Abstract: A representation of an audio signal having a first, a second and a third frame is derived by estimating first warp information for the first and second frames and second warp information for the second and third frames, the warp information describing pitch information of the audio signal. First or second spectral coefficients for first and second frames or second and third frames are derived using first or second warp information and a first or second weighted representation of the first and second frames or second and third frames, the first or second weighted representation derived by applying a first or second window function to the first and second frames or second and third frames, wherein the first or second window function depends on the first or second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients.

Abstract: Method, apparatus, and system for encoding and decoding signals are disclosed. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag is a first flag; or obtaining a second-domain predictive signal according to the LP processing result when the long-term flag is a second flag; converting the second-domain predictive signal into a first-domain predictive signal, calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract: The present disclosure includes processing a signal to generate a first sub-set of data, transmitting the first sub-set of data for generation of a reconstructed audio signal, the reconstructed audio signal having a fidelity relative to the signal, processing the signal to generate a second sub-set of data and a third sub-set of data, the second sub-set of data defining a second portion of the signal and comprising data that is different than data of the first sub-set of data, and the third sub-set of data defining a third portion of the signal and comprising data that is different than data of the first and second sub-sets of data, comparing a priority of the second sub-set of data to a priority of the third sub-set of data, and transmitting one of the second sub-set of data and the third sub-set of data over the network for improving the fidelity.

Abstract: An audio encoding method and apparatus and an audio decoding method and apparatus are provided. The audio signal decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating a modified downmix signal based on the downmix signal and extracted information which is extracted from the object-based side information; generating channel-based side information based on the object-based side information and control data for rendering the downmix signal; and generating a multi-channel audio signal based on the modified downmix signal and the channel-based side information.

Abstract: Disclosed is a stereo acoustic signal encoding apparatus in which the signal quality does not deteriorate if there are a plurality of sound sources. A peak tracing unit (401) splits frames of a right channel signal and a left channel signal into a plurality of sub frames; detects the peaks of wave shapes of the split sub frames; and estimates a frame delay time D for each frame of the right channel signal and the left channel signal by comparing the positions of the detected peaks. A time adjusting unit (402) adjusts the time of the right channel signal on the basis of the frame time delay D. A down-mix operation is carried out using the right channel signal which has been subjected to the time adjustment and the left channel signal to generate a mono signal and a sub signal. A mono signal encoding unit (403) encodes the mono signal. A sub signal encoding unit (404) encodes the sub signal. The time delay encoding unit (405) encodes the frame time delay D.

Abstract: A method of encoding a time-domain audio signal is presented. A device transforms the time-domain signal into a frequency-domain signal including a sequence of sample blocks, wherein each block includes a coefficient for each of multiple frequencies. The coefficients of each block are grouped into frequency bands. For each frequency band of each block, a scale factor is estimated for the band, and the energy of the band for the block is compared with the energy of the band of an adjacent sample block, wherein the blocks may be adjacent to each other in either or both of an interchannel and a temporal sense. If the ratio of the band energy for the first block to the band energy for the adjacent block is less than some value, the scale factor of the band for the first block is increased. The coefficients of the band for each block are quantized based on the resulting scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.

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 embodiments of the present invention relate to a compression coding and decoding method, a coder, a decoder and a coding device. The compression coding method includes: extracting sign information of an input signal to obtain an absolute value signal of the input signal; obtaining a residual signal of the absolute value signal by using a prediction coefficient, where the prediction coefficient is obtained by prediction and analysis that are performed according to a signal characteristic of the absolute value signal of the input signal; and multiplexing the residual signal, the sign information and a coding parameter to output a coding code stream, after the residual signal, the sign information and the coding parameter are respectively coded, so as to improve compression efficiency of a voice and audio signal.

Abstract: A method for detecting a transient in an audio signal that has been broken up into frames includes obtaining a time domain feature of the frames and comparing the domain feature with a predetermined value. If the time domain feature is greater than the predetermined value, the frames are taken as transient and if the time domain feature is less than the predetermined value, the frames are taken as non-transient. The method has a low computational intensity and is thus very suitable for devices with limited processing resources.

Abstract: A method and an apparatus for generating a lattice vector quantizer codebook are disclosed. The method includes: storing an eigenvector set that includes amplitude vectors and/or length vectors, where the amplitude vectors and/or length vectors are different from each other and correspond to a root leader of a lattice vector quantizer; storing storage addresses of the amplitude vectors and length vectors, where the amplitude vectors and length vectors correspond to the root leader and are in the eigenvector set; and generating a lattice vector quantizer codebook according to the eigenvector set and the storage addresses.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: An input frame data producing unit produces from data stored in an input buffer input frames each including a predetermined number of sub-frames of a first hopsize determined based on the first frame size and the overlapping rate. A frame processing unit executes a window function on the input frames and shifts the windowed input frames by the first hopsize and overlaps the shifted input frames, storing the overlapped frames in an output frame. An output buffer data producing frame unit stores data from the output frame to an output buffer including a predetermined number of sub-frames of a second hopsize. A CPU sets the first hopsize and overlapping rate in a slow-speed reproduction when the reproducing speed ratio is set lower than 1 different from in a high-speed reproduction when the reproducing speed ratio is set larger than 1.

Abstract: Provided are parametric audio encoding and decoding apparatuses and methods thereof. In the parametric audio encoding method, an audio signal is segmented into a plurality of segments. At least one sine wave is extracted from each of the segments, and the extracted sine waves are connected. It is determined whether an extracted sine wave is a birth sine wave. If the extracted sine wave is a birth sine wave, a bit stream is generated by encoding the phase of the birth sine wave on the basis of the frequency of the birth sine wave, wherein the number of bits allocated to encode the phase of the birth sine wave is adjusted according to the frequency of the birth sine wave.

Abstract: Data compression technology (“the technology”) is disclosed that can employ two or more prediction models contemporaneously. The technology receives data from one or more sources; shifts or re-sample one of more corresponding signals; creates a prediction model of uncompressed samples using at least two different individual or composite models; selects a subset of the models for prediction of samples; determines an order in which signals will be compressed; formulates a combined predictions model using the selected subset of models; predicts a future value for the data using the combined compression model; defines a function that has as parameters at least the predicted future values for the data and actual values; selects a compression method for the values of the function; and compresses the data using at least the predicted value of the function.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. At least one frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, and (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes.

Abstract: A frame error concealment method is provided that includes predicting a parameter by performing a regression analysis on a group basis for a plurality of groups formed from a first plurality of bands forming an error frame and concealing an error in the error frame by using the parameter predicted on a group basis.

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: In recent years, it has become commonplace for portable devices to generate analog audio signals from numerous sources, meaning that the codecs employed in these portable devices need to be able to utilize various digital bit streams at different sampling rates. To date, however, the circuitry for asynchronous sampling rate conversions for multiple bit streams has been complex, rigid, and power hungry. Here, a codec is provided which uses miniDSP cores to perform asynchronous sampling rate conversion efficiently and with reduced power consumption compared to other conventional codecs.

Abstract: There is provided an audio encoding device capable of effectively encoding stereo audio in audio encoding having monaural-stereo scalable configuration. In this device, a correlation degree comparison unit (304) calculates correlation in a first channel (correlation degree between the past signal and the current signal in the first channel) from the first channel audio signal and calculates correlation in a second channel (correlation degree between the past signal and the current signal in the second channel) from the second channel audio signal. The correlation in the first channel is compared to the correlation in the second channel. A channel having the greater correlation is selected.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: There is provided an audio encoding device capable of effectively encoding a stereo audio even when a correlation between channels of the stereo audio is small. In the device, a monaural signal generation unit (110) generates a monaural signal by using a first channel signal and a second channel signal contained in the stereo signal. An encoding channel selection unit (120) selects one of the first channel signal and the second channel signal. An encoding unit including a monaural signal encoding unit (112), a first channel encoding unit (122), a second channel encoding unit (124), and a switching unit (126) encodes the generated monaural signal to obtain core-layer encoded data and encodes the selected channel signal to obtain extended layer encoded data corresponding to the core-layer encoded data.

Abstract: A technique of spectral noise shaping in an audio coding system is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. The tonality of each sub-band is determined. If a sub-band is tonal, time domain linear prediction (TDLP) processing is applied to the sub-band, yielding a residual signal and linear predictive coding (LPC) coefficients of an all-pole model representing the sub-band signal. The residual signal is further processed using a frequency domain linear prediction (FDLP) method. The FDLP parameters and LPC coefficients are transferred to a decoder. At the decoder, an inverse-FDLP process is applied to the encoded residual signal followed by an inverse TDLP process, which shapes the quantization noise according to the power spectral density of the original sub-band signal. Non-tonal sub-band signals bypass the TDLP process.

Abstract: The invention relates in general to a method for combining frequency domain encoded signals from at least two signal sources. To allow combining signals without decoding the signals entirely, the invention provides decoding the encoded signals obtaining quantized spectral components, inverse quantizing the quantized spectral component of the decoded signals obtaining window sequences, and combining the at least inverse quantized signals obtaining a combined signal.

Abstract: An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoder encodes an input signal and generates first encoded information, a first layer decoder decodes the first encoded information and generates a first decoded signal, a characteristic judger analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoder changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information.

Abstract: A representation of an audio signal having a first frame, a second frame following the first frame, and a third frame following the second frame, is derived by estimating first warp information for the first and the second frame and second warp information for the second frame and the third frame, the warp information describing a pitch information of the audio signal. First spectral coefficients for the first and the second frame are derived using the first warp information and a first weighted representation of the first and the second frame, the first weighted representation derived by applying a first window function to the first and the second frames, wherein the first window function depends on the first warp information.

Abstract: Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.

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: This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set.

Abstract: The signal processing is based on the concept of using a time-domain aliased (12, TDA) frame as a basis for time segmentation (14) and spectral analysis (16), performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall ?segmented? time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: Provided are a method and apparatus for encoding the frequency of a continuation sinusoidal signal and a method and apparatus for decoding the same. In the encoding method, a continuation sinusoidal signal successive to a sinusoidal signal in a previous section is extracted from a current section; a frequency of the continuation sinusoidal signal at the boundary between the current and previous sections is changed to a first frequency, based on representative frequencies of the continuation sinusoidal signal and at least one sinusoidal signal that belongs to a section adjacent to the current section and is successive to the continuation sinusoidal signal; and the first frequency is encoded.

Abstract: A frame is received that has the wideband audio signal. The low band audio signal is encoded to generate an encoded low band signal. The high band signal is analyzed to determine whether the high band signal is perceptually relevant to the low band signal. If the high band signal is not perceptually relevant to the low band signal, the low band signal is encoded and provided in a frame to the decoder without including parameters corresponding to characteristics of the high band signal. If the high band signal is perceptually relevant, the high band signal is encoded to generate an encoded high band signal. The resultant frame that is sent to the decoder will include a combination of the encoded low band signal and the encoded high band signal.

Abstract: A scalable audio codec encodes an input audio signal as a base layer at a high compression ratio and one or more residual signals as an enhancement layer of a compressed bitstream, which permits a lossless or near lossless reconstruction of the input audio signal at decoding. The scalable audio codec uses perceptual transform coding to encode the base layer. The residual is calculated in a transform domain, which includes a frequency and possibly also multi-channel transform of the input audio. For lossless reconstruction, the frequency and multi-channel transforms are reversible.

Abstract: The embodiments of a transcoding method, a transcoding device, and a communication apparatus are provided. The embodiment of a method includes: receiving a bit stream input from a sending end; determining an attribute of discontinuous transmission (DTX) used by a receiving end and a frame type of the input bit stream; and transcoding the input bit stream in a corresponding processing manner according to a determination result. Thereby, a corresponding transcoding operation is performed on the input bit stream according to the attribute of DTX used by the receiving end and the frame type of the input bit stream. In such a manner, input bit streams of various types can be processed, and the input bit streams can be correspondingly transcoded according to the requirements of the receiving end. Therefore, the average computational complexity and peak computational complexity can be effectively decreased without decreasing the quality of the synthesized speech.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

Abstract: Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.

Abstract: A method for perceptual spectral decoding comprises decoding of spectral coefficients recovered from a binary flux into decoded spectral coefficients of an initial set of spectral coefficients. The initial set of spectral coefficients are spectrum filled. The spectrum filling comprises noise filling of spectral holes by setting spectral coefficients in the initial set of spectral coefficients not being decoded from the binary flux equal to elements derived from the decoded spectral coefficients. The set of reconstructed spectral coefficients of a frequency domain formed by the spectrum filling is converted into an audio signal of a time domain. A perceptual spectral decoder comprises a noise filler, operating according to the method for perceptual spectral decoding.

Abstract: A signal recording and reproducing apparatus includes an encoder encoding an input signal to produce a first group of encoded data, and a second group of encoded data used for reproducing a signal of higher quality than a signal resulting from decoding of the first group of encoded data, a recording unit recording record-data, including the first group and the second group of encoded data, into a recording medium, a reproducing unit reproducing the record-data from the recording medium, a decoder decoding at least the first group of encoded data out of the record-data from the reproducing unit, and a controller controlling an operation of each part of the recording and reproducing apparatus, and the controller performs control so as to cause the recording unit to erase the second group of encoded data according to a command to increase the amount of free storage capacity of the recording medium.

Abstract: An apparatus for performing improved audio processing may include a processor. The processor may be configured to divide respective signals of each channel of a multi-channel audio input signal into one or more spectral bands corresponding to respective analysis frames, select a leading channel from among channels of the multi-channel audio input signal for at least one spectral band, determine a time shift value for at least one spectral band of at least one channel, and time align the channels based at least in part on the time shift value.

Abstract: A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi-channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down-mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102).

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: An apparatus and method for encoding and decoding using mutual information between a high band signal and a low band signal to increase a coding efficiency in a portable terminal are provided. The apparatus includes a bandwidth extender for extracting auxiliary information relating to a characteristic of a high band signal using the high band signal and a low band signal and an encoder for encoding residual high band signal obtained by subtracting auxiliary information acquired from the low band signal from auxiliary information acquired from the high band signal.

Abstract: A method, a device, and a system for transcoding between two embedded codecs are disclosed. The method includes: delaying a first encoded stream in input streams for integer number of frames, where the first encoded stream includes a stream of at least one extension layer in the input streams obtained after input signals are encoded by using a first codec; and using the first codec to decode other encoded streams in the input streams to obtain the first decoded signal; and performing delay aligning and adjusting to obtain an adjusted signal so as to reduce the transcoding complexity and enhance quality of the transcoded signals.

Abstract: An image browsing apparatus has: a display unit for displaying image data; a reproducing unit for reproducing audio data; a detector for detecting a feature of the audio data reproduced by the reproducing unit; and a controller for, when predetermined audio data is reproduced by the reproducing unit, controlling an updating interval of the image data displayed to the display unit on the basis of the feature of the predetermined audio data detected by the detector.

Abstract: Hierarchical coding of a source audio signal in the form of a data stream including a base level and at least two hierarchical enhancement levels, each of the levels being organized in successive frames. At least one frame of at least one enhancement level has a duration less than the duration of at least one frame of the base level. At least one indication representative of an order used for a set of enhancement level frames corresponding to the duration of at least one frame of the base level is inserted into the data stream.

Abstract: An encoding data processing apparatus generates a marked version of an audio signal. The marked copy is generated by embedding data representative of a payload data word into the audio signal. A code word generator that generates a water mark code word from the payload data word and reads data representing the water mark code word into a shuffle data store. A shuffle processor that generates pseudo randomly at least one address within an address space of the shuffle data store for each predetermined period and reads data representing the water mark code word out from the data store at locations identified by the randomly generated address. A data embedding processor that receives the audio signal and embeds the data representing the water mark code word read out from the shuffle data store into the audio signal for each predetermined period.