Abstract: A module (402) for binary coding of a signal envelope, comprising coding module (502) for coding a variable length first mode. The coding module for coding the first mode incorporates an envelope saturation detector and said coding module (402) further includes a second coding module (503) for coding a second mode in parallel with the coding module (502) for coding the first mode and a mode selector (504) adapted to select one of the two coding modes as a function of a code length criterion and of the result from the envelope saturation detector.

Abstract: A decoding device (30a) comprises a narrow-band decoding unit (31) operable to reproduce a PCM signal (P1) from a narrow-band bit stream included in a wide-band bit stream (S0), a wide-band decoding unit (32) operable to reproduce a PCM signal (P2) having a frequency band which is wider than that of the PCM signal (P1) reproduced by the narrow-band decoding unit (31) from the narrow-band bit stream and a band expanding bit stream included in the wide band bit stream (S0) and a selecting unit (34) operable to select either the PCM signal (P1) reproduced by the narrow-band decoding unit (31) or the PCM signal (P2) reproduced by the wide-band decoding unit (32), and to output the selected sound digital signal.

Abstract: 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.

Abstract: A method of encoding a time-domain audio signal is presented. In the method, an electronic device receives the time-domain audio signal. The time-domain audio signal is transformed into a frequency-domain signal including a coefficient for each of a plurality of frequencies, which are grouped into frequency bands. For each frequency band, the energy of the band is determined, a scale factor for the band is determined based on the energy of the band, and the coefficients of the band are quantized based on the associated scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors.

Abstract: An encoding device reduces the encoding distortion as compared to the conventional technique and obtains a preferable sound quality for auditory sense. In the encoding device, a shape quantization unit quantizes the shape of an input spectrum with a small number of pulse positions and polarities. The shape quantization unit sets a pulse amplitude width to be searched later upon search of the pulse position to a value not greater than the pulse amplitude width which has been searched previously. A gain quantization unit calculates a gain of a pulse searched by the shape quantization unit for each of bands.

Abstract: An audio signal expansion and compression method for expanding and compressing an audio signal in a time domain, includes the steps of setting an initial value of a signal comparison length of a first comparison interval and a second comparison interval, used for detection of two similar waveforms in the audio signal, equal to or larger than a minimum waveform detection length, determining an interval length of the two similar waveforms while changing a shift amount of the first comparison interval and the second comparison interval so that the shift amount does not exceed the signal comparison length, and expanding or compressing the audio signal in the time domain on the basis of the interval length of the two similar waveforms.

Abstract: In one method embodiment, receiving from the network device a multiplex of a compressed video stream and a compressed audio stream, the multiplex comprising a succession of intervals corresponding to a video program corresponding to a first playout rate; and at the start of each interval, replacing the compressed audio stream with a compressed, pitch-preserving audio stream corresponding to a second playout rate different than the first.

Abstract: An audio waveform processing not imparting any feeling of strangeness and high in definition, in which time stretch and pitch shift are performed by a vocoder method, and the variation of phase over the whole waveform caused by the vocoder method at all times is reduced. An audio input waveform is handled as one band as it is or subjected to frequency band division into bands. While performing time stretch and pitch shift of each band waveform like conventional vocoder methods, the waveforms are combined. The combined waveform of the band is phase-synchronized at regular intervals to reduce the variation of phase. The phase-synchronized waveforms of the band are added, thus obtaining the final output waveform.

Abstract: The present invention provides a signal processing apparatus, a signal processing method and a program for outputting a high-quality coded string. A signal processing apparatus according to an embodiment of the present invention includes a normalization coefficient information increasing/decreasing circuit 12 for modifying normalization coefficient information of a signal component of a frame and normalization coefficient information of a primary additional signal component according to a normalization coefficient information primary increase/decrease amount, and an additional signal component normalization coefficient information increasing/decreasing circuit 14 for modifying normalization coefficient information of a secondary additional signal component, which is a copy of the primary additional signal component, according to a normalization coefficient information secondary increase/decrease amount.

Abstract: A system, method and computer program product for classification of an analog electrical signal using statistical models of training data. A technique is described to quantize the analog electrical signal in a manner which maximizes the compression of the signal while simultaneously minimizing the diminution in the ability to classify the compressed signal. These goals are achieved by utilizing a quantizer designed to minimize the loss in a power of the log-likelihood ratio. A further technique is described to enhance the quantization process by optimally allocating a number of bits for each dimension of the quantized feature vector subject to a maximum number of bits available across all dimensions.

Abstract: A method, medium, and apparatus encoding and/or decoding an audio signal to surround data. While encoding spatial information, which can up-mix an audio signal to a surround signal, to extension data, a length of a payload corresponding to the spatial information is encoded and a payload of the spatial information is decoded using the length of the payload. Accordingly, compatibility of the spatial information can be provided, and the spatial information can be transmitted by effectively embedding the spatial information.

Abstract: 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.

Abstract: Embodiments are generally directed to systems and methods for bit allocation and band partitioning for gain-shape vector quantization in an audio codec. An audio codec implements a method that uses an implicit, dynamic scheme to allow an encoder and decoder to recreate a series of bit allocation decisions for gain and shape without transmitting additional side information for each decision, based on the number of bits that are left remaining and available in a given packet. For implementation in practical codecs, the band comprising the allocation of bits for the shape is recursively split into equal partitions until the number of bits allocated to each partition is less than the maximum codebook size.

Abstract: 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.

Abstract: 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.

Abstract: An exemplary system and method are directed at receiving an audio signal and process the audio signal into a remapped audio signal based on a plot profile. The plot profile may include at least one of an identified range of audio frequencies. The processing may comprise retrieving an identified range of audio frequencies from the plot profile; determining a range of impaired audio frequencies in the audio signal based on the identified range of audio frequencies; shifting the frequency of at least a portion of the impaired audio frequencies to outside of the identified range; and continuing to retrieve identified ranges of audio frequencies from the plot profile. The shifting of the impaired audio frequencies of the audio signal may be performed until no further identified ranges of audio frequencies are available for consideration.

Abstract: 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.

Abstract: 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.

Abstract: There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).

Abstract: 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.

Abstract: In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters.

Abstract: Presented herein are systems and methods for processing sound signals for use with electronic speech systems. Sound signals are temporally parsed into frames, and the speech system includes a speech codebook having entries corresponding to frame sequences. The system identifies speech sounds in an audio signal using the speech codebook.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: The mobile radio terminal includes a speech input unit which inputs a speech signal obtained from speech of a speaking person, an estimating unit which estimates a speech style of the speaking person from the speech signal, and a converting unit which converts the speech signal into a converted speech signal in accordance with the estimated speech style.

Abstract: A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

Abstract: An electronic device includes a connector, at least two kinds of signal transmitting circuits, and a selecting system. The connector is capable of transmitting audio and video signals when connected to a peripheral device. The selecting system includes a detecting module, a memory module, a comparing module, and a connecting module. The detecting module is capable of detecting and measuring the voltage of a signal line of the connector. The memory module is capable of storing predetermined voltage ranges corresponding to different kinds of peripheral devices. The comparing module is capable of comparing the voltage of the signal line measured by the detecting module to the predetermined voltage ranges and determining what kind of peripheral device is connected to the connector. The connecting module is capable of connecting the connector to one of the signal transmitting circuits according to the comparing module.

Abstract: A digital electronic device includes a time shifter/tone adapter that eliminates unwanted audio effects at recipient device, that includes stutter and loss of audio synchronization, as a result of video quality adaptation (the video quality adapter varies frame rate, pixel and color resolutions without having a discernable difference in picture quality, that is, drops many frames in every frame set). The tone adaptation involves gradual frequency shifting, that is, gradual up shifting until synchronization with video is obtained (time shifting), then gradual down shifting. The recipient device (or a set top box) may contain a time shifter/tone adapter that eliminates unwanted audio effects at the recipient devices that may include stutter and loss of audio synchronization, as a result of loss of packets in channel.

Abstract: A method for the selection of synthesis units of a piece of information that can be decomposed into synthesis units, comprises at least the following steps for a considered information segment: determining the mean fundamental frequency value F0 for the information segment considered; selecting a sub-set of synthesis units defined as being the sub-set whose mean pitch values are the closest to the pitch value F0; applying one or more proximity criteria to the selected synthesis units to determine a synthesis unit representing the information segment.

Abstract: A speech decoding device of the invention smoothes, in decoding speech signal in a voice-less period, RMS and filter coefficients which is discontinuously transmitted, and provides them to a synthesis filter. Thereby, it is capable of preventing discontinuous changing of the filter coefficient caused by the intermittent transmission of the filter coefficient. As a result, a quality of decoding can be improved. Also, to remove an effect, caused by the smoothing process, from the filter coefficients or the RMS which are transmitted in the past frames, a smoothing factor is adjusted not to perform smoothing while a certain time period (or a certain number of frames) from when a transition is made from a voice period from a voice-less period, or when a decoded feature parameter satisfies a predetermined condition.

Abstract: In one alternative, an audio signal is analyzed using multiple psychoacoustic criteria to identify a region of the signal in which time scaling and/or pitch shifting processing would be inaudible or minimally audible, and the signal is time scaled and/or pitch shifted within that region. In another alternative, the signal is divided into auditory events, and the signal is time scaled and/or pitch shifted within an auditory event. In a further alternative, the signal is divided into auditory events, and the auditory events are analyzed using a psychoacoustic criterion to identify those auditory events in which the time scaling and/or pitch shifting processing of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: Playback by a decoder of a lossy compressed digital media file without quantization gaps is disclosed. The digital media file can be formed of a number of audio samples grouped into a corresponding number of audio frames. As a method, one embodiment can be carried out by identifying an encoder used to compress the digital media file; obtaining an encoder delay value for the identified encoder; obtaining a decoder delay value for the decoder; determining a audio sample count corresponding to a last valid audio sample; setting a re-synchronization after seek option marker N audio frames from the last valid audio sample; and decoding valid audio samples using the encoder delay value, the decoder delay value, and the sample count corresponding to the last valid audio sample.

Abstract: 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.

Abstract: A system and method to change codec information to provide a coloring service in a Voice over Internet Protocol (VoIP) terminal uses different compression methods depending on a calling state and a call connecting state between the VoIP terminals so that a more efficient coloring service is provided. The system for changing codec information includes a gateway adapted to compress ring back tone data in a calling state and to compress voice signal data in a call connecting state between communication terminals according to preset different compression information and to transmit both data to a receiving terminal.

Abstract: Method of processing a transmitted encoded media data stream is received. If a data element arrives prior to, or at, a predetermined playout deadline, the data element is decoded, the media represented by the decoded data element is played, and the data element is provided to a decoder state machine to update a decoder state. If a data element arrives after the predetermined playout deadline, the data element is provided to the decoder state machine to update the decoder state. In one embodiment, if the specified data element fails to arrive by the playout deadline, a subsequently received data element is saved in memory. Then, if the specified data element arrives after the predetermined playout deadline, the specified data element and the saved, subsequently received, data element are provided to the decoder state machine to update the decoder state.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

Abstract: Disclosed is a reproducing apparatus comprising: a reproduction section to reproduce reproduction data comprising sound data and/or image data; a selection section to calculate evaluation values between a link source set for the reproduction data and each of a plurality of link destinations corresponding to the link source by a predetermined arithmetic expression based on link information of the plurality of link destinations, and to select a link destination having a highest evaluation among the evaluation values out of the plurality of link destinations; and a reproduction control section to move a reproduction point of the reproduction data reproduced by the reproduction section to a position corresponding to the link destination by linking the link source with the link destination when the reproduction point reaches a given point with respect to a position corresponding to the link source, and to instruct the reproduction section to reproduce the reproduction data.

Abstract: Apparatus and method for generating high quality synthesized speech having smooth waveform concatenation. The apparatus includes a pitch frequency calculation section, a pitch synchronization position calculation section, a unit waveform storage, a unit waveform selection section, a unit waveform generation section, and a waveform synthesis section. The unit waveform generation section includes a conversion ratio calculation section, a sampling rate conversion section, and a unit waveform re-selection section. The conversion ratio calculation section calculates a sampling rate conversion ratio from the pitch information and the position of pitch synchronization, and the sampling rate conversion section converts the sampling rate of the unit waveform, delivered as input, based on the sampling rate conversion ratio.

Abstract: A lossless encoding and/or decoding apparatus which encodes audio data on a real-time basis includes a lossless compression unit which losslessly compression encodes the audio data stored in an input buffer in units of predetermined data and outputs the encoded data in sequence, and an output buffer which stores the encoded audio data output from the lossless compression unit. A bitrate controller divides a plurality of the encoded audio data stored in the output buffer into first data having a data amount exceeding the maximum bitrate and second data having a data amount less than the maximum bitrate, divides the first data into third data being the encoded audio data having a data amount of the maximum bitrate and fourth data being the encoded data of the portion exceeding the maximum bitrate, and controls the output buffer so that the fourth data is output together with the second data.

Abstract: This invention involves time-scale modification of audio signals. The invention describes overlap and add time scale modification with variable input and output buffer sizes. Seamless speed change is achieved by keeping track of previously processed data to avoid discontinuities during playback speed transitions.

Abstract: Disclosed is a stereo audio encoding device capable of improving a spatial image of a decoded audio in stereo audio encoding. In this device, an original cross correlation calculation unit (101) calculates a mutual relationship coefficient (C1) between the original L channel signal and the original R channel signal. A stereo audio reconfiguration unit (104) subjects the inputted L channel signal and the R channel signal to encoding and decoding so as to generate an L channel reconfigured signal (L?) and an R channel reconfigured signal (R?). A reconfiguration cross correlation calculation unit (105) calculates a cross correlation coefficient (C2) between the L channel reconfigured signal (L?) and the R channel reconfigured signal (R?). A cross correlation comparison unit (106) calculates and outputs a comparison result &agr; between the cross correlation coefficient (C1) and the cross correlation coefficient (C2).

Abstract: Systems are disclosed for operating a communications network. The system includes a module to buffer frames of a signal, and a module to determine an access delay. The system also includes a module to compress a portion of the signal based on the access delay by removing a first portion of a frame of the signal and generating an overlap-added segment from a first segment and a second segment of the frame. In another embodiment, the system includes a module to buffer frames of a signal, a module to establish a communication channel with a handset, and a module to determine an access delay. The system also includes a module to compress a portion of the signal based on the access delay by removing a first portion of a frame of the signal and generating an overlap-added segment from a first segment and a second segment of the frame.

Abstract: In a multi-channel encoder generating several different parameter sets for reconstructing a multi-channel output signal using at least one transmission channel, the data stream is written such that the two parameter sets are decodable independently of each other. Thus, a multi-channel decoder is enabled to skip a parameter set which is marked as optional and/or has a higher version number when reading the data stream and still to perform a valid multi-channel reconstruction using a data set marked as mandatory or a data set having a sufficiently low version number. This achieves a flexible encoder/decoder concept suitable for future updates characterized by backward compatibility and reliability.

Abstract: A method, device and system for signal encoding and decoding, the method comprising: 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. In embodiments of the present invention, 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.

Abstract: Methods and apparatuses for encoding a signal and decoding a signal and a system for encoding and decoding are provided. The method for encoding a signal includes performing a classification decision process on high frequency signals of input signals, adaptively encoding the high frequency signals according to the result of the classification decision process, and outputting a bitstream including codes of low frequency signals of the input signals, adaptive codes of the high frequency signals, and the result of the classification decision process. The classification decision process is performed on the high frequency signals, and adaptive encoding or adaptive decoding is performed according to the result of the classification decision process, so the quality of voice and audio output signals is improved.

Abstract: Embodiments related to utilizing substantially optimal entries for a relatively low complexity dictionary for matching pursuits coding is disclosed. In various embodiments, methods are invoked for determining a substantially optimal entry from a bases dictionary comprising a plurality of entries; and utilizing the substantially optimal entry in a relatively low complexity matching pursuits data coding. In various embodiments, a system is provided comprising a bases dictionary comprising a plurality of entries each with a width of 15 or less; a signal to be coded; and a selection module configured to receive at least one of the plurality of entries from the bases dictionary, to calculate an inner product between the at least one of the plurality of entries and the signal to be coded, and to select the entry from the at least one of the plurality of entries that produces a maximum inner product for use in at least partially coding the signal to be coded.

Abstract: The disclosure relates to a communication terminal having a bandwidth expansion device for expanding the bandwidth of a narrowband voice signal, on a low-frequency and/or high-frequency side, by synthesizing at least one frequency band on the basis of the narrowband voice signal. A qualitatively satisfactory bandwidth expansion is thus performed using a plurality of net bit rates. The bandwidth expansion device is further connected to a memory containing a lookup table comprising at least one parameter value for the bandwidth expansion, for at least two net bit rates of the narrowband voice signal. A method for expanding a bandwidth of a narrowband voice signal having at least two net bit rates in a communication terminal is also disclosed herein.

Abstract: An encoding device and an encoding method are provided for encoding by reducing the number of samples to be processed 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. When encoding higher-band spectrum data in a signal to be encoded, according to lower-band spectrum data in the signal, only for a part (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.

Abstract: 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.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes 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.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information.