for resolution-independent dither sample insertion for audio transmissions. Audio transmitters transmit audio data streams to audio receivers. Data inactivity in the data stream triggers the audio receivers to enter power-saving/sleep modes in which synchronization with the audio transmitters is lost thus requiring time consuming resynchronization. Inactivity such as silence or zero-value data is detected in a data stream. Upon detection, a dither sample is inserted into the data stream and transmitted to the audio receiver to prevent the loss of synchronization. The dither sample may have a negative value and be formatted for resolution-independence with respect to the audio receiver. | 03-05-2015 |
| 20150066518 | AUDIO ENCODING APPARATUS AND METHOD, AUDIO DECODING APPARATUS AND METHOD, AND AUDIO REPRODUCING APPARATUS - An audio encoding apparatus and method that encodes hybrid contents including an object sound, a background sound, and metadata, and an audio decoding apparatus and method that decodes the encoded hybrid contents are provided. The audio encoding apparatus may include a mixing unit to generate an intermediate channel signal by mixing a background sound and an object sound, a matrix information encoding unit to encode matrix information used for the mixing, an audio encoding unit to encode the intermediate channel signal, and a metadata encoding unit to encode metadata including control information of the object sound. | 03-05-2015 |
| 20150073812 | SERVER SIDE CROSSFADING FOR PROGRESSIVE DOWNLOAD MEDIA - In exemplary embodiments of the present invention systems and methods are provided to implement and facilitate cross-fading, interstitials and other effects/processing of two or more media elements in a personalized media delivery service so that each client or user has a consistent high quality experience. The effects or crossfade processing can occur on the broadcast, publisher or server-side, but can still be personalized to a specific user, thus still allowing a personalized experience for each individual user, in a manner where the processing burden is minimized on the downstream side or client device. This approach enables a consistent user experience, independent of client device capabilities, both static and dynamic. The cross-fade can be implemented after decoding the relevant chunks of each component clip, processing, recoding and rechunking, or, in a preferred embodiment, the cross-fade or other effect can be implemented on the relevant chunks to the effect in the compressed domain, thus obviating any loss of quality by re-encoding. A large scale personalized content delivery service can be implemented by limiting the processing to essentially the first and last chunks of any file, since there is no need to processing the full clip. In exemplary embodiments of the present invention this type of processing can easily be accommodated in cloud computing technology, where the first and last files may be conveniently extracted and processed within the cloud to meet the required load. Processing may also be done locally, for example, by the broadcaster, with sufficient processing power to manage peak load. | 03-12-2015 |
| 20150081310 | METHOD AND APPARATUS FOR DECODING STEREO LOUDSPEAKER SIGNALS FROM A HIGHER-ORDER AMBISONICS AUDIO SIGNAL - Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA. The desired panning functions can be derived from a panning law for placement of virtual sources between the loudspeakers. For each loudspeaker a desired panning function for all possible input directions at sampling points is defined. The panning functions are approximated by circular harmonic functions, and with increasing Ambisonics order the desired panning functions are matched with decreasing error. For the frontal region between the loudspeakers, a panning law like the tangent law or vector base amplitude panning (VBAP) are used. For the rear directions panning functions with a slight attenuation of sounds from these directions are defined. | 03-19-2015 |
| 20150081311 | MEDIA SYNCHRONISATION SYSTEM - A communications system distributes code word pairs within the audio of a television or radio program or the like. Each pair of code words includes an ID code word that is the same for a given program and a synchronization code word that is unique within the program. A portable user device is able to synchronize itself to the program using the embedded synchronization code words. | 03-19-2015 |
| 20150081312 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AND AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A MODIFICATION OF A NUMBER REPRESENTATION OF A NUMERIC PREVIOUS CONTEXT VALUE - An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value, and determines the numeric current context value in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder modifies a number representation of a numeric previous context value, describing a context state associated with one or more previously decoded spectral values, in dependence on a context subregion value, to acquire a number representation of a numeric current context value describing a context state associated with one or more spectral values to be decoded. An audio encoder uses a similar concept. | 03-19-2015 |
| 20150088527 | BANDWIDTH EXTENSION OF HARMONIC AUDIO SIGNAL - Methods and arrangements in a codec for supporting bandwidth extension, BWE, of an harmonic audio signal. The method in the decoder part of the codec comprises receiving a plurality of gain values associated with a frequency band b and a number of adjacent frequency bands of band b. The method further comprises determining whether a reconstructed corresponding frequency band b′ comprises a spectral peak. When the band b′ comprises a spectral peak, a gain value associated with the band b′ is set to a first value based on the received plurality of gain values; and otherwise the gain value is set to a second value based on the received plurality of gain values. The suggested technology enables bringing gain values into agreement with peak positions in a bandwidth extended frequency region. | 03-26-2015 |
| 20150088528 | DECODING APPARATUS AND METHOD, AUDIO SIGNAL PROCESSING APPARATUS AND METHOD, AND PROGRAM - The present technique relates to a decoding apparatus and method, an audio signal processing apparatus and method, and a program that enable generation of an interpolation signal with less incongruity through a smaller amount of calculation. | 03-26-2015 |
| 20150088529 | ENCODING METHOD, ENCODER, PROGRAM AND RECORDING MEDIUM - A value of gain is updated so that the greater the difference between the number of bits or estimated number of bits in a code obtained by encoding a string of integer value samples obtained by dividing each sample in a sample string derived from an input audio signal in a given interval by gain before the update and a predetermined number B of allocated bits, the greater the difference between the gain before the update and the updated gain. A gain code corresponding to the updated gain and an integer signal code obtained by encoding a string of integer value samples obtained by dividing each sample in the sample string by the gain are obtained. | 03-26-2015 |
| 20150088530 | Method and Apparatus for Decoding an Audio Signal - Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes extracting a downmix signal and spatial information from a received audio signal, generating surround converting information using the spatial information and rendering the downmix signal to generate a pseudo-surround signal in a previously set rendering domain, using the surround converting information. The apparatus for decoding an audio signal includes a demultiplexing part extracting a downmix signal and spatial information from a received audio signal, an information converting part generating surround converting information using the spatial information and a pseudo-surround generating part rendering the downmix signal to generate a pseudo-surround signal in a previous set rendering domain, using the surround converting information. | 03-26-2015 |
| 20150095038 | SPEECH/AUDIO SIGNAL PROCESSING METHOD AND CODING APPARATUS - The present disclosure provides a speech/audio signal processing method based on wideband switching and a coding apparatus. The method includes: if a first wideband speech/audio signal is a harmonic signal, adjusting a determining condition for determining that a second wideband speech/audio signal is a harmonic signal, to obtain a first determining condition, where the first wideband speech/audio signal is a signal before wideband switching, and the second wideband speech/audio signal is a signal after the wideband switching; and determining, according to the first determining condition, whether the second wideband speech/audio signal is a harmonic signal. In the case of wideband switching, signal types of speech/audio signals remain as consistent as possible before and after the switching, so that continuity of the speech/audio signal decoded by a decoder device is ensured as much as possible, further improving speech communication service quality. | 04-02-2015 |
| 20150095039 | 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. | 04-02-2015 |
| 20150095040 | OBJECT-BASED AUDIO-VISUAL TERMINAL AND BITSTREAM STRUCTURE - As information to be processed at an object-based video or audio-visual (AV) terminal, an object-oriented bitstream includes objects, composition information, and scene demarcation information. Such bitstream structure allows on-line editing, e.g. cut and paste, insertion/deletion, grouping, and special effects. In the interest of ease of editing, AV objects and their composition information are transmitted or accessed on separate logical channels (LCs). Objects which have a lifetime in the decoder beyond their initial presentation time are cached for reuse until a selected expiration time. The system includes a de-multiplexer, a controller which controls the operation of the AV terminal, input buffers, AV objects decoders, buffers for decoded data, a composer, a display, and an object cache. | 04-02-2015 |
| 20150100324 | AUDIO ENCODER PERFORMANCE FOR MIRACAST - A method for encoding audio comprises receiving an unencoded audio signal and monitoring a user interface for user interface events. The method continues by selecting one of a plurality of transform windows to hold a defined quantity of audio samples based upon a detected one or more user interface interaction events and associated transient information. The plurality of transform windows comprises a long window sequence comprising a single window with a first quantity of samples, and a short window sequence comprising a plurality of second windows each comprising a second quantity of samples. A sum of samples of the plurality of second windows equals the first plurality of samples. The short window sequence is selected when a particular user interface interaction event is received from the user interface. | 04-09-2015 |
| 20150100325 | DIGITAL AUDIO TRANSMITTER AND RECEIVER - A method for increasing the fidelity of digitally encoded audio, comprising interleaving the signal, frequency conversion, and polynomial interpolation along with comparison to a second, redundant signal. | 04-09-2015 |
| 20150106106 | SYSTEMS AND METHODS OF COMMUNICATING REDUNDANT FRAME INFORMATION - A method includes receiving a second audio frame at a decoder. The second audio frame follows a first audio frame in an audio signal and includes a first number of bits allocated to primary coding information associated with the second audio frame, a second number of bits allocated to redundant coding information associated with the first audio frame, and an indicator of a frame type of the first audio frame. In response to a frame erasure condition associated with the first audio frame, the second number of bits is determined based on the indicator and used to decode the first audio frame. In clean channel conditions, the first audio frame is received and decoded based on primary coding bits in the first audio frame, and the first number of bits is determined based on the indicator and used to decode the second audio frame. | 04-16-2015 |
| 20150106107 | SYSTEMS AND METHODS OF ENERGY-SCALED SIGNAL PROCESSING - A method includes determining a first modeled high-band signal based on a low-band excitation signal of an audio signal, where the audio signal includes a high-band portion and a low-band portion. The method also includes determining scaling factors based on energy of sub-frames of the first modeled high-band signal and energy of corresponding sub-frames of the high-band portion of the audio signal. The method includes applying the scaling factors to a modeled high-band excitation signal to determine a scaled high-band excitation signal and determining a second modeled high-band signal based on the scaled high-band excitation signal. The method includes determining gain parameters based on the second modeled high-band signal and the high-band portion of the audio signal. | 04-16-2015 |
| 20150106108 | LINEAR PREDICTION BASED AUDIO CODING USING IMPROVED PROBABILITY DISTRIBUTION ESTIMATION - Linear prediction based audio coding is improved by coding a spectrum composed of a plurality of spectral components using a probability distribution estimation determined for each of the plurality of spectral components from linear prediction coefficient information. The linear prediction coefficient information is available anyway. Accordingly, it may be used for determining the probability distribution estimation at both encoding and decoding side. The latter determination may be implemented in a computationally simple manner by using, for example, an appropriate parameterization for the probability distribution estimation at the plurality of spectral components. The coding efficiency as provided by the entropy coding is compatible with probability distribution estimations as achieved using context selection, but its derivation is less complex. The derivation may be purely analytically and/or does not require any information on attributes of neighboring spectral lines such as previously coded/decoded spectral values of neighboring spectral lines as is the case in spatial context selection. | 04-16-2015 |
| 20150112692 | APPARATUS AND METHOD FOR EXTENDING BANDWIDTH OF SOUND SIGNAL - Disclosed is an apparatus for extending a bandwidth of a sound signal. The apparatus includes a database that stores predetermined training information as a result of at least one of Gaussian mixture model (GMM) training and hidden Markov model (HMM) training; a modified discrete cosine transform (MDCT) transformer that transforms a first band signal through MDCT, a feature extractor that extracts a feature parameter of the first band signal from an MDCT coefficient output from the MDCT transformer; an extender that provides an extended MDCT coefficient for a second band signal based on the MDCT coefficient of the first band signal output from the MDCT transformer, a subband energy estimator that estimates subband energy of the second band signal with reference to information stored in the database based on the feature parameter. | 04-23-2015 |
| 20150112693 | AUDIO ENCODER, AUDIO DECODER, METHODS FOR ENCODING AND DECODING AN AUDIO SIGNAL, AND A COMPUTER PROGRAM - An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. | 04-23-2015 |
| 20150120306 | METHOD AND APPARATUS FOR QUADRATURE MIRROR FILTERING - A method of performing quadrature mirror filter (QMF) synthesis filtering includes recording new samples corresponding to a current time slot at positions of samples to be discarded in a first array that includes modulated QMF sub-band samples. The method further includes extracting samples from the first array to remove aliasing between adjacent sub-bands, determining filter coefficients corresponding to the extracted samples by using modulo operation, and synthesizing a time domain sample where aliasing is removed by using the extracted samples and the filter coefficients. | 04-30-2015 |
| 20150120307 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD, ENCODER AND ENCODING METHOD, DECODER AND DECODING METHOD, AND PROGRAM - The present invention relates to a signal processing apparatus and a signal processing method, an encoder and an encoding method, a decoder and a decoding method, and a program capable of reproducing music signal having a better sound quality by expansion of frequency band. | 04-30-2015 |
| 20150120308 | Computationally-Assisted Musical Sequencing and/or Composition Techniques for Social Music Challenge or Competition - An application that manipulates audio (or audiovisual) content, automated music creation technologies may be employed to generate new musical content using digital signal processing software hosted on handheld and/or server (or cloud-based) compute platforms to intelligently process and combine a set of audio content captured and submitted by users of modern mobile phones or other handheld compute platforms. The user-submitted recordings may contain speech, singing, musical instruments, or a wide variety of other sound sources, and the recordings may optionally be preprocessed by the handheld devices prior to submission. | 04-30-2015 |
| 20150120309 | ERROR CONCEALMENT METHOD AND APPARATUS FOR AUDIO SIGNAL AND DECODING METHOD AND APPARATUS FOR AUDIO SIGNAL USING THE SAME - An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme. | 04-30-2015 |
| 20150127354 | NEAR FIELD COMPENSATION FOR DECOMPOSED REPRESENTATIONS OF A SOUND FIELD - In general, techniques are described for compressing higher order ambisonics (HOA) audio data. A device comprising one or more processors may be configured to perform the techniques. The one or more processors may be configured to obtain a plurality of spherical harmonic coefficients from a plurality of near field compensated spherical harmonic coefficients by, at least in part, counterbalancing application of a near field compensation filter to the plurality of spherical harmonic coefficients. | 05-07-2015 |
| 20150127355 | METHOD FOR INSERTING WATERMARK TO IMAGE AND ELECTRONIC DEVICE THEREOF - A method for operating an electronic device is provided. The method includes determining one or more images; determining at least one first sound sources; dividing the first sound source into a plurality of second sound sources; and inserting at least one of the plurality of the second sound sources into the one or more images. | 05-07-2015 |
| 20150127356 | METHOD, TERMINAL, SYSTEM FOR AUDIO ENCODING/DECODING/CODEC - Audio encoding methods/terminals, audio decoding methods/terminals, and audio codec systems are provided. A plurality of audio signals that are continuous is obtained. It is determined whether each audio signal of the plurality of audio signals includes a designated signal type, according to an audio parameter of each audio signal. A marked audio encoding stream is obtained by performing a marking to each audio signal as having or not having the designated signal type. The marking is used, at a decoding terminal, to perform an enhancement-process to one or more audio signals having the designated signal type. The enhancement-process is not performed to audio signals that do not have the designated signal type. | 05-07-2015 |
| 20150134342 | Enhancement of Narrowband Audio Signals Using Single Sideband AM Modulation - The present document relates to the efficient processing of audio signals for enhancing the perceptual quality of the audio signal. An audio processing unit configured to generate an enhanced audio signal from an input audio signal is described. The input audio signal is sampled at a first sampling rate and the enhanced audio signal is sampled at a second sampling rate, wherein the second sampling rate is higher than the first sampling rate. The input audio signal comprises spectral content in a frequency range up to a first frequency and the enhanced audio signal comprises spectral content in a frequency range up to a second frequency, wherein the second frequency is higher than the first frequency. The audio processing unit comprises an upsampling and interpolation unit configured to generate an upsampled audio signal at the second sampling rate from the input audio signal. | 05-14-2015 |
| 20150142450 | Sound Processing using a Product-of-Filters Model - Sound processing using a product-of-filters model is described. In one or more implementations, a model is formed by one or more computing devices for a time frame of sound data as a product of filters. The model is utilized by the one or more computing devices to perform one or more sound processing techniques on the time frame of the sound data. | 05-21-2015 |
| 20150142451 | ERROR CONCEALMENT STRATEGY IN A DECODING SYSTEM - A decoding system reconstructs an audio signal based on an input signal representing the audio signal by parametric coding or by n discretely coded channels. Parametric decoding proceeds on the basis of a core signal and mixing parameters controlling a spatial synthesis stage, which is supplied with a downmix signal. A controller is responsible for controlling the components of the decoding system, whether in steady-state parametric mode, steady-state discrete decoding mode and transitions between these. In defective frames of the input signal, which do not allow the mixing parameters to be decoded, the controller is configured to perform various error handling procedures including: parametric decoding using previous values of the mixing parameters; continuing parametric decoding for a limited duration, and/or outputting the core signal without spatial synthesis. | 05-21-2015 |
| 20150142452 | METHOD AND APPARATUS FOR CONCEALING FRAME ERROR AND METHOD AND APPARATUS FOR AUDIO DECODING - Disclosed is a frame error concealment (FEC) method. The method includes: selecting an FEC mode based on states of a current frame and a previous frame of the current frame in a time domain signal generated after time-frequency inverse transform processing; and performing corresponding time domain error concealment processing on the current frame based on the selected FEC mode, wherein the current frame is an error frame or the current frame is a normal frame when the previous frame is an error frame. | 05-21-2015 |
| 20150142453 | ENCODING AND DECODING OF AUDIO SIGNALS - An encoder ( | 05-21-2015 |
| 20150142454 | HANDLING OVERLAPPING AUDIO RECORDINGS - Apparatus is configured to:
| 05-21-2015 |
| 20150142455 | SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD, AND COMPUTER PROGRAM - There is provided a signal processing device including a signal coincidence detection portion which detects samples, in which values based on a number of times of appearance of bits coincide with each other over a plurality of samples within a pre-set period, between a first modulated signal obtained by delaying an input signal obtained by ΣΔ modulation and a second modulated signal obtained by subjecting the input signal to the ΣΔ modulation again, a signal changeover portion which switches between the first modulated signal and the second modulated signal for outputting, and a switching control portion which controls the switching between the first modulated signal and the second modulated signal by the signal changeover portion in the samples in which the values based on the number of times of the appearance coincide with each other obtained by the signal coincidence detection portion. | 05-21-2015 |
| 20150149184 | APPARATUS FOR DISPLAYING IMAGE AND DRIVING METHOD THEREOF, APPARATUS FOR OUTPUTTING AUDIO AND DRIVING METHOD THEREOF - An image display apparatus, a method for driving an image display apparatus, a sound output apparatus and a method for driving a sound output apparatus, are provided. The image display apparatus comprising a signal separator configured to separate an audio signal and a video signal from an input image signal, an audio decoder configured to decode the audio signal, a sound outputter configured to output the decoded audio signal, a sound effect generator configured to generate a sound effect at a user's request, a communication interface configured to transmit the separated audio signal and the generated sound effect to a surrounding sound output apparatus, respectively, and a controller configured to control the communication interface to transmit the audio signal and the sound effect to the sound output apparatus, wherein the separated audio signal is transmitted when the sound output apparatus is connected. | 05-28-2015 |
| 20150149185 | AUDIO ENCODING DEVICE AND AUDIO CODING METHOD - An audio encoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: calculating a similarity in phase of a first channel signal and a second channel signal contained in a plurality of channels of an audio signal; and selecting, based on the similarity, a first output that outputs one of the first channel signal and the second channel signal, or a second output that outputs both of the first channel signal and the second channel signal. | 05-28-2015 |
| 20150149186 | DATA PROCESSING METHOD AND SYSTEM - The present disclosure relates to a data processing method and system. The method includes obtaining network data and a sound wave synthesized with the network data by a terminal, the sound wave being obtained by performing an encoding conversion on resource data; and according to an operation performed by a user on the network data on the terminal, invoking an audio playback apparatus of the terminal to play the sound wave synthesized with the network data to terminals of one or more users nearby. | 05-28-2015 |
| 20150149187 | DECODER AND METHOD FOR MULTI-INSTANCE SPATIAL-AUDIO-OBJECT-CODING EMPLOYING A PARAMETRIC CONCEPT FOR MULTICHANNEL DOWNMIX/UPMIX CASES - A decoder for generating an audio output signal having one or more audio output channels from a downmix signal having three or more downmix channels, wherein the downmix signal encodes three or more audio object signals is provided. The decoder includes an input channel router and at least two channel processing units. Each channel processing unit of the at least two channel processing units is configured to generate one or more of at least two processed channels depending on side information and depending on one or more of the three or more downmix channels received by the channel processing unit from the input channel router. | 05-28-2015 |
| 20150302860 | AUDIO SIGNAL CODING AND DECODING METHOD AND DEVICE - An audio signal encoding method is provided. The method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved. | 10-22-2015 |
| 20150310870 | SYSTEMS AND METHODS FOR ANALYZING AUDIO CHARACTERISTICS AND GENERATING A UNIFORM SOUNDTRACK FROM MULTIPLE SOURCES - Systems and methods analyze audio characteristics of digital audio files and generate a uniform soundtrack based on more than one of the digital audio files. The systems and method comprise equalizing a content of each input digital audio file such that all input digital audio files are processible as a group, wherein the input digital audio files are storable in a database and comprise a list of original recorded digital audio files from original recorded digital video files that were recorded from at least two different digital sources at the same event and the input digital audio files have previously been synchronized such that exact locations of the input digital audio files within the same event has been determined or identified. Moreover, the systems and methods comprise analyzing audio characteristics of the input digital audio files to detect a content quality of each input digital audio file, retrieving highest possible content qualities of the input digital audio files by cleaning the input digital audio files, and generating a unified soundtrack for one or more portions of the same event by merging more than one of the input digital audio files into an output digital audio file. | 10-29-2015 |
| 20150310872 | Multistage IIR Filter and Parallelized Filtering of Data with Same - In some embodiments, a multistage filter whose biquad filter stages are combined with latency between the stages, a system (e.g., an audio encoder or decoder) including such a filter, and methods for multistage biquad filtering. In typical embodiments, all biquad filter stages of the filter are operable independently to perform fully parallelized processing of data. In some embodiments, the inventive multistage filter includes a buffer memory, at least two biquad filter stages, and a controller coupled and configured to assert a single stream of instructions to the filter stages. Typically, the multistage filter is configured to perform multistage filtering of a block of input samples in a single processing loop with iteration over a sample index but without iteration over a biquadratic filter stage index. | 10-29-2015 |
| 20150317982 | Method and compressor for compressing audio dynamics - A method of compressing the audio dynamics of an audio signal. The method includes a step of acquiring an audio signal; a step of selecting a first instant of the audio signal; a step of calculating a plurality of partial gains corresponding, respectively, to a plurality of observation windows of the audio signal that are centered on the first instant, the width of the observation windows following a geometric progression with a rate and first term that are predefined; a step of summing the partial gains calculated in a first corrective term; and a step of applying the first corrective term to the audio signal at the first instant. | 11-05-2015 |
| 20150317983 | METHODS AND SYSTEMS FOR PROCESSING AND MIXING SIGNALS USING SIGNAL DECOMPOSITION - A method for mixing, processing and enhancing signals using signal decomposition is presented. A method for improving sorting of decomposed signal parts using cross-component similarity is also provided. | 11-05-2015 |
| 20150317985 | Signal Adaptive FIR/IIR Predictors for Minimizing Entropy - The present document relates to coding. In particular, the present document relates to coding using linear prediction in combination with entropy encoding. A method ( | 11-05-2015 |
| 20150317990 | DEEP SCATTERING SPECTRUM IN ACOUSTIC MODELING FOR SPEECH RECOGNITION - Deep scattering spectral features are extracted from an acoustic input signal to generate a deep scattering spectral feature representation of the acoustic input signal. The deep scattering spectral feature representation is input to a speech recognition engine. The acoustic input signal is decoded based on at least a portion of the deep scattering spectral feature representation input to a speech recognition engine. | 11-05-2015 |
| 20150317993 | METHOD AND SYSTEM TO PLAY BACKGROUND MUSIC ALONG WITH VOICE ON A CDMA NETWORK - A method and system for compressing an audio signal. The method includes receiving a segment of an audio signal and selectively disabling noise suppression for the received segment. The segment is filtered in a noise-suppression module if noise suppression is not disabled. The method also includes calculating an autocorrelation coefficient and an LSP coefficient, predicting a short-term coefficient and long-term coefficients according to the LSP coefficient and calculating one or more bandwidth-expanded correlation coefficients. Further, the method includes determining the type of packet in which to encode the segment. An encoding rate is selected from among a full rate encode, a half-rate encode, and an eight-rate encode if noise suppression is not disabled. An encoding rate is selected from among a full rate encode and a half-rate encode if noise suppression is disabled. Furthermore, the segment is formed into a packet of the determined type and selected rate. | 11-05-2015 |
| 20150325246 | REVERSIBLE AUDIO DATA HIDING - The present invention provides a method of reversible audio data hiding. The method of data hiding and restoring comprises the steps of: protecting audio by embedding information into the audio according to variance calculation associated to the audio, wherein the quality of the protected audio is degraded after embedding the information into the audio; publishing the protected audio widely as a trial for listen version; and decoding the protected audio for a user who purchased the copyright of the audio by extracting the original audio from the protected audio. | 11-12-2015 |
| 20150332676 | AUDIO ENCODERS, AUDIO DECODERS, SYSTEMS, METHODS AND COMPUTER PROGRAMS USING AN INCREASED TEMPORAL RESOLUTION IN TEMPORAL PROXIMITY OF ONSETS OR OFFSETS OF FRICATIVES OR AFFRICATES - An audio encoder for providing an encoded audio information on the basis of an input audio information has a bandwidth extension information provider configured to provide bandwidth extension information using a variable temporal resolution and a detector configured to detect an onset of a fricative or affricate. The audio encoder is configured to adjust a temporal resolution used by the bandwidth extension information provider such that bandwidth extension information is provided with an increased temporal resolution at least for a predetermined period of time before a time at which an onset of a fricative or affricate is detected and for a predetermined period of time following the time at which the onset of the fricative or affricate is detected. Alternatively or in addition, the bandwidth extension information is provided with an increased temporal resolution in response to a detection of an offset of a fricative or affricate. Audio encoders and methods use a corresponding concept. | 11-19-2015 |
| 20150332678 | UNOBTRUSIVE AUDIO MESSAGES - A method for providing audio messages includes receiving a first image set and a second image set. The first image set includes visually encoded audio data for rendering audio on an electronic computing device. The method also includes displaying images from the first and second image sets interspersed in an image sequence. In the image sequence, a time interval between each image from the first image set and at least one image from the second image set is less than a critical flicker interval (CFI) for a human eye. | 11-19-2015 |
| 20150332683 | CROSSFADING BETWEEN HIGHER ORDER AMBISONIC SIGNALS - In general, techniques are described for crossfading sets of spherical harmonic coefficients. An audio encoding device or audio decoding device comprising a memory and a processor may be configured to perform the techniques. The memory may be configured to store a first set of spherical harmonic coefficients (SHCs) and a second set of SHCs. The first set of SHCs describe a first sound field. The second set of SHCs describe a second sound field. The processor may be configured to crossfade between the first set of SHCs and a second set of SHCs to obtain a first set of crossfaded SHCs. | 11-19-2015 |
| 20150332684 | Apparatus For Processing An Audio Signal And Method Thereof - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting control restriction information from the side information; receiving control information for controlling gain or panning at least one object signal; generating at least one of first multi-channel information and first downmix processing information based on the control information and object information, without using the control restriction information; and, generating an output signal by applying the at least one of the first multichannel information and the first downmix processing information to the downmix signal, wherein the control restriction information relates to a parameter indicating limiting degree of the control information. | 11-19-2015 |
| 20150332685 | METHOD AND APPARATUS FOR NORMALIZED AUDIO PLAYBACK OF MEDIA WITH AND WITHOUT EMBEDDED LOUDNESS METADATA ON NEW MEDIA DEVICES - A decoder device for decoding a bitstream so as to produce therefrom an audio output signal, the bitstream having audio data and optionally loudness metadata containing a reference loudness value, wherein a gain control device has a reference loudness decoder configured to create a loudness value, wherein the loudness value is the reference loudness value in case that the reference loudness value is present in the bitstream; wherein the gain control device has a gain calculator configured to calculate a gain value based on the loudness value and based on a volume control value, which is provided by an external user interface allowing a user to control the volume control value, and a loudness processor configured to control the loudness of the audio output signal based on the gain value. | 11-19-2015 |
| 20150332688 | Method for Predicting Bandwidth Extension Frequency Band Signal, and Decoding Device - A method for predicting a bandwidth extension frequency band signal includes demultiplexing a received bitstream to obtain a frequency domain signal; determining whether a highest frequency bin, to which a bit is allocated, of the frequency domain signal is less than a preset start frequency bin of a bandwidth extension frequency band; predicting an excitation signal of the bandwidth extension frequency band according to the determination; and predicting the bandwidth extension frequency band signal according to the predicted excitation signal of the bandwidth extension frequency band and a frequency envelope of the bandwidth extension frequency band. | 11-19-2015 |
| 20150332689 | NOISE FILLING CONCEPT - Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal. | 11-19-2015 |
| 20150332693 | CONCEPT FOR CODING MODE SWITCHING COMPENSATION - A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition. | 11-19-2015 |
| 20150332696 | NOISE FILLING WITHOUT SIDE INFORMATION FOR CELP-LIKE CODERS - An audio decoder provides a decoded audio information on the basis of an encoded audio information including linear prediction coefficients (LPC) and includes a tilt adjuster to adjust a tilt of a noise using linear prediction coefficients of a current frame to acquire a tilt information and a noise inserter configured to add the noise to the current frame in dependence on the tilt information. Another audio decoder includes a noise level estimator to estimate a noise level for a current frame using a linear prediction coefficient of at least one previous frame to acquire a noise level information; and a noise inserter to add a noise to the current frame in dependence on the noise level information provided by the noise level estimator. Thus, side information about a background noise in the bit-stream may be omitted. Methods and computer programs serve a similar purpose. | 11-19-2015 |
| 20150332698 | APPARATUS AND METHOD FOR SELECTING ONE OF A FIRST ENCODING ALGORITHM AND A SECOND ENCODING ALGORITHM - An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures. | 11-19-2015 |
| 20150332701 | DECODER FOR GENERATING A FREQUENCY ENHANCED AUDIO SIGNAL, METHOD OF DECODING, ENCODER FOR GENERATING AN ENCODED SIGNAL AND METHOD OF ENCODING USING COMPACT SELECTION SIDE INFORMATION - A decoder for generating a frequency enhanced audio signal, includes: a feature extractor for extracting a feature from a core signal; a side information extractor for extracting a selection side information associated with the core signal; a parameter generator for generating a parametric representation for estimating a spectral range of the frequency enhanced audio signal not defined by the core signal, wherein the parameter generator is configured to provide a number of parametric representation alternatives in response to the feature, and wherein the parameter generator is configured to select one of the parametric representation alternatives as the parametric representation in response to the selection side information; and a signal estimator for estimating the frequency enhanced audio signal using the parametric representation selected. | 11-19-2015 |
| 20150332702 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR PROVIDING AN ENCODED AUDIO INFORMATION, METHOD FOR PROVIDING A DECODED AUDIO INFORMATION, COMPUTER PROGRAM AND ENCODED REPRESENTATION USING A SIGNAL-ADAPTIVE BANDWIDTH EXTENSION - An audio encoder has a low frequency encoder which encodes a low frequency portion of the input audio information to obtain an encoded representation of the low frequency portion, and a bandwidth extension information provider which provides bandwidth extension information. The audio encoder is configured to selectively include bandwidth extension information into the encoded audio information in a signal-adaptive manner. An audio decoder has a low frequency decoder which decodes an encoded representation of a low frequency portion to obtain a decoded representation of the low frequency portion, and a bandwidth extension which obtains a bandwidth extension signal using a blind bandwidth extension for portions of an audio content for which no bandwidth extension parameters are included in the encoded audio information, and which obtains the bandwidth extension signal using a parameter-guided bandwidth extension for portions of the audio content for which bandwidth extension parameters are included in the encoded audio information. | 11-19-2015 |
| 20150340043 | MULTICHANNEL ENCODER AND DECODER WITH EFFICIENT TRANSMISSION OF POSITION INFORMATION - A receiver ( | 11-26-2015 |
| 20150340046 | Systems and Methods for Audio Encoding and Decoding - Systems and methods are provided for audio encoding. For example, first quantization encoding is performed on audio data associated with a current frame of an audio data stream to obtain first quantization-encoded data; second quantization encoding is performed on the audio data to obtain second quantization-encoded data; the first quantization-encoded data is coupled to the current frame of the audio data stream; and the second quantization-encoded data is coupled to a next frame of the audio data stream. | 11-26-2015 |
| 20150348558 | Audio Bitstreams with Supplementary Data and Encoding and Decoding of Such Bitstreams - Methods for generating or decoding an encoded audio bitstream including audio data and supplementary data (e.g., metadata and/or unrelated audio data), where at least some of the supplementary data is included as LSBs of audio segments, and/or at least some of the supplementary data is included in guard bands. Typical embodiments provide a scalable and video synchronous format compatible with real-time and file-based infrastructure components that support the SMPTE 337 format for carrying data in AES3 serial bitstreams, and/or provide a framework for extending distribution codecs to scale beyond an 8-channel limit to support multiples of 8 channels synchronously across multiple AES3 interfaces. Another aspect is an audio processing unit configured to perform any embodiment of the method or including a buffer memory storing at least one segment of an audio bitstream generated in accordance with any embodiment of the method. | 12-03-2015 |
| 20150348559 | APPARATUS AND METHOD FOR SPATIAL AUDIO OBJECT CODING EMPLOYING HIDDEN OBJECTS FOR SIGNAL MIXTURE MANIPULATION - An apparatus for encoding one or more audio objects to obtain an encoded signal is provided. The apparatus includes a for downmixing the one or more audio objects to obtain one or more unprocessed downmix signals. Moreover, the apparatus includes a processing module and a signal calculator. The signal calculator is configured to calculate each of one or more additional signals based on a difference between one of one or more processed downmix signals and one of the one or more unprocessed downmix signals. Moreover, the apparatus includes an object information generator. Furthermore, the apparatus includes an output interface for outputting the encoded signal. Moreover, a corresponding apparatus for decoding is provided. | 12-03-2015 |
| 20150348560 | APPARATUS FOR ENCODING/DECODING MULTICHANNEL SIGNAL AND METHOD THEREOF - Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme. | 12-03-2015 |
| 20150348564 | DECODER, ENCODER AND METHOD FOR INFORMED LOUDNESS ESTIMATION EMPLOYING BY-PASS AUDIO OBJECT SIGNALS IN OBJECT-BASED AUDIO CODING SYSTEMS - A decoder for generating an audio output signal having one or more audio output channels is provided, having a receiving interface for receiving an audio input signal having a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated, further having a signal processor for generating the one or more audio output channels of the audio output signal, configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information, and configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. One or more by-pass audio object signals are employed for generating the audio output signal. Moreover, an encoder is provided. | 12-03-2015 |
| 20150356976 | AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHOD FOR PROVIDING AN UPMIX SIGNAL REPRESENTATION, METHOD FOR PROVIDING A DOWNMIX SIGNAL REPRESENTATION, COMPUTER PROGRAM AND BITSTREAM USING A COMMON INTER-OBJECT-CORRELATION PARAMETER VALUE - An audio signal decoder for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information and in dependence on a rendering information has an object parameter determinator. The object parameter determinator is configured to obtain inter-object-correlation values for a plurality of pairs of audio objects. The object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether to evaluate individual inter-object-correlation bitstream parameter values to obtain inter-object-correlation values for a plurality of pairs of related audio objects, or to obtain inter-object-correlation values for a plurality of pairs of related audio objects using a common inter-object-correlation bitstream parameter value. The audio signal decoder also has a signal processor configured to obtain the upmix signal representation on the basis of the downmix signal representation and using the inter-object-correlation values for a plurality of pairs of related objects and the rendering information. | 12-10-2015 |
| 20150356977 | AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHOD FOR PROVIDING AN UPMIX SIGNAL REPRESENTATION, METHOD FOR PROVIDING A DOWNMIX SIGNAL REPRESENTATION, COMPUTER PROGRAM AND BITSTREAM USING A COMMON INTER-OBJECT-CORRELATION PARAMETER VALUE - An audio signal decoder for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information and in dependence on a rendering information has an object parameter determinator. The object parameter determinator is configured to obtain inter-object-correlation values for a plurality of pairs of audio objects. The object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether to evaluate individual inter-object-correlation bitstream parameter values to obtain inter-object-correlation values for a plurality of pairs of related audio objects, or to obtain inter-object-correlation values for a plurality of pairs of related audio objects using a common inter-object-correlation bitstream parameter value. The audio signal decoder also has a signal processor configured to obtain the upmix signal representation on the basis of the downmix signal representation and using the inter-object-correlation values for a plurality of pairs of related objects and the rendering information. | 12-10-2015 |
| 20150356979 | AUDIO DECODING USING MODULATOR-DEMODULATOR - An electronic device is provided. The electronic device includes: a first processing unit; a storage unit, configured to store at least one audio file; a first memory unit; and a modulator-demodulator (modem), configured to perform audio processing of the electronic device during a phone call, wherein when the electronic device is used to play the audio file, the first processing unit reads the audio file from the storage unit, retrieves header information of the audio file, and writes the audio file into the first memory unit, wherein the modem accesses the audio file stored in the first memory unit based on the header information, and performs audio decoding on the audio file. | 12-10-2015 |
| 20150364144 | COMFORT NOISE ADDITION FOR MODELING BACKGROUND NOISE AT LOW BIT-RATES - The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal. | 12-17-2015 |
| 20150371640 | AUDIO CODING DEVICE, AUDIO CODING METHOD, AND AUDIO CODEC DEVICE - An audio coding device includes a memory; and a processor configured to execute a plurality of instructions stored in the memory, the instructions comprising: selecting a main lobe among a plurality of lobes detected from a frequency signal configuring an audio signal on a basis of bandwidth and power of the lobes; and coding the audio signal in such a manner that a first amount of bits per a unit frequency domain allocated to coding of the frequency signal of the main lobe is larger than a second amount of bits per the unit frequency domain allocated to the coding of the frequency signal of a side lobe as a lobe other than the main lobe. | 12-24-2015 |
| 20150371649 | Processing Audio Signals with Adaptive Time or Frequency Resolution - In one aspect, a method for processing an encoded audio signal is disclosed. The method includes decoding the encoded audio signal to obtain a time-domain audio signal and then analyzing the time-domain audio signal with an analysis filter bank to obtain a plurality of complex-valued subband samples in a first frequency region. The method further includes processing the audio signal by generating a plurality of subband samples in a second frequency region based at least in part on the complex-valued subband samples in the first frequency region, grouping at least some of the plurality of subband samples in the second frequency region with an adaptive time resolution and an adaptive frequency resolution to obtain an adaptive grouping, and determining a spectral profile of at least some of the subband samples in the second frequency region based at least in part on the adaptive grouping. | 12-24-2015 |
| 20150371650 | Communicating Information Between Devices Using Ultra High Frequency Audio - A client device encodes data into an audio signal and communicates the audio data to an additional client device, which decodes the data from the audio signal. The data is partitioned into characters, which are subsequently partitioned into a plurality of sub-characters. Each sub-character is encoded into a frequency, and multiple frequencies that encode sub-characters are combined by the client device to generate an audio signal. Frequencies encoding sub-characters may be above 16 kilohertz, so the sub-characters are transmitted using frequencies that are inaudible to humans. The audio signal is communicated to an additional client device, which decodes frequencies from the audio signal to sub-characters, which are then combined into characters by the additional client device to generate the data. | 12-24-2015 |
| 20150380005 | SYSTEMS AND METHODS FOR COMPRESSING A DIGITAL SIGNAL - A system may include a delta-sigma analog-to-digital converter and a digital compression circuit. The delta-sigma analog-to-digital converter may include a loop filter having a loop filter input configured to receive an input signal and generate an intermediate signal responsive to the input signal, a multi-bit quantizer configured to quantize the intermediate signal into an uncompressed digital output signal, and a feedback digital-to-analog converter having a feedback output configured to generate a feedback output signal responsive to the uncompressed digital output signal in order to combine the input signal and the feedback output signal at the loop filter input. The digital compression circuit may be configured to receive the uncompressed digital output signal and compress the uncompressed digital output signal into a compressed digital output signal having fewer quantization levels than that of the uncompressed digital output signal. | 12-31-2015 |
| 20150380008 | HIGH-BAND SIGNAL CODING USING MISMATCHED FREQUENCY RANGES - A method includes generating a first signal corresponding to a first component of a high-band portion of an audio signal. The first component has a first frequency range. The method includes generating a high-band excitation signal corresponding to a second component of the high-band portion of the audio signal. The second component has a second frequency range differs from the first frequency range. The high-band excitation signal is provided to a filter having filter coefficients generated based on the first signal to generate a synthesized version of the high-band portion of the audio signal. | 12-31-2015 |
| 20160005408 | THREE-DIMENSIONAL SOUND COMPRESSION AND OVER-THE-AIR-TRANSMISSION DURING A CALL - A method for encoding three dimensional audio by a wireless communication device is disclosed. The wireless communication device detects an indication of a plurality of localizable audio sources. The wireless communication device also records a plurality of audio signals associated with the plurality of localizable audio sources. The wireless communication device also encodes the plurality of audio signals. | 01-07-2016 |
| 20160005410 | SYSTEM, APPARATUS, AND METHOD FOR AUDIO FINGERPRINTING AND DATABASE SEARCHING FOR AUDIO IDENTIFICATION - Client device for audio fingerprinting and database searching for audio identification comprises processor; audio fingerprint (“FP”) generator, query FP storage, FP database storage that stores audio FP database, signature generator, searching module, and display device. Audio FP generator receives audio signals recorded by client device, and generate audio FP of the recorded audio signals that is a query FP stored in query FP storage. Signature generator generates a database of signatures from the FP database, and generates a signature of the query FP. Searching module searches the signature of the query FP in the database of signatures, searches the query audio FP in the FP database when a potential match is obtained for the signature of the query FP, and generates a result of the search of the query audio FP. Display device displays the result of the search which may be an advertisement corresponding to query FP. Other embodiments are described. | 01-07-2016 |
| 20160005415 | AUDIO SIGNAL PROCESSING APPARATUS AND AUDIO SIGNAL PROCESSING METHOD THEREOF - An audio signal processing apparatus and an audio signal processing method thereof are provided. The audio signal processing apparatus is configured to receive an audio signal and divide the audio signal into a plurality of frames. The audio signal processing apparatus is also configured to apply Fourier Transform on each of the frames to obtain a plurality of acoustic spectra. The audio signal processing apparatus is also configured to apply Fourier Transform again on each of component combinations corresponding to respective acoustic frequencies in these acoustic spectra to obtain a two-dimensional joint frequency spectrum. The two-dimensional joint frequency spectrum has an acoustic frequency dimension and a modulation frequency dimension. The audio signal processing apparatus is also configured to calculate at least one feature of the audio signal according to the two-dimensional joint frequency spectrum. | 01-07-2016 |
| 20160012826 | METHOD AND SYSTEM FOR DIGITAL WATERMARKING | 01-14-2016 |
| 20160019898 | TIME DOMAIN LEVEL ADJUSTMENT FOR AUDIO SIGNAL DECODING OR ENCODING - An audio signal decoder for providing a decoded audio signal representation on the basis of an encoded audio signal representation has a decoder preprocessing stage for obtaining a plurality of frequency band signals from the encoded audio signal representation, a clipping estimator, a level shifter, a frequency-to-time-domain converter, and a level shift compensator. The clipping estimator analyzes the encoded audio signal representation and/or side information relative to a gain of the frequency band signals in order to determine a current level shift factor. The level shifter shifts levels of the frequency band signals according to the level shift factor. The frequency-to-time-domain converter converts the level shifted frequency band signals into a time-domain representation. The level shift compensator acts on the time-domain representation for at least partly compensating a corresponding level shift and for obtaining a substantially compensated time-domain representation. | 01-21-2016 |
| 20160019901 | AUDIO WATERMARKING FOR PEOPLE MONITORING - Methods, apparatus, systems and articles of manufacture (e.g., physical storage media) to utilize audio watermarking for people monitoring are disclosed. Example people monitoring methods disclosed herein include determining, at a user device, whether a first trigger condition for emitting an audio watermark identifying at least one of the user device or a user of the user device is satisfied. Such example methods also include, in response to determining that the first trigger condition is satisfied, providing a first audio signal including the audio watermark to an audio circuit that is to output an acoustic signal from the user device. | 01-21-2016 |
| 20160019902 | OPTIMIZED PARTIAL MIXING OF AUDIO STREAMS ENCODED BY SUB-BAND ENCODING - The invention relates to a method for combining a plurality of audio streams encoded by frequency sub-band encoding, comprising the following steps: decoding (E | 01-21-2016 |
| 20160019903 | OPTIMIZED MIXING OF AUDIO STREAMS ENCODED BY SUB-BAND ENCODING - The invention relates to a method for mixing a plurality of audio streams coded according to a frequency sub-band coding, comprising the steps for decoding (E | 01-21-2016 |
| 20160019908 | COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION - Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation. | 01-21-2016 |
| 20160019911 | FREQUENCY BAND EXTENDING DEVICE AND METHOD, ENCODING DEVICE AND METHOD, DECODING DEVICE AND METHOD, AND PROGRAM - The present invention relates to a frequency band extending device and method, an encoding device and method, a decoding device and method, and a program, whereby music signals can be played with higher sound quality due to the extension of frequency bands. | 01-21-2016 |
| 20160027448 | LOW-COMPLEXITY TONALITY-ADAPTIVE AUDIO SIGNAL QUANTIZATION - The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value. | 01-28-2016 |
| 20160035355 | AUDIO DECODER AND DECODING METHOD USING EFFICIENT DOWNMIXING - A method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M| 02-04-2016 | |
| 20160035356 | EDITING OF HIGHER-ORDER AMBISONIC AUDIO DATA - In general, techniques are described for editing of higher-order ambisonic audio data. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may be configured to store spherical harmonic (SH) basis functions. The one or more processors may be configured to manipulate the SH basis functions associated with higher order ambisonics coefficients to alter a direction of an audio object represented by the higher order ambisonics coefficients. | 02-04-2016 |
| 20160035365 | SOUND ENCODING DEVICE, SOUND ENCODING METHOD, SOUND DECODING DEVICE AND SOUND DECODING METHOD - A sound encoding device includes: a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: converting a sound signal into a frequency signal by time-frequency converting the sound signal in a unit of a frame having a given time length; detecting a first frequency band in which a phase component of the frequency signal is random for each frame; determining outline information representative of an outline of an amplitude component of the frequency signal included in the first frequency band for each frame; encoding the frequency signal included in a frequency band other than the first frequency band for each frame; and producing a data stream including the encoded frequency signal and the outline information. | 02-04-2016 |
| 20160049155 | ARTICLE OF MANUFACTURE, SYSTEM AND COMPUTER-READABLE STORAGE MEDIUM FOR PROCESSING AUDIO SIGNALS - Embodiments of an article of manufacture, a system for processing audio signals and a computer-readable storage medium containing program instructions for processing audio signals are described. In one embodiment, an article of manufacture comprising at least one non-transitory, tangible machine readable storage medium containing executable machine instructions for processing audio signals, where execution of the executable machine instructions by a processing device causes the processing device to perform steps, which include estimating a spectral difference between a first audio signal and a second audio signal that carry the same audio content, transforming the second audio signal based on the spectral difference and generating an output audio signal based on the transformed second audio signal. Other embodiments are also described. | 02-18-2016 |
| 20160049157 | METHOD FOR CODING PULSE VECTORS USING STATISTICAL PROPERTIES - Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information. | 02-18-2016 |
| 20160049158 | METHOD FOR CODING PULSE VECTORS USING STATISTICAL PROPERTIES - Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information. | 02-18-2016 |
| 20160049159 | METHOD FOR CODING PULSE VECTORS USING STATISTICAL PROPERTIES - Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information. | 02-18-2016 |
| 20160049160 | CONTEXT-BASED ARITHMETIC ENCODING APPARATUS AND METHOD AND CONTEXT-BASED ARITHMETIC DECODING APPARATUS AND METHOD - A context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method are provided. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code. | 02-18-2016 |
| 20160055853 | Method for processing sound data and circuit therefor - A sound data processing apparatus includes a central processing unit for controlling predetermined processing in the apparatus, a rewritable RAM, a decoder performing the decoding processing for sound data, and an interface unit for being fitted with an external memory. The sound data processing apparatus reads a driver from the external memory mounted in the interface unit and stores the read driver into the RAM, and reads the sound data from the external memory with the driver and processes the read sound data. As a result, the wastefully using of the memory capacity of the memory mounted in the sound data processing apparatus is reduced. | 02-25-2016 |
| 20160055855 | AUDIO PROCESSING SYSTEM - An audio processing system ( | 02-25-2016 |
| 20160055857 | SYSTEM AND METHOD FOR GENERATING DYNAMIC SOUND ENVIRONMENTS - A method comprising receiving a message request, obtaining a set of geographic coordinates from the received message request, conducting a first search of one or more meta-element databases using the set of geographic coordinates to obtain a plurality of metatags associated with the set of geographic coordinates, conducting a second search of one or more audio content databases using the plurality of metatags to obtain a plurality of audio sound files, generating a sound-stream using the plurality of audio sound files, the audio sound files comprising stored representations of simulated audio content and synthetic audio content associated with the set of geographic coordinates, encoding the sound-stream for rendering on a client device using one or more device-specific parameters in the message request, and transmitting the encoded sound-stream to the client device. | 02-25-2016 |
| 20160064004 | MULTIPLE CHANNEL AUDIO SIGNAL ENCODER MODE DETERMINER - It is inter alia disclosed a method comprising: determining an indication of similarity between a first audio frame of a multiple channel input audio signal and a second audio frame of the multiple channel input audio signal; and determining a coding mode for a multiple channel audio spatial encoder dependent on each of: data indicating a coding mode of a mono audio encoder for the first audio frame of the multiple channel input audio signal; a coding mode of the multichannel spatial audio encoder for the first audio frame of the multiple channel input audio signal; and the indication of similarity. | 03-03-2016 |
| 20160064006 | AUDIO OBJECT SEPARATION FROM MIXTURE SIGNAL USING OBJECT-SPECIFIC TIME/FREQUENCY RESOLUTIONS - An audio decoder is proposed for decoding a multi-object audio signal including a downmix signal X and side information PSI. The side information includes object-specific side information PSI | 03-03-2016 |
| 20160064013 | APPARATUS AND METHOD FOR ENCODING AND DECODING SIGNAL FOR HIGH FREQUENCY BANDWIDTH EXTENSION - An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal. | 03-03-2016 |
| 20160086611 | MUSIC REPRODUCING APPARATUS - In a smartphone that outputs PCM data and DSD data of different data format from PCM data to a USB DAC, to prevent generation of noises from the USB DAC when the smartphone accepts stopping instruction of outputting digital audio data to the USB DAC while outputting DSD data to the USB DAC. | 03-24-2016 |
| 20160086613 | Signal Decoding Method and Device - Embodiments of the present invention provide a signal decoding method and device. The method includes decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal; predicting an excitation signal of an extension band according to the decoded signal, where the extension band is adjacent to a band of the decoded signal, and the band of the decoded signal is lower than the extension band; selecting a first band and a second band from the decoded signal, and predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band; and determining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band. | 03-24-2016 |
| 20160086616 | PITCH FILTER FOR AUDIO SIGNALS - In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information. | 03-24-2016 |
| 20160093310 | Spectral Translation/Folding in the Subband Domain - The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications. | 03-31-2016 |
| 20160093312 | AUDIO ENCODER AND DECODER WITH MULTIPLE CODING MODES - In one embodiment, an audio decoder for decoding an audio bitstream is disclosed. The decoder includes a first decoding module adapted to operate in a first coding mode and a second decoding module adapted to operate in a second coding mode, the second coding mode being different from the first coding mode. The decoder further includes a pitch filter in either the first coding mode or the second coding mode, the pitch filter adapted to filter a preliminary audio signal generated by the first decoding module or the second decoding module to obtain a filtered signal. The pitch filter is selectively enabled or disabled based on a value of a first parameter encoded in the audio bitstream, the first parameter being distinct from a second parameter encoded in the audio bitstream, the second parameter specifying a current coding mode of the audio decoder. | 03-31-2016 |
| 20160099000 | POST-ENCODING BITRATE REDUCTION OF MULTIPLE OBJECT AUDIO - A post-encoding bitrate reduction system and method for generating one more scaled compressed bitstreams from a single encoded plenary file. The plenary file contains multiple audio object files that were encoded separately using a scalable encoding process having fine-grained scalability. Activity in the data frames of the encoded audio object files at a time period are compared with each other to obtain a data frame activity comparison. Bits from an available bitpool are assigned to all of the data frames based on the data frame activity comparison and corresponding hierarchical metadata. The plenary file is scaled down by truncating bits in the data frames to conform to the bit allocation. In some embodiments frame activity is compared to a silence threshold and the data frame contains silence if the frame activity is less than or equal to the threshold and minimal bits are used to represent the silent frame. | 04-07-2016 |
| 20160104498 | APPARATUS AND METHOD FOR TRANSMITTING WATERMARK ROBUST TO ACOUSTIC CHANNEL DISTORTION - A method and an apparatus for transmitting a watermark robust to an acoustic channel distortion are disclosed. The method of transmitting the watermark may include extracting a watermark from a first audio signal including the watermark; modifying the extracted watermark based on a state of an acoustic channel; and embedding the modified watermark into the first audio signal to output a second audio signal. | 04-14-2016 |
| 20160118051 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD - An audio encoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: detecting a plurality of lobes based on a frequency signal constituting an audio signal; calculating a masking threshold value of the frequency signal; allocating an amount of bits per unit frequency region to be allocated for encoding of the frequency signal on a basis of the masking threshold value; selecting a main lobe on a basis of bandwidth and power of the lobes; and controlling the encoding by reducing the amount of bits in a first region including a maximum value of the power in the main lobe. | 04-28-2016 |
| 20160125886 | AUDIO DATA TRANSMITTING METHOD AND DATA TRANSMITTING SYSTEM - An audio data transmitting method applied to an audio data transmitting device. The audio data transmitting method comprises: (a) receiving first audio data from at least one audio data source, wherein the first audio data follows a first audio format; and (b) outputting the first audio data from the audio data transmitting device without encoding or decoding the first audio data. | 05-05-2016 |
| 20160133265 | APPARATUS AND METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL WITH INTELLIGENT GAP FILLING IN THE SPECTRAL DOMAIN - An apparatus for decoding an encoded audio signal, includes a spectral domain audio decoder for generating a first decoded representation of a first set of first spectral portions, the decoded representation having a first spectral resolution; a parametric decoder for generating a second decoded representation of a second set of second spectral portions having a second spectral resolution being lower than the first spectral resolution; a frequency regenerator for regenerating every constructed second spectral portion having the first spectral resolution using a first spectral portion and spectral envelope information for the second spectral portion; and a spectrum time converter for converting the first decoded representation and the reconstructed second spectral portion into a time representation. | 05-12-2016 |
| 20160140969 | DETERMINING MEDIA DEVICE ACTIVATION BASED ON FREQUENCY RESPONSE ANALYSIS - Methods, apparatus, systems and articles of manufacture (e.g., physical storage media) to determine media device activation based on frequency response analysis are disclosed. Example methods disclosed herein include determining a reference frequency response based on first frequency values of an audio signal used to perform watermark detection for a first time interval during which a media device has been determined to be active. Such example methods also include determining a second frequency response based on second frequency values of the audio signal used to perform watermark detection for a second time interval different from the first time interval. Such example methods further include comparing the second frequency response with the reference frequency response to determine whether the media device was active during the second time interval. | 05-19-2016 |
| 20160140982 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD, ENCODER AND ENCODING METHOD, DECODER AND DECODING METHOD, AND PROGRAM - Methods and apparatus for performing signal processing. The signal processing comprises demultiplexing input encoded data into data including information for a segment including frames and coefficient information for a coefficient selected in the frames of the segment, and low band encoded data, decoding the low band encoded data to produce a low band signal, selecting a coefficient of a frame to be processed from a plurality of the coefficients based on the data, calculating a high band sub-band power of a high band sub-band signal of each sub-band constituting a high band signal of the frame to be processed based on a low band sub-band signal of each sub-band constituting the low band signal of the frame to be processed and the selected coefficient, and producing the high band signal of the frame to be processed based on the high band sub-band power and the low band sub-band signal. | 05-19-2016 |
| 20160148621 | APPARATUS AND METHOD FOR AUDIO SIGNAL ENVELOPE ENCODING, PROCESSING, AND DECODING BY SPLITTING THE AUDIO SIGNAL ENVELOPE EMPLOYING DISTRIBUTION QUANTIZATION AND CODING - An apparatus for decoding to obtain a reconstructed audio signal envelope includes a signal envelope reconstructor for generating the reconstructed audio signal envelope depending on one or more splitting points and an output interface for outputting the reconstructed audio signal envelope. The signal envelope reconstructor is configured to generate the reconstructed audio signal envelope such that the one or more splitting points divide the reconstructed audio signal envelope into two or more audio signal envelope portions, and to generate the reconstructed audio signal envelope such that, for each of the two or more signal envelope portions, an absolute value of its signal envelope portion value is greater than half of an absolute value of the signal envelope portion value of each of the other signal envelope portions. | 05-26-2016 |
| 20160155449 | METHOD AND SYSTEM FOR LOSSLESS VALUE-LOCATION ENCODING | 06-02-2016 |
| 20160155450 | Audio Encoding/Decoding based on an Efficient Representation of Auto-Regressive Coefficients | 06-02-2016 |
| 20160163321 | Processing of Time-Varying Metadata for Lossless Resampling - Embodiments are directed to a method of representing spatial rendering metadata for processing in an object-based audio system that allows for lossless interpolation and/or re-sampling of the metadata. The method comprises time stamping the metadata to create metadata instances, and encoding an interpolation duration to with each metadata instance that specifies the time to reach a desired rendering state for the respective metadata instance. The re-sampling of metadata is useful for re-clocking metadata to an audio coder and for the editing audio content. | 06-09-2016 |
| 20160163323 | APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING A COMBINATION IN AN OVERLAP RANGE - An apparatus for processing an audio signal including a sequence of blocks of spectral values includes: a processor for processing the sequence of blocks using at least one modification values for a first block to obtain aliasing-reduced or aliasing-free first result signal in an overlap range and using at least one second different modification value for a second block of the sequence of blocks to obtain an aliasing-reduced or aliasing-free second result signal in the overlap range; and a combiner for combining the first result signal and the second result signal in the overlap range to obtain a processed signal for the overlap range. | 06-09-2016 |
| 20160163324 | APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING AN ALIASING ERROR SIGNAL - An apparatus for processing an audio signal including a sequence of blocks of spectral values, includes: a processor for calculating an aliasing-affected signal using at least one first modification value for a first block of the sequence of blocks and using at least one different second modification value for a second block of the sequence of blocks and for estimating an aliasing-error signal representing an aliasing-error in the aliasing-affected signal; and a combiner for combining the aliasing-affected signal and the aliasing-error signal such that a processed signal obtained by the combining is an aliasing-reduced or aliasing-free signal. | 06-09-2016 |
| 20160171987 | SYSTEM AND METHOD FOR COMPRESSED AUDIO ENHANCEMENT | 06-16-2016 |
| 20160180860 | High order B-spline sampling rate conversion (SRC) | 06-23-2016 |
| 20160189719 | FRAME ERROR CONCEALMENT METHOD AND APPARATUS, AND AUDIO DECODING METHOD AND APPARATUS - Disclosed are a frame error concealment method and apparatus and an audio decoding method and apparatus. The frame error concealment (FEC) method includes: selecting an FEC mode based on at least one of a state of at least one frame and a phase matching flag, with regard to a time domain signal generated after time-frequency inverse transform processing; and performing corresponding time domain error concealment processing on the current frame based on the selected FEC mode, wherein the current frame is an error frame or the current frame is a normal frame when the previous frame is an error frame. | 06-30-2016 |
| 20160196828 | Acoustic Matching and Splicing of Sound Tracks | 07-07-2016 |
| 20160196829 | BANDWIDTH EXTENSION METHOD AND APPARATUS | 07-07-2016 |
| 20160196830 | AUDIO ENCODER AND DECODER WITH PROGRAM INFORMATION OR SUBSTREAM STRUCTURE METADATA | 07-07-2016 |
| 20160203825 | ENCODING APPARATUS AND ENCODING METHOD | 07-14-2016 |
| 20160203826 | OPTIMIZED SCALE FACTOR FOR FREQUENCY BAND EXTENSION IN AN AUDIO FREQUENCY SIGNAL DECODER | 07-14-2016 |
| 20160254004 | AUDIO CODING METHOD AND APPARATUS | 09-01-2016 |
| 20160254006 | Multistage IIR Filter and Parallelized Filtering of Data with Same | 09-01-2016 |
| 20160379645 | 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 |
| 20160379646 | 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 |
| 20160379647 | 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 |
| 20160379648 | 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 |
| 20160379649 | 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 |
| 20160379650 | 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 |
| 20160379652 | 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 |
| 20160379654 | ENCODING APPARATUS AND ENCODING METHOD - A coding apparatus, including a memory and a processor that, when executing instructions stored in the memory, performs operations including encoding low-band transform coefficients in a first band and calculating, for each extension-band subband obtained by splitting an extension band, a threshold amplitude based on an analysis of statistics on extension-band transform coefficients included in the subband. The processor further compares, for each of the extension-band subbands, an amplitude of the extension-band transform coefficients with the threshold amplitude to extract a representative transform coefficient, updates, when a number of the extracted representative transform coefficients is less than a predetermined number, the threshold amplitude, performs processing to again extract a transform coefficient using the updated threshold amplitude, calculates, for each of the extension-band subbands, a value of correlation between the representative transform coefficient and a normalized encoded low-band transform coefficient. | 12-29-2016 |
| 20170236525 | Digital Audio Processing Apparatus, Digital Audio Processing Method, and Digital Audio Processing Program | 08-17-2017 |
| 20170236530 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD, ENCODER AND ENCODING METHOD, DECODER AND DECODING METHOD, AND PROGRAM | 08-17-2017 |
| 20180025736 | ENCODING AND DECODING DIGITAL DATA SETS | 01-25-2018 |
| 20180025737 | DECODING AUDIO BITSTREAMS WITH ENHANCED SPECTRAL BAND REPLICATION METADATA IN AT LEAST ONE FILL ELEMENT | 01-25-2018 |
| 20180025738 | DECODING AUDIO BITSTREAMS WITH ENHANCED SPECTRAL BAND REPLICATION METADATA IN AT LEAST ONE FILL ELEMENT | 01-25-2018 |
| 20180025739 | Time-Alignment of QMF Based Processing Data | 01-25-2018 |
| 20180026728 | DATA SENDING/RECEIVING METHOD AND DATA TRANSMISSION SYSTEM OVER SOUND WAVES | 01-25-2018 |
| 20190147892 | APPARATUSES AND METHODS FOR ENCODING AND DECODING A MULTICHANNEL AUDIO SIGNAL | 05-16-2019 |
