Abstract: In a coder, a method for producing forward aliasing cancellation (FAC) parameters for cancelling time-domain aliasing caused to a coded audio signal in a first transform-coded frame by a transition between the first transform-coded frame using a first coding mode with overlapping window and a second frame using a second coding mode with non-overlapping window, comprising: calculating a FAC target representative of a difference between the audio signal of the first frame prior to coding and a synthesis of the coded audio signal of the first transform-coded frame; and weighting the FAC target to produce the FAC parameters. In a decoder, weighted forward aliasing cancellation (FAC) parameters are received and inverse weighted to produce a FAC synthesis. Upon synthesis of the coded audio signal in the first frame, the time-domain aliasing is cancelled from the audio signal synthesis using the FAC synthesis.

Abstract: A signal decoding apparatus that can suppress any large unusual sounds to provide decoded signals of improved audibility even when the number of hierarchical layers to be used in the decoding process varies due to a packet loss or the like in communication utilizing a scalable encoding/decoding technique. In the signal decoding apparatus, a gain adjusting part (2308) adjusts, based on a control of a decoding control part (2301), the gain of a basic layer decoded signal outputted from a basic layer decoding part (2302). A gain adjusting part (2309) adjusts, based on a control of the decoding control part (2301), the gain of a first expansion layer decoded signal outputted from a first expansion layer decoding part (2303). A gain adjusting part (2310) adjusts, based on a control of the decoding control part (2301), the gain of a second expansion layer decoded signal outputted from a second expansion layer decoding part (2304).

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 “STAC Codec” provides audio transcoding and decoding by processing an encoded audio signal using a backward-adaptive run-length Golomb-Rice (RLGR) decoder to recover transform coefficients of the encoded audio signal. The transform coefficients are then either transcoded in the transform domain to lossy or other formats, or decoded to the time domain by applying an inverse integer-reversible modulated lapped transform (MLT) to the recovered transform coefficients to recover an uncompressed time domain representation compressed audio signal. In additional embodiments, an inter-block spectral estimation and inverse data sorting strategy is used in recovering the transform coefficients from the encoded audio signal.

Abstract: An audio-signal time-axis expansion/compression method for subjecting an audio signal to time-axis expansion/compression at a time domain includes the steps of: cross-fade-signal generating wherein a first period and a second period which are similar within the audio signal are employed to generate the cross-fade signal of the first period signal and the second period signal; correction-signal generating wherein the difference signal between the first period signal and the second period signal is subjected to time-axis reversal, and is multiplied with a window function to generate a correction signal; and connection-waveform generating wherein the cross-fade signal and the correction signal are added to generate a connection waveform for subjecting the audio signal to time-axis expansion/compression at the time domain.

Abstract: Systems and methods for booting a programmable processor such as a DSP that is incorporated into an HDA codec. The codec and a system memory containing boot program instructions are connected to an HDA bus. In a first mode, the DSP receives boot program instructions via the HDA bus and boots using these instructions. In a second mode, the DSP boots from instructions that are contained in a memory that is connected to the DSP. In one embodiment, the memory connected to the DSP is a component of a plug-in card, and the DSP is configured to determine whether the plug-in card is present, then boot from the memory on the plug-in card if it is present or boot from the system memory via the HDA bus if the plug-in card is not present.

Abstract: A modified synchronized overlap add (SOLA) algorithm for performing high-quality, low-complexity audio time scale modification (TSM) is described. The algorithm produces good output audio quality with a very low complexity and without producing additional audible distortion during dynamic change of the audio playback speed. The algorithm may achieve complexity reduction by performing the maximization of normalized cross-correlation using decimated signals. By updating the input buffer and the output buffer in a precise sequence with careful checking of the appropriate array bounds, the algorithm may also achieve seamless audio playback during dynamic speed change with a minimal requirement on memory usage.

Abstract: A plurality of pairs of segments to be weighted/added are selected non-linearly with respect to a time axis of audio data. A speed conversion is achieved by performing the weighting/addition on the selected pairs of segments. The non-linear selection is performed by (a) obtaining all possible pairs of segments constituting the audio data, (b) calculating a degree of similarity pertaining to each possible pair, (c) ranking the all possible pairs of segments according to the degrees of similarity, and (d) overlapping at least one of the all possible pairs of segments that holds the highest degree of similarity.

Abstract: To provide an acoustic signal processing apparatus which can reduce the amount of calculation in matrix arithmetic. An acoustic signal processing apparatus converts down-mixed acoustic signals of NI channels to acoustic signals of NO channels, where NO>NI.

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 scalable encoding method, apparatus, and medium. The method includes: encoding a base layer and encoding a first enhancement layer and a second enhancement layer in a frame having the base layer; and generating an encoded frame by synthesizing the encoded results. Accordingly, only if the loss of the encoding frame is not as great as the encoded first enhancement layer is damaged, a case where speech restoration with respect to partial frequency bands must be given up does not occur. Furthermore, since an encoder divides the second enhancement layer into a plurality of layers in a horizontal or vertical direction, considering a distribution pattern of data belonging to the second enhancement layer and first encodes a layer in which lots of data are distributed among the divided layers, loss of audio information can be minimized even if a portion of the encoded second enhancement layer is damaged.

Abstract: Circuits and methods for providing zero-gap playback of consecutive data streams in portable electronic devices, such as media players, are described. In some embodiments, a circuit includes a decoder circuit configured to receive encoded audio data and to output decoded audio data including data streams associated with a data file and a subsequent data file. Moreover, a predictive circuit, which is electrically coupled to the decoder circuit, is configured to selectively generate additional samples based on samples in the data file, where the additional samples correspond to times after the end of a data stream associated with the data file. Additionally, a filter circuit, which is electrically coupled to the decoder circuit and selectively electrically coupled to the predictive circuit, is configured to selectively combine or blend samples at a beginning of the subsequent data file with the additional samples. Note that the circuit may be included in an integrated circuit.

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: A method for supporting an encoding of an audio signal is shown, wherein at least a first and a second coder mode are available for encoding a section of the audio signal. The first coder mode enables a coding based on two different coding models. A selection of a coding model is enabled by a selection rule which is based on signal characteristics which have been determined for a certain analysis window. In order to avoid a misclassification of a section after a switch to the first coder mode, it is proposed that the selection rule is activated only when sufficient sections for the analysis window have been received. The invention relates equally to a module in which this method is implemented, to a device and a system comprising such a module and to a software program product including a software code for realizing the proposed method.

Abstract: A system and apparatus for establishing whether a received signal frame is an audio signal frame is disclosed. In one embodiment, the system includes a predetermined position in an audio signal frame containing a piece of secondary information for an audio characteristic of the audio data, with a selection device for selecting a succession of bits which is arranged at the predetermined position in the received signal frame. A decision-making device flags the received signal frame as an audio signal frame if the succession of bits represents the piece of secondary information.

Abstract: 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(0), s(1), . . . , s(N?1), respectively. The restoring step includes detecting s(0) from the audio signal, where s(0) represents the code parameter of the first sub-block; detecting a difference s(i)?s(i?1) from the audio signal for i=1, . . . N?1, where s(i) representing the code parameter of each sub-block following the first sub-block. The difference s(i)?s(i?1) is encoded by using first entropy code. The restoring step further includes calculating s(i) for i=1, . . . , N?1 using s(0) and the detected differences, and the method further includes decoding the N sub-blocks using the restored code parameters.

Abstract: There is described a method of encoding an input signal (20) to generate a corresponding encoded output signal (30), and also encoders (10) arranged to implement the method. The method comprises steps of: (a) distributing the input signal to sub-encoders (300, 310, 320) of the encoder (10); (b) processing the distributed input signal (20) at the sub-encoders (300, 310, 320) to generate corresponding representative parameter outputs (200, 210, 220) from the sub-encoders (300, 310, 320); and (c) combining the parameter outputs (200, 210, 220) of the sub-encoders (300, 310, 320) to generate the encoded output signal (30). Processing of the input signal (20) in the sub-encoders (300, 310, 320) involves segmenting the input signal (20) for analysis, such segments having associated temporal durations which are dynamically variable at least partially in response to information content present in the input signal (20).

Abstract: 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.

Abstract: This invention locally controls the pitch of speech and audio signals. The invention is based on a seamless time scale modification (S-TSM) scheme connected to a synchronized sampling rate converter that switches between different time scale factors in a seamless manner and controls pitch during playback in a nearly continuous way.

Abstract: An entropy encoder includes an apparatus for producing a data stream which comprises two reference points, of code words of variable lengths, the apparatus comprising a first device for writing at least a part of a code word into the data stream in a first direction of writing, starting from a first reference point, and a second device for writing at least a part of a code word into the data stream in a second direction of writing, which is opposite to the first direction of writing, starting from the other reference point. In particular, when a raster having a plurality of segments is used to write the code words of variable lengths into the data stream, the number of the code words which can be written starting at raster points is doubled, in the best case, such that the data stream of code words of variable lengths is robust toward a propagation of sequence errors.

Abstract: Methods and apparatus for coordinating audio data processing and network communication processing in a communication device are disclosed. An exemplary method begins with demodulating a series of received communication frames, using a network communication processing circuit, to produce received encoded audio frames. An event report for each of one or more of the received encoded audio frames is generated, the event report indicating a network communication circuit processing time associated with the corresponding received encoded audio frames. The received encoded audio frames are decoded, using an audio data processing circuit, and the decoded audio is output to an audio circuit. The timing of the outputting of the decoded audio is adjusted, based on the generated event reports.

Abstract: 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.

Abstract: 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.

Abstract: A source sampling rate is associated with first or second groups of sampling rates. A playback rate is determined by: (a) selecting the source sampling rate if the source sampling rate is supported by a playback environment; (b) otherwise if there is a highest first rate from the first or second groups of playback sampling rates which is supported by the playback environment and is lower than the source sampling rate, selecting the first rate; (c) otherwise if there is a slowest second rate from the group that the source sampling rate is associated with that is supported by the playback environment and is higher than the source sampling rate, selecting the second rate; (d) otherwise selecting the slowest sampling rate supported by the playback environment from the group that the source sampling rate is not associated with as the playback rate.

Abstract: In one embodiment, the method includes receiving the audio signal including at least one block of audio data and configuration information, and reading coding type information and partitioning information from the configuration information. The coding type information indicates an entropy coding scheme used in encoding the audio signal, and the partitioning information indicates a sub-block partition scheme by which the block is divided into sub-blocks. Sub-block information is read from the block of audio data, and the sub-block information indicates a number of the sub-blocks into which the block is partitioned given the sub-block partitioning scheme. The number of the sub-blocks is determined based on the entropy coding scheme and the sub-block partition scheme. The partitioned sub-blocks are decoded based on the entropy coding scheme.

Abstract: Techniques for determining scale factor values when encoding audio data are described. According to one technique, a particular scale factor value (SFV) is estimated using an audio quality estimator function that is non-linear. After a certain point, a decrease in noise results in a smaller increase in audio quality. According to another technique, an initial SFV is estimated for each scale factor band (SFB). When estimating the cost of transitioning from one SFB to another, only a proper subset of possible SFVs are considered. The proper subset is based, at least in part, on the initial SFV.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract: 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.

Abstract: 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.

Abstract: Techniques of this disclosure provide for adjustment of a conversion rate of a sampling rate converter (SRC) in real-time. The SRC determines relative timing of generated output samples based on non-approximated integer components that are recursively updated. The SRC may further base relative timing of output samples on a value of one or more step size components associated with the integer components. Also according to techniques of this disclosure, a conversion rate of an SRC may be adjusted in real-time based on a detected mismatch between a source clock of a digital input signal and a local clock.

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: A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.

Abstract: An audio coding device that optimizes quantization parameters for fast convergence of iterations. A quantized bit counter calculates a codeword length representing the number of bits of a Huffman codeword corresponding to quantized values. The quantized bit counter also calculates a codebook number bit count representing how many bits are consumed for optimal Huffman codebook numbers, and a scale factor bit count representing how many bits are consumed for scale factors of each subband. In a first stage of quantization, the quantized bit counter accumulates lengths of Huffman codewords corresponding to quantized values of every nth subband. A bit count estimator calculates a total bit count estimate by adding up n times the accumulated codeword length, the codebook number bit count, and the scale factor bit count. A parameter updater updates quantization parameters if the total bit count estimate exceeds a bit count limit.

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: Aspects of a method and system for decoding GSM speech data using redundancy are provided. A decoding algorithm in a frame process may be utilized to generate a bit-sequence for GSM speech data received via a burst process. The decoding algorithm may be a modified Viterbi algorithm, for example. The frame process may comprise verifying a CRC for the bit-sequence and/or decrypting the bit-sequence. In some instances, estimates of the bit-sequence may be fed back to the decoding algorithm. A speech stream that satisfies speech constraints may be generated based on the generated bit-sequence. The speech constraints may comprise gain and/or pitch continuity, for example. The generated speech stream may be decoded via a voice decoder that supports full rate (FR), adaptive multi-rate (AMR), and/or enhanced full-rate (EFR) speech coding. Frame process results may be fed back to the burst process to improve decoding of received GSM speech data.

Abstract: Each of N audio signals are filtered with a unique decorrelating filter (38) characteristic, the characteristic being a causal linear time-invariant characteristic in the time domain or the equivalent thereof in the frequency domain, and, for each decorrelating filter characteristic, combining (40, 44, 46), in a time and frequency varying manner, its input (Zi) and output (Z-i) signals to provide a set of N processed signals (X i). The set of decorrelation filter characteristics are designed so that all of the input and output signals are approximately mutually decorrelated. The set of N audio signals may be synthesized from M audio signals by upmixing (36), where M is one or more and N is greater than M.

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: An audio coding apparatus includes an audio coding unit 101 for, when it has received an audio signal input, performing coding to the audio signal input and outputting coded audio data and coding related data which is data relating to the coding, as specific data, an ancillary audio coding unit 102 for, when it has received an ancillary audio signal, performing coding to the ancillary audio signal and outputting coded ancillary audio data, and an auxiliary data output unit 103 for producing, from the specific data and the coded auxiliary data, auxiliary data including both of the specific data and the coded ancillary audio data, and outputting the auxiliary data.

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: An apparatus calculates a number of spectral envelopes to be derived by a spectral band replication (SBR) encoder, wherein the SBR encoder is adapted to encode an audio signal using a plurality of sample values within a predetermined number of subsequent time portions in an SBR frame extending from an initial time to a final time, the predetermined number of subsequent time portions being arranged in a time sequence given by the audio signal. The apparatus has a decision value calculator for determining a decision value, the decision value measuring a deviation in spectral energy distributions of a pair of neighboring time portions. The apparatus further has a detector for detecting a violation of a threshold by the decision value and a processor for determining a first envelope border between the pair of neighboring time portions when the violation of the threshold is detected.

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: There is provided a method of encoding audio and including said encoded audio into a digital transport stream, comprising receiving at an encoder input a plurality of temporally co-located audio signals, assigning identical time stamps per unit time to all of the plurality of temporally co-located audio signals and incorporating the identically time stamped audio signals into the digital transport stream. There is also provided a method decoding said encoded data, and encoding apparatus and decoding apparatus.

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: An audio frame format includes a channel field indicating a number of audio multi-channels, an ignore bit indicating whether or not an audio sample is present in a predetermined region of a packet format, and an A channel audio sample field for transmitting the audio sample. Further, the audio frame format includes a B channel audio sample field for transmitting the audio sample, and a payload of the packet that includes a repetition of an audio frame.

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 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: An oral care implement (100), comprising an oral care region (124,127) attached to a body (125), a portion (126) of the body being configured for gripping by a user; a memory configured to store a plurality of audio signals from an external signal source; at least one measurement component configured to measure a parameter of use of the oral care region (124,128); and a processor configured to change output of a first audio signal of the plurality of audio signals to a second audio signal of the plurality of audio signals based on the measured parameter from. the at least one measurement component.

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: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: 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.