Abstract: This invention improves the perceived quality of frequency-domain time scale modification by selection of spectral bands used in phase locking based upon a Bark scale according to the variation in human hearing frequency response. A spectral peak is identified for each band. At these peaks the phases are rotated using the phase vocoder algorithm. For a few spectral lines near these peaks, the phase differences are copied from the non-rotated spectrum. The number selected is preferably 4. Remaining spectral lines within each spectral band located farther from the peak are phase rotated using the phase vocoder algorithm. The boundaries of the spectral bands may be adjusted based upon the digital audio data to maintain important frequency groups within the same spectral band.

Abstract: 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.

Abstract: In a signal encoding apparatus (1) a frequency normalization unit (11) normalizes each spectrum of spectral signals by using respectively normalization factors and supplies a normalization factor index per spectrum to a quantization accuracy determining unit (13). The quantization accuracy determining unit (13) adds a weighting factor using auditory properties to the normalization factor index per spectrum of range conversion spectral signals which are subjected to normalization as well as range conversion, and the quantization accuracy is determined according to the result of addition. Then, a quantization unit (14) performs quantization with the quantization accuracy corresponding to a quantization accuracy index supplied from the quantization accuracy determining unit (13), while the encoding/code string generating unit (15) encodes the weighting factor supplied from the quantization accuracy determining unit (13), together with the normalization factor index and the quantized spectral signal.

Abstract: Techniques for generating a target digital media item based on a source digital media item are described. A digital media item may be a song, a video clip, an album, or any length of audio or video. When adjusting the bit count for a portion of the target digital media item, instead of using the same set of parameter values used in a perceptual model for each portion of the source media item, the set of parameter values may be modified to encode the portion of the source digital media item. In this way, how audio or video is perceived is taken into account when adjusting a proposed bit count for a given portion of the target digital media item. Thus, while maintaining the same statistical bitrate as before increased digital media quality is achieved.

Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

Abstract: A speech band extension device (100), which generates an audio signal capable of realizing natural audibility after speech band extension, includes a band-extended audio generator which generates a band-extended audio signal from an original audio signal, the band-extended audio signal including components lying within a frequency band that is not included in a frequency band of the original audio signal, and an adjustment adder (20) which detects a timing shift between the original audio signal and the band-extended audio signal, adjusts timing of the original audio signal and timing of the band-extended audio signal in accordance with the detected timing shift, and combines the both signals after the adjusting of the timing, wherein the detection of the timing shift is performed, for example, using zero-crossing and cross-correlation.

Abstract: A “STAC Codec” provides lossless audio compression and decompression by processing an audio signal using integer-reversible modulated lapped transforms (MLT) to produce transform coefficients. Transform coefficients are then encoded using a backward-adaptive run-length Golomb-Rice (RLGR) encoder to produce losslessly compressed audio signals. In additional embodiments, further compression gains are achieved via an inter-block spectral estimation and data sorting strategy. Further, compression in the transform domain allows the bitstream to be partially decoded, using the corresponding RLGR decoder, to reconstruct the frequency-domain coefficients. These frequency-domain coefficients are then directly used to speed up various transform-domain based applications such as transcoding media to lossy or other formats, search, identification, visualization, watermarking, etc.

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: An apparatus for replicating an engine and/or exhaust sound of a predetermined vehicle includes files stored in computer-readable form in a memory library of the engine and/or exhaust sound of the predetermined vehicle. A microcomputer operated controller receives engine data such as the RPM of the engine of the personal vehicle and continually selects which files to obtain and use. The controller adjusts the sound pattern to optimally correspond in a preferred way with the RPM of the engine of the personal vehicle and the controller provides an output signal that is amplified and fed to a speaker for audio playback of a replicated sound of the engine and/or exhaust sound of the predetermined vehicle in the personal vehicle. A speaker disposed outside the vehicle includes a conical or cylindrical shape. A diagnostic capability is also disclosed. Connection to an OBD II connector provides the desired engine data.

Abstract: A parameter calculator calculates lower resolution parametric information and interpolation information. On a decoder-side, an upmixer is used for generating the output channels. The upmixer uses high resolution parametric information generated by a parameter interpolator using the low resolution parametric information and decoder-side derived interpolation information or encoder-generated interpolation information for selecting one of a plurality of different interpolation characteristics.

Abstract: There is provided a system capable of distributing code-compressed data based on audio data on a music composition via the Internet to a mobile telephone so that a user can cut out a desired range from the code-compressed data and register it as a call sound. The system has a data structure of a content frame (3GPP, 3GPP2) containing code-compressed data (AAC) of audio data. The content frame has at least one cut-out position information in the AAC data in its extended function section. A mobile telephone has a content storage unit, a cut-out selection unit to be used by the user to select at least one cut-out position information contained in the extended function section of the content frame, and a data cut-out section for cutting out data from the code-compressed data. The code-compressed data which has been cut out is decompressed when called and the sound is outputted from a loudspeaker.

Abstract: An intermediate channel representation of a multi-channel signal can be reconstructed highly efficient and with high fidelity, when upmix parameters for upmixing a transmitted downmix signal to the intermediate channel representation are derived that allow for an upmix using the same upmixing algorithms as within the multi-channel reconstruction. This can be achieved when a parameter re-calculator is used to derive the upmix parameters that takes into account also parameters having information on channels that are not included in the intermediate channel representation.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A method for changing reproduction speed of speech sound, includes the steps of: storing an input sound signal in a buffer; leaving a sound signal from the buffer as it is or extending the sound signal from the buffer in a sound section where a power of the input sound signal exceeds a threshold value; leaving the sound signal from the buffer as it is, compressing the sound signal from the buffer, or extending the sound signal from the buffer, in a no-sound section, so that the reproduction speed of speech sound is changed; wherein a speech head protection section is set prior to the sound section being set to be a storing amount of the buffer limited by a designated limited value; and compression or deletion of the sound signal is adjusted by a compression ratio or prevented if there is the sound section in the speech head protection section, so that speech head protection is performed.

Abstract: A method for encoding sound signals on multiple channels includes extracting an arbitrary number of sine waves from each of the sound signals. The sine waves include at least a first sine wave, extracted from a first one of the channels and having first-channel information, and a second sine wave, extracted from a second one of the channels and having second-channel information. Using the first-channel information and one of the second-channel information and sine wave information corresponding to a predetermined sine wave, one of the second-channel information and the sine wave information corresponding to the predetermined sine wave is selected as a to-be-correlated object for encoding in a correlation with the first-channel information.

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 time-scale of a digital signal is efficiently modified. A system suitable for embedded or stand-alone processing includes a module that can transform the time-scale of the signal according to a user's preference. An improved method for time-scale modification is based on envelope-matching but introduces a new function that is very fast to compute, the use of which avoids the computation of correlation coefficients where they are not needed. The invention is demonstrably faster than other methods related to SOLA (synchronized-overlap-and-add) with envelope matching, yet with no sacrifice in quality of the processed output.

Abstract: 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: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state (715) located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation (745) of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: In an audio signal processing procedure, auto-regressive (AR) modeling is used to create a residual signal from an input audio signal. The residual signal is further added to the input audio in order to produce a processed output audio signal. The AR modeling can be performed frame-by-frame or sample-by-sample employing frequency warped Burg's method.

Abstract: 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.

Abstract: 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.

Abstract: An audio encoder, which is capable of encoding multiple-channel signals so that only a downmixed signal is decoded and of further generating specific auxiliary information necessary for dividing the downmixed signal, is provided. An audio encoder (10), which compresses and encodes audio signals of N-channels (N>1), includes a downmixed signal encoding unit (11) which encodes the downmixed signal obtained by downmixing the audio signals, and an auxiliary information generation unit (12a) which generates auxiliary information necessary for decoding the downmixed signal encoded by the downmixed signal encoding unit (11) into N-channel audio signals.

Abstract: An audio encoding scheme or a stream that encodes audio and video data is disclosed. The scheme has particular application in mezzanine-level coding in digital television broadcasting. The scheme has a mean effective audio frame length F that equals the video frame length 1/fV over an integral number M video frames, by provision of audio frames variable in length F in a defined sequence where length=F(j) at encoding. The length of the audio frames may be varied by altering the length of overlap between adjacent frames in accordance with an algorithm that repeats after a sequence of M frames. An encoder and a decoder for such a scheme are also disclosed.

Abstract: In one embodiment, at least one audio data frame having at least one channel is generated. Each channel is divided into a plurality of blocks. A sub-block partitioning scheme is selected, and a number of sub-blocks into which the block is to be partitioned is selected. The selected number of sub-blocks is chosen from numbers of sub-blocks available for the selected sub-block partitioning scheme. The block of audio data is partitioned into sub-blocks according to the selected number of sub-blocks, and the partitioned sub-blocks are coded according to a selected entropy coding scheme.

Abstract: A scalable audio data arithmetic decoding method, medium, and apparatus, and a method, medium, and apparatus truncating an audio data bitstream. The arithmetic decoding method of decoding a scalable arithmetic coded symbol may include arithmetic decoding of a symbol by using the symbol and a probability value for the symbol desired to be decoded, and determining whether or not to continue decoding by checking an ambiguity indicating whether or not decoding of the symbol to be decoded is completed. According to a method, medium, and apparatus of the present invention, data to which scalability is applied when arithmetic coding is performed in MPEG-4 scalable lossless audio coding can be efficiently decoded. Even when a bitstream is truncated, a decoding termination point can be known such that additional decoding of the truncated part can be performed.

Abstract: A data redundancy elimination system.

Abstract: An audio data processing device including: a first processor; and a second processor which is connected to the first processor wherein the first processor includes: an audio data acquisition which acquires audio data of digital data; an omitting section which omits a bit corresponding to low volume which is hard to be heard by human ears from the audio data; and a transmitter which transmits the audio data in which the bit corresponding to the low volume is omitted by the omitting section from the first processor to the second processor; wherein the second processor includes: a receiver which receives the audio data transmitted from the first processor; and a reproduction data generator which generates audio reproduction data necessary to reproduce the audio data based on the received audio data.

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: Provided is a method and apparatus for encoding/decoding a multi-channel audio signal. The apparatus for encoding a multi-channel audio signal includes a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed multi-channel audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.

Abstract: An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a pilot reference value corresponding to a plurality of data and a pilot difference value corresponding to the pilot reference value and obtaining the data using the pilot reference value and the pilot difference value.

Abstract: 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.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A decoder receives (501) a bitstream comprising an encoded mono signal and stereo data. A time scale processor (503) generates a time scaled mono signal. A time-to frequency processor generates frequency sample blocks of the time scaled signal, the block length being fixed and independent of the time scaling. A parametric stereo decoder (509) generates a stereo signal for the frequency sample blocks and these are converted to the time domain by a frequency-to-time processor (511). A synchronization processor (515) synchronizes the stereo data with the time scaled signal by determining a time association between a parameter value and a frequency sample block. The parameter value and time association is used to determine synchronized stereo parameter values for that and other frequency sample blocks. The invention is particularly suitable for low complexity generation of time scaled stereo signals from MPEG-4 encoded signals.

Abstract: The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

Abstract: Signals of different channels are combined into one mono signal. A set of adaptive filters, preferably one for each channel, is derived in a respective filter adaptation unit. When an adaptive filter is applied to the mono signal it reconstructs the signal of the respective channel under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal, in encoded form, and the parameters of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract: An MPEG-1 layer 3 audio encoder, including a scalefactor generator for determining first scalefactors for encoding a block of audio data if a temporal masking transient is not detected in said block of audio data; and for selecting the maximum of said scalefactors for encoding said block of audio data if a temporal masking transient is detected in said block of audio data to enable greater compression of said audio data. Increases in quantization error, due to use of the maximum scalefactor are pre-masked or post-masked by the temporal masking transient. In cases where the last portion of a block includes a temporal masking transient that masks the preceding portions of the block, the maximum scalefactor is only used to encode the block if the resulting increase in quantization error is less than 30% of the quantization error for the block.

Abstract: Coding an audio signal of a sequence of audio values into a coded signal includes determining first and second listening thresholds for first and second blocks of audio values of the sequence of audio values; calculating versions of first second parameterizations of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first and second listening thresholds, respectively; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block which is quantized; forming a difference between the versions of the first and second parameterizations; integrating information on, inter alias, the difference into the coded signal.

Abstract: A voice and musical tone coding apparatus is provided that can perform high-quality coding by executing vector quantization taking the characteristics of human hearing into consideration. In this voice and musical tone coding apparatus, a quadrature transformation processing section (201) converts a voice and musical tone signal from time components to frequency components. An auditory masking characteristic value calculation section (203) finds an auditory masking characteristic value from a voice and musical tone signal. A vector quantization section (202) performs vector quantization changing a calculation method of a distance between a code vector found from a preset codebook and a frequency component based on an auditory masking characteristic value.

Abstract: A method for providing a motion signal with a sound signal using an existing sound signal encoding format. The method comprises providing the motion signal, providing the sound signal, inserting the motion signal in an available data field provided in the existing encoding algorithm, encoding the sound signal with the inserted motion signal according to the existing encoding algorithm to generate an encoded bitstream sound signal and providing the encoded bitstream sound signal comprising the motion signal and the sound signal.

Abstract: A multi-channel synthesizer for generating at least three output channels using an input signal having at least one base channel, the base channel being derived from the original multi-channel signal, the input signal further including at least two different up-mixing parameters, and an up-mixer mode indication indicating, in a first state that a first up-mixing rule is to be performed, and, indicating, in a second state, that a different second up-mixing rule is to be performed, uses an up-mixer for up-mixing the at least one base channel using the at least two different up-mixing parameters based on the first or the second up-mixing rule in response to the up-mixer mode indication so that the at least three output channels are obtained.

Abstract: There is provided a communication device for effectively encoding an audio/music signal while maintaining a predetermined quality by controlling the transmission bit rate of the transmission side considering the use environment of the reception side. In this device, a transmission mode decision unit (101) detects an environment noise contained in the background of the audio/music signal in the input signal and decides the transmission mode controlling the transmission bit rate of the signal transmitted from a communication terminal device (150), which is a communication terminal of the partner side, according to the environment noise level. A signal decoding unit (103) decodes encoded information transmitted from the communication terminal device (150) via a transmission path (110) and outputs the obtained signal as an output signal.

Abstract: A system extends a bandwidth of bandlimited audio signals by analyzing bandlimited audio signals at a transmission cycle rate. The analyzer may obtain a bandlimited parameter at a transmission cycle rate. A mapping device or logic in the system obtains a wideband parameter based on the bandlimited parameter. An audio signal generator generates a highband and/or lowband audio signal based on the wideband parameter at the transmission cycle rate. In some systems, the bandlimited audio signal is analyzed at the transmission cycle rate. The highband and/or lowband audio signals and the combined wideband audio signal are generated at the transmission cycle rate.

Abstract: A sound signal encoder for high efficiency encoding of sound signals from a plurality of channels is provided which includes a to-be-correlated object setter (52), to-be-correlated object selector (56) and a variable-length encoder (58). The to-be-correlated object setter (52) sets, on the basis of left-channel frequency information held in a left-channel frequency information holder (50) and right-channel frequency information held in a right-channel frequency information holder (51), index [i] indicating which ones of sine waves on the left channel are to be correlated with, namely, are to be subtracted from, sine waves on the right channel. The to-be-correlated object selector (56) selects a default value read from a storage unit (55) or index [i]-th amplitude information read from a left-channel amplitude information holder (53) as an object to be subtracted from the i-th amplitude information on the right channel according to the index [i].

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: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.