Abstract: A method of concealing bit errors in a signal is provided. The method includes decoding an encoded signal parameter based upon constraints placed on a signal parameter, comparing the decoded signal parameter against the constraints, and declaring the decoded signal parameter invalid when the constraints are violated.

Abstract: Facts are extracted from speech and recorded in a document using codings. Each coding represents an extracted fact and includes a code and a datum. The code may represent a type of the extracted fact and the datum may represent a value of the extracted fact. The datum in a coding is rendered based on a specified feature of the coding. For example, the datum may be rendered as boldface text to indicate that the coding has been designated as an “allergy.” In this way, the specified feature of the coding (e.g., “allergy”-ness) is used to modify the manner in which the datum is rendered. A user inspects the rendering and provides, based on the rendering, an indication of whether the coding was accurately designated as having the specified feature. A record of the user's indication may be stored, such as within the coding itself.

Abstract: An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mask ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a three-dimensional (3D) down-mix signal from an input bitstream, generating a down-mix signal with 3D effects removed therefrom by performing a 3D rendering operation on the extracted 3D down-mix signal, and generating a 3D down-mix signal with 3D effects by performing a 3D rendering operation on the generated down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: The invention relates to a method for quantifying components, wherein certain components are each determined based on a plurality of audio signals and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function to be applied to the components by testing a condition relative to an audio signal and depending on a comparison made between a psycho-acoustic masking threshold relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Abstract: A system and method for providing a scalable spoken dialog system are disclosed. The method comprises receiving information which may be internal to the system or external to the system and dynamically modifying at least one module within a spoken dialog system according to the received information. The modules may be one or more of an automatic speech recognition, natural language understanding, dialog management and text-to-speech module or engine. Dynamically modifying the module may improve hardware performance or improve a specific caller's speech processing accuracy, for example. The modification of the modules or hardware may also be based on an application or a task, or based on a current portion of a dialog.

Abstract: Disclosed herein is a quantization method and apparatus of an audio encoder. The quantization method comprises calculating an absolute value of a maximum frequency spectrum of a first frame, externally received, by analyzing frequency spectrum data of the first frame, setting an initial value of a common scale factor to be used to quantize the first frame based on the absolute value of the maximum frequency spectrum of the first frame and an absolute value of a maximum frequency spectrum of a second frame, which has previously been calculated, and quantizing the frequency spectrum data of the first frame based on the set initial value of the common scale factor. Accordingly, before quantization is performed, an initial value of a common scale factor which is almost close to a value of an actual common scale factor can be previously set.

Abstract: An advanced audio coding (AAC) encoder quantization architecture is described. The architecture includes an efficient, low computation complexity approach for estimating scalefactors in which a base scalefactor estimate is adjusted by a delta scalefactor estimate that is based, in part, on global scalefactor adjustments applied to the previously quantized/encoded frame. Using such feedback, the AAC encoder quantization architecture is able to produce scalefactor estimates that are very close to the actual scalefactor applied by the subsequent quantization and encoding process. The architecture further includes a frequency hole avoidance approach that reduces a magnitude of an estimated scalefactor to avoid generating frequency holes in quantized SFBs.

Abstract: A method and an apparatus to encode and decode a speech signal using a code excited linear prediction (CELP) algorithm. In order to reduce a bit rate without degrading performance in an enhancement layer based on CELP, each of a fixed codebook of a core layer and a fixed codebook of the enhancement layer is divided into a plurality of spaces. The spaces of the fixed codebook of the enhancement layer excludes a space corresponding to a least distorted space determined from among the spaces of the fixed codebook of the core layer are searched.

Abstract: A vocoder and method transcodes Mixed Excitation Linear Prediction (MELP) encoded data for use at different speech frame rates. Input data is converted into MELP parameters such as used by a first MELP vocoder. These parameters are buffered and a time interpolation is performed on the parameters with quantization to predict spaced points. An encoding function is performed on the interpolated data as a block to produce a reduction in bit-rate as used by a second MELP vocoder at a different speech frame rate than the first MELP vocoder.

Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

Abstract: Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.

Abstract: The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S?MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S?MIC).

Abstract: A method and apparatus to search a codebook including pulses that model a predetermined component of a speech signal. The method includes the operations of selecting a predetermined number of paths corresponding to a predetermined number of pulse locations that are most consistent with the predetermined component, from among paths corresponding to pulse locations of a predetermined pulse location set allocated to at least one branch that connects one state of a predetermined Trellis structure to another state, performing the path selecting operation on each of states other than the one state, and selecting a path corresponding to pulse locations that are most consistent with the predetermined component from among paths including the selected paths, wherein each path corresponds to a union of plural tracks of an algebraic codebook. Accordingly, a number of calculations required during a codebook search is reduced.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: A method and coding device are provided for coding a digital audio input signal in a hierarchical coder, which includes a core coding stage with B bits and at least one current improvement coding stage k, delivering quantization indices which are concatenated to form the indices of a preceding embedded coder. The method includes: obtaining possible quantization values for the current improvement stage k by determining absolute reconstruction levels of just the current stage k on the basis of the indices of the preceding embedded coder; and quantizing the input signal of the hierarchical coder having undergone or not a perceptual weighting processing, on the basis of the possible quantization values so as to form a scalar quantization index for the stage k and a quantized signal corresponding to one of the possible quantization values.

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 voice encoding device accurately encodes a spectrum shape of a signal having a strong tonality such as a vowel. The device includes: a sub-band divider which divides a first layer error conversion coefficient to be encoded into M sub-bands so as to generate M sub-band conversion coefficients; a shape vector encoder which performs encoding on each of the M sub-band conversion coefficients so as to obtain M shape encoded information and calculates a target gain of each of the M sub-band conversion coefficients; a gain vector former which forms one gain vector by using M target gains; a gain vector encoder which encodes the gain vector so as to obtain gain encoded information; and a multiplexer which multiplexes the shape encoded information with the gain encoded information.

Abstract: An audio decoding device can adjust the high-range emphasis degree in accordance with a background noise level. The audio decoding device includes: a sound source signal decoder which performs a decoding process by using sound source encoding data separated by a separator so as to obtain a sound source signal; an LPC synthesis filter which performs an LPC synthesis filtering process by using a sound source signal and an LPC generated by an LPC decoder so as to obtain a decoded sound signal; a mode judger which determines whether a decoded sound signal is a stationary noise period by using a decoded LSP inputted from the LPC decoder a power calculator which calculates the power of the decoded audio signal; an SNR calculator which calculates an SNR of the decoded audio signal by using the power of the decoded audio signal and a mode judgment result in the mode judger and a post filter which performs a post filtering process by using the SNR of the decoded audio signal.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19 or 20 unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality.

Abstract: Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band.

Abstract: The invention relates to the coding/decoding of a signal into several sub-bands, in which at least a first and a second sub-bands which are adjacent are transform coded (601, 602). In particular, in order to apply a perceptual weighting, in the transformed domain, to at least the second sub-band, the method comprises:—determining at least one frequency masking threshold (606) to be applied on the second sub-band; and normalizing said masking threshold in order to provide a spectral continuity between the above-mentioned first and second sub-bands. An advantageous application of the invention involves a perceptual weighting of the high-frequency band in the TDAC transform coding of a hierarchical encoder according to standard G.729.1.

Abstract: A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and an auto-regressive predictor produces a predicted parameter by multiplying a predictive coefficient with a past decoded parameter. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue.

Abstract: A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.

Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19, or 20 unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: For an enhanced sequential compression of data vectors in a respective compression pass, a current data vector is mapped to at least one current code vector of at least one codebook in at least one quantization stage. The at least one codebook is reordered taking account of at least one intermediate result from the current compression pass and at least one intermediate result from a preceding compression pass. At least one codebook index that is associated in the at least one reordered codebook to the at least one current code vector is then provided for further use. For a decompression of compressed data vectors represented by such codebook indices, at least one codebook index is mapped to at least one code vector of at least one equally reordered codebook.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19 or 20 of unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: An encoding apparatus and a decoding apparatus for reducing the quantization error of a G.711 codec and improving sound quality are provided. The encoding apparatus includes a G.711 encoder which generates a G.711 bitstream by encoding an input audio signal; an enhancement-layer encoder which chooses one of a static bit allocation method and a dynamic bit allocation method that can produce less quantization error based on the input audio signal and the G.711 bitstream, and outputs an enhancement-layer bitstream including encoded additional mantissa information obtained by using the chosen bit allocation method; and a multiplexer which multiplexes the G.711 bitstream and the enhancement-layer bitstream. Therefore, it is possible to reduce the quantization error of a G.711 codec and improve sound quality.

Abstract: The APPARATUSES, METHODS AND SYSTEMS FOR SPARSE SINUSOIDAL AUDIO PROCESSING AND TRANSMISSION (hereinafter “SS-Audio”) provides a platform for encoding and decoding audio signals based on a sparse sinusoidal structure. In one embodiment, the SS-Audio encoder may encode received audio inputs based on its sparse representation in the frequency domain and transmit the encoded and quantized bit streams. In one embodiment, the SS-Audio decoder may decode received quantized bit streams based on sparse reconstruction and recover the original audio input by reconstructing the sinusoidal parameters in the frequency domain.

Abstract: A communications network is used to transfer user attribute information about participants in a communication session to their respective communication terminals for storage and use thereon to configure a speech codec to operate in a speaker-dependent manner, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals.

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: Provided are a quantization device and quantization method which reduce coding distortion with a small degree of calculation, and achieve adequate coding performance thereby. A multistage vector quantization unit (102) treats a number of candidates N which are designated prior to operation in the first-stage vector quantization unit (201-1), decrements the number of candidates by one beginning with the second-stage vector quantization unit (201-2-201-J) and continuing with each stage thereafter, and, if the number of candidates is three or less, assesses the quantization distortion at each such stage, treating the number of candidates at the following stage as a predetermined value P if the quantization distortion is greater than a prescribed threshold, and treating the number of candidates at the following stage as a value Q that is less than the predetermined value P if the quantization distortion is less than or equal to the predetermined threshold.

Abstract: A noise feedback coding (NFC) system and method that utilizes a simple and relatively inexpensive general structural configuration, but achieves improved flexibility with respect to controlling the shape of coding noise. The NFC system and method utilizes an all-zero noise feedback filter that is configured to approximate the response of a pole-zero noise feedback filter.

Abstract: A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.

Abstract: Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo.

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 parameter decoding device performs a parameter compensation process so as to suppress degradation of a main observation quality in a prediction quantization. The parameter decoding device includes first amplifiers which multiply inputted quantization prediction residual vectors by a weighting coefficient. A further amplifier multiplies the preceding frame decoding LSF vector yn?1 by the weighting coefficient. An additional amplifier multiplies the code vector xn+1 outputted from a codebook by the weighting coefficient ?0. An adder calculates the total of the vectors outputted from the amplifiers, the further amplifier, and the additional amplifier. A selector switch selects the vector outputted from the adder if the frame erasure coding Bn of the current frame indicates that ‘the n-th frame is an erased frame’ and the frame erasure coding Bn+1 of the next frame indicates that ‘the n+1-th frame is a normal frame’.

Abstract: The invention discloses a multi-stage quantization method, which includes the following steps: obtaining a reference codebook according to a previous stage codebook; obtaining a current stage codebook according to the reference codebook and a scaling factor; and quantizing an input vector by using the current stage codebook. The invention also discloses a multi-stage quantization device. With the invention, the current stage codebook may be obtained according to the previous stage codebook, by using the correlation between the current stage codebook and the previous stage codebook. As a result, it does not require an independent codebook space for the current stage codebook, which saves the storage space and improves the resource usage efficiency.

Abstract: A method, system and program for encoding speech.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving downmix information, object information and mix information, generating and transferring multi-channel information using at least one of the downmix information, the object information and the mix information, and selectively generating and transferring either first gain information or extra multi-channel information including second gain information in accordance with a decoding mode using at least one of the object information and the mix information.

Abstract: An audio encoder encodes side information into a compressed audio bitstream containing encoding parameters used by the encoder for one or more encoding techniques, such as a noise-mask-ratio curve used for rate control. A transcoder uses the encoder generated side information to transcode the audio from the original compressed bitstream having an initial bit-rate into a second bitstream having a new bit-rate. Because the side information is derived from the original audio, the transcoder is able to better maintain audio quality of the transcoding. The side information also allows the transcoder to re-encode from an intermediate decoding/encoding stage for faster and lower complexity transcoding.

Abstract: An iterative rate-distortion optimization algorithm for MPEG I/II Layer-3 (MP3) encoding based on the method of Lagrangian multipliers. Generally, an iterative method is performed such that a global quantization step size is determined while scale factors are fixed, and thereafter the scale factors are determined while the global quantization step size is fixed. This is repeated until a calculated rate-distortion cost is within a predetermined threshold. The methods are demonstrated to be computationally efficient and the resulting bit stream is fully standard compatible.

Abstract: In an encoding process, a CPU transforms an audio signal from the real-time domain to the frequency domain, and transforms the signal into spectra consisting of MDCT coefficients. The CPU separates the audio signal into several frequency bands, and performs bit shifting in each band such that the MDCT coefficients can be expressed with pre-configured numbers of bits. The CPU re-quantizes the MDCT coefficients at a precision differing for each band, and transmits the values acquired thereby and shift bit numbers as encoded data. Meanwhile, in a decoding process, a CPU receives encoded data and inverse re-quantizes and inverse bit shifts the data, thereby restoring the MDCT coefficients. Furthermore, the CPU transforms the data from frequency domain to the real-time domain by using the inverse MDCT, and restores and outputs the audio signal.

Abstract: In one aspect, a method of processing a voice signal to extract information to facilitate training a speech synthesis model is provided. The method comprises acts of detecting a plurality of candidate features in the voice signal, performing at least one comparison between one or more combinations of the plurality of candidate features and the voice signal, and selecting a set of features from the plurality of candidate features based, at least in part, on the at least one comparison. In another aspect, the method is performed by executing a program encoded on a computer readable medium. In another aspect, a speech synthesis model is provided by, at least in part, performing the method.

Abstract: Systems, methods and apparatuses are described herein for distributing user attribute information about users of a communications system to communication terminals, which use the user attribute information to configure a speech codec to operate in a speaker-dependent manner during a communication session, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals.

Abstract: A vector quantizer which improves the accuracy of vector quantization in switching over a vector quantization codebook on a first stage depending on the type of feature having the correlation with a quantization target vector. In the vector quantizer, a classifier generates classification information representing a type of narrowband LSP vector having the correlation with wideband LSP (Line Spectral Pairs) of the plural types. A first codebook selects one sub-codebook corresponding to the classification information as a codebook used for the quantization of the first stage from plural sub-codebooks corresponding to each of the types of narrowband LSP vectors. A multiplier multiplies the quantization residual vector of the first stage inputted from an adder by a scaling factor corresponding to the classification information of plural scaling factors stored in a scaling factor determiner and outputs it to an adder as the quantization target of a second stage.

Abstract: At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction.

Abstract: In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding.