EP1495464B1 - Dispositif et procede pour coder un signal audio a temps discret et dispositif et procede pour decoder des donnees audio codees - Google Patents

Dispositif et procede pour coder un signal audio a temps discret et dispositif et procede pour decoder des donnees audio codees Download PDF

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EP1495464B1
EP1495464B1 EP02792858A EP02792858A EP1495464B1 EP 1495464 B1 EP1495464 B1 EP 1495464B1 EP 02792858 A EP02792858 A EP 02792858A EP 02792858 A EP02792858 A EP 02792858A EP 1495464 B1 EP1495464 B1 EP 1495464B1
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block
integer
difference
spectral values
quantization
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EP1495464A1 (fr
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Ralf Geiger
Thomas Sporer
Karlheinz Brandenburg
Jürgen HERRE
Jürgen Koller
Joachim Deguara
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0212—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/0017—Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032—Quantisation or dequantisation of spectral components

Definitions

  • the present invention relates to audio coding / audio decoding and more particularly to scalable encoding / decoding algorithms with a psychoacoustic first Scaling layer and a second scaling layer, the additional audio data for lossless decoding includes.
  • Eg MPEG Layer3 (MP3) or MPEG AAC use transforms such as the so-called modified discrete cosine transformation (MDCT), to a block-wise frequency representation of an audio signal to obtain.
  • MP3 MPEG Layer3
  • MPEG AAC uses transforms such as the so-called modified discrete cosine transformation (MDCT), to a block-wise frequency representation of an audio signal to obtain.
  • MDCT modified discrete cosine transformation
  • Such an audio encoder usually receives a stream of time discrete audio samples. The stream of audio samples is windowed to one windowed block of, for example, 1024 or 2048 windowed To obtain audio samples.
  • window functions such as B. a sine window, etc.
  • the windowed discrete-time audio samples are then by means of a filter bank in a spectral representation implemented.
  • a Fourier transformation or for special reasons a variety of Fourier transform, such as. B. an FFT or, as it has been executed, an MDCT be used.
  • an MDCT be used.
  • the Block of audio spectral values at the output of the filter bank can then be further processed as needed.
  • Both The above-mentioned audio coders are followed by quantization the audio spectral values, the quantization levels typically be chosen so that by quantizing introduced quantization noise below the psychoacoustic Masking threshold is, d. H. "Masked away" becomes.
  • the quantization is a lossy one Encoding.
  • the quantized spectral values are subsequently added entropy-coded for example by means of a Huffman coding.
  • page information such as B. scale factors, etc. is from the entropy-coded quantized spectral values by means of a bitstream multiplexer a bitstream is formed, which is stored or can be transferred.
  • the bitstream is by means of a bit stream demultiplexer in coded quantized spectral values and split page information.
  • the entropy-coded quantized spectral values are first entropy-decoded, to obtain the quantized spectral values.
  • the quantized spectral values are then inversely quantized, to obtain decoded spectral values, the quantization noise but below the psychoacoustic level Masking threshold is and therefore inaudible will be.
  • These spectral values are then determined by means of a Synthesis filter bank converted into a temporal representation, to obtain time discrete decoded audio samples.
  • the synthesis filter bank must be a transformation algorithm used inverse transformation algorithm become.
  • the windows are undone.
  • Fig. 4a For example, 2048 become discrete-time audio samples taken and windowed by means 402.
  • the window that embodies device 402 has a window length of 2N samples and provides output a block of 2N windowed samples.
  • a Window overlap is by means of a Device 404, which for reasons of clarity shown separated from the device 402 in Fig. 4a , a second block of 2N windowed samples is formed.
  • the 2048 samples fed to the device 404 are not those immediately adjacent to the first window subsequent discrete-time audio samples, but involve the second half of the through the institution 402 windowed samples and additionally include only 1024 "new" samples.
  • the overlap is through a device 406 in Fig. 4a shown symbolically, the causes a degree of overlap of 50%.
  • Both the through means 402 output 2N windowed samples and the 2N output by means 404 windowed samples are then sent by means 408 and 410 subjected to the MDCT algorithm.
  • the device 408 provides N according to the known MDCT algorithm Spectral values for the first window while the device 410 also provides N spectral values, but for the second window, wherein between the first window and the second window has an overlap of 50%.
  • the N spectral values of the first window as shown in Figure 4b, means 412, which is an inverse modified discrete cosine transformation performs, fed.
  • These become a facility 414, which is also an inverse modified discrete Cosine transformation.
  • Both the decor 412 and device 414 each provide 2N Samples for the first window or 2N samples for the second window.
  • a sample value y 1 of the second half of the first window that is to say with an index N + k, is summed with a sample value y 2 from the first half of the second window, ie with an index k, so that on the output side, ie in the decoder, N decoded temporal samples result.
  • Window function with w (k) When implemented by means 402 or 404 Window function with w (k) is called, where the index k represents the time index, the condition must be satisfied that the window weight w (k) squared adds to the window weight w (N + k) squared together 1 where k is from 0 to N-1. If a sine window is used, whose window weightings of the first Half wave follow the sine function, so is this condition always fulfilled, since the square of the sine and the square of the Cosines for each angle together give the value 1.
  • a disadvantage of the window method described in FIG. 4a with subsequent MDCT function is the fact that the fenestration by multiplying a discrete-time Sample when thinking of a sine window, is achieved with a floating-point number, since the sine of a Angle between 0 and 180 degrees apart from the angle 90 Degree does not yield an integer. Even if integer discrete time Samples are windowed Windows so floating point numbers.
  • German Patent DE 197 42 201 C1 discloses.
  • an audio signal in its spectral Presentation converted and quantized to quantized To obtain spectral values.
  • the quantized spectral values are again inversely quantized, transferred to the time domain and compared to the original audio signal. Is the error, so the error between the original Audio signal and the quantized / inverse quantized Audio signal, above an error threshold, then the Quantizer feedback set finer, and the Comparison is performed again. The iteration is over, if the error threshold is fallen below.
  • the next to the time domain encoded residual signal also includes coded spectral values corresponding to the quantizer settings have been quantified at the time the termination of the iteration were present. It was noted that the quantizer used is not must be controlled by a psychoacoustic model, so that the coded spectral values are typically more accurate are quantized as this is due to the psychoacoustic Model would have to be.
  • the first step is a conversion of a two's complement format in a sign magnitude format.
  • the second step is the conversion of a vertical magnitude sequence into a horizontal bit sequence in a processing block.
  • the lossless data conversion is executed to maximize the number of zeros or the number of consecutive zeros in a sequence to maximize compression as much as possible the temporal error signal due to digital Numbers to reach.
  • This principle is based on a bit-slice arithmetic coding (BSAC) scheme, in the technical publication "Multi-Layer Bit Sliced Bit Rate Scalable Audio Coder ", 103rd AES Convention, Preprint No. 4520, 1997.
  • the encoder must therefore in addition to its inherent encoder functionality also contain the complete decoder functionality. If the encoder is implemented by software, so For this purpose, both storage capacities and processor capacities are used needed to an encoder implementation leads with increased effort.
  • the object of the present invention is a less elaborate concept by which an audio stream can be generated, which is at least almost lossless is decodable.
  • a device for coding a Discrete-time audio signal according to claim 1 by a method of encoding a discrete-time audio signal according to claim 21, by a device for decoding of coded audio data according to claim 22, by a method for decoding encoded audio data Claim 31 or by a computer program according to claim 32 or 33 solved.
  • the present invention is based on the knowledge that the additional audio data, a lossless decoding allow the audio signal to be obtained thereby that a block of quantized spectral values as usual is provided, and then inversely quantized, to have inverse quantized spectral values due to quantization by means of a psychoacoustic model are lossy. These inverse quantized spectral values are then rounded to a rounding block of rounded to obtain inverse quantized spectral values.
  • a Integer transformation algorithm uses which a block of integer discrete-time samples Integer block of spectral values, which is only integer Spectral values generated.
  • the combination block which comprises the difference spectral values due to the integer transform algorithm and of the rounded quantization values are only integer ones Values that entropy-encode in some known manner can be. It should be noted that the Entropy coding of the combination block of any entropy coder can be used, such. Huffman coder or arithmetic coders etc.
  • the quantized spectral values of the quantization block can also be used any encoder be such.
  • any encoder be such.
  • inventive coding / decoding concept compatible with modern coding tools, such as Window switching, TNS or center / page encoding for multi-channel audio signals.
  • Invention is used to provide a quantization block of quantized using a psychoacoustic model Spectral values used an MDCT. Furthermore it is preferred as an integer transform algorithm to use a so-called IntMDCT.
  • Invention can be dispensed with the usual MDCT, and it can use the IntMDCT as an approach to the MDCT to the effect that the integer spectrum, that is obtained by the integer transform algorithm is fed to a psychoacoustic quantizer, to obtain quantized IntMDCT spectral values, then again inversely quantized and rounded to coincide with the original integer spectral values.
  • IntMDCT which is integer discrete-time samples integer spectral values generated.
  • processors work with integers, or Each floating-point number can be represented as an integer.
  • integer arithmetic in a processor is, so can on the rounds of the inverse quantized Spectral values are waived because of arithmetic of the processor anyway rounded values, namely within the accuracy of the LSB, d. H. the least significant bit, available.
  • d. H. a processing within the accuracy of the processor system used.
  • a rounding to a coarser accuracy be performed, in that the difference signal in the combination block on the by a rounding function specified accuracy is rounded.
  • flexibility allows the "degree" to influence the losslessness of the coding in the Meaning of a data compression, a nearly lossless encoder to accomplish.
  • the decoder according to the invention is characterized that from the audio data both the psychoacoustically coded Audio data as well as the additional audio data are extracted, subject to possibly existing entropy decoding and then processed as follows. First the quantization block in the decoder becomes inverse quantized and using the same rounding function, which has also been used in the encoder, rounded, then added to the entropy-decoded auxiliary audio data to become.
  • the decoder then has both a psychoacoustically compressed spectral representation of the audio signal as well as a lossless representation of the audio signal before, the psychoacoustically compressed spectral Representation of the audio signal in the time domain implement is a lossy coded / decoded To receive audio signal while the lossless presentation using an integer transform algorithm inverse integer transformation algorithm implemented in the time domain is going to be a lossless or, as it has been stated, to obtain almost lossless coded / decoded audio signal.
  • the inventive encoder shown in Fig. 5 includes an input 50 into which a discrete-time Audio signal can be fed, and an output 52, off the encoded audio data can be output. This at the entrance 50 fed discrete-time audio signal is in a device 52 is fed to provide a quantization block, the output side a quantization block of provides discrete-time audio signal using of a psychoacoustic model 54 quantized spectral values the discrete-time audio signal 50 has.
  • the inventive Encoder also includes means for Generating an integer block using an integer transform algorithm 56, where the integer algorithm is effective to discrete integer discrete Samples to generate integer spectral values.
  • the encoder according to the invention further comprises a device 58 for inverse quantization of the quantization block, issued by the device 52, and, if a different accuracy than the processor accuracy is required, a rounding function. If to the accuracy of the processor system as it has been executed is to be gone, so the rounding function is already inherent in the inverse quantization of the quantization block included as a processor that uses integer arithmetic has, anyway, is unable, non-integer Deliver values.
  • the device 58 provides thus a so-called rounding block, which inverse quantized Includes spectral values that are integer, d. H. inherent or have been rounded explicitly.
  • Both the rounding block as well as the integer block become a combination device fed using a Difference formation a difference block with difference spectral values returns, where the term "difference block" should indicate that the difference spectral values Values are the differences between the Include integer block and the rounding block.
  • Both the quantization block resulting from the device 52 is output, as well as the difference block, from the Difference generator 58 is output, one Processing device 60 is supplied, the z. Legs usual processing of the quantization block, and the further z. B. an entropy coding of the difference block causes.
  • the device 60 for processing indicates the output 52 coded audio data containing both information about the quantization block as well as information over the difference block.
  • the discrete-time audio signal by means of an MDCT converted into its spectral representation and then quantized.
  • the device 52 for delivering the Quantization block thus consists of the MDCT device 52a and a quantizer 52b.
  • integer block an IntMDCT 56 is preferred to use as an integer transformation algorithm to create.
  • FIG. 6 also shows the processing device shown in FIG 60 as bit stream encoder 60a for bit stream encoding of the quantization block that passes through the Means 52b, and by an entropy coder 60b for entropy coding the difference block shown.
  • the bit stream encoder 60a outputs the psychoacoustically coded audio data while the entropy coder 60b outputs an entropy-coded difference block.
  • the two output data of blocks 60a and 60b can be suitably combined into a bit stream, as the first scaling layer the psychoacoustically coded audio data, and the second scaling layer the additional audio data for lossless decoding Has.
  • the scaled bitstream then equals the in Fig. 5 shown coded audio data at the output 52 of the encoder.
  • the MDCT block 52a of FIG. 6 is omitted, as is indicated by a dashed arrow 62 in Fig. 5 is.
  • the integer spectrum that passes through the integer transformation means 56 is supplied, both fed to the difference generator 58 and in the quantizer 52b of FIG. 6.
  • the spectral values, which are generated by the integer transformation, become here as an approximation for a usual MDCT spectrum used.
  • This embodiment has the advantage that only the IntMDCT algorithm in the encoder is present, and that not both the IntMDCT algorithm as well as the MDCT algorithm present in the encoder have to be.
  • Fig. 7 shows a schematic block diagram of an inventive Decoder for decoding at the output 52 of Fig. 5 output coded audio data. These will be first in psychoacoustically coded audio data on the one hand and the supplemental audio data, on the other hand, decomposed.
  • the psychoacoustic encoded audio data becomes a conventional bit stream decoder 70 while the additional audio data, if they have been entropy-coded in the coder, entropy-decoded by means of an entropy decoder 72 become.
  • quantized spectral values that are inverse Quantizers 74 are supplied, which are identical in principle to the inverse quantizer in the device of FIG. 6 can be constructed.
  • a rounding device 76 is also provided, which the same rounding algorithm or the same rounding function to map a real number to an integer, as also implemented in the device 58 of FIG can be.
  • a decoder-side combiner 78 become the rounded inverse quantized spectral values with the entropy-coded additional audio data spectral valuewise preferably combined in an additive manner, so that in the decoder on the one hand inversely quantized spectral values at the output of the Device 74 are present and secondly integer spectral values present at the output of the combiner 78.
  • the output spectral values of the device 74 may then by means 80 for performing a inverse modified discrete cosine transformation in the time range are implemented to be a lossy one psychoacoustically coded and decoded audio signal to obtain.
  • a means 82 for performing an inverse integer MDCT also becomes the output signal of the combiner 78 in its temporal representation converted to a lossless coded / decoded audio signal or a, if a corresponding coarser rounding has been used, a nearly lossless coded and to generate again decoded audio signal.
  • a particular preferred embodiment of the entropy coder 60b of FIG. Having multiple in a conventional modern MPEG encoder Code tables that depend on an average Statistics of quantized spectral values selected are present, it is preferred to use the same code tables or codebooks also for the entropy coding of the difference block at the output of the combiner 58 to use.
  • the amount of the difference block, ie the residual IntMDCT spectrum on the accuracy of quantization may be a codebook selection for the entropy coder 60b performed without additional page information become.
  • the spectral coefficients are ie the quantized spectral values in the quantization block grouped into scale factor bands, where the Spectral values are weighted by a gain factor, that of a corresponding scale factor, which is a scale factor band is assigned, is derived. Because in this known coder concept an uneven quantizer is used to quantize the weighted spectral values, depends on the size of the residual values, ie the spectral values at the output of combiner 58, not just from the Scale factors, but also from the quantized values However, after both the scale factors and also the quantized spectral values in the bit stream, the is generated by the device 60a of Fig.
  • an audio encoder In an audio encoder according to the MPEG-2 AAC standard used a window switch to transpose Vorechos into transients Avoid audio signal areas. This technique is based on the possibility of individual window forms in each Half of the MDCT window, and allows you to to vary the block size in successive blocks.
  • the integer transform algorithm in the form of IntMDCT to the reference 1 to 3, with reference to FIGS. to likewise different window forms at Windows and the time-domain aliasing section of the MDCT decomposition to use. It is therefore preferred for both the integer transformation algorithm as well as for the Transformation algorithm for generating the quantization block to use the same window decisions.
  • TNS Temporal Noise Shaping
  • MS center / side
  • Prediction filter that is signal adaptive from a standard TNS module is calculated, is preferably also to used to predict the integer spectral values, where if this results in non-integer values, one Downstream rounding can be used to get back to generate integer values. This rounding is preferably done after each prediction step.
  • the original spectrum can be reconstructed again, by the inverse filter and the same rounding function be used.
  • the MS encoding also applied to IntMDCT spectral values be done by making rounded Givens rotations at an angle of ⁇ / 4, based on the lifting scheme. This allows the original IntMDCT values in the Decoders are reconstructed again.
  • inventive concept in its preferred form with the IntMDCT as integer Transformation algorithm on all MDCT-based audio-coded encoder.
  • coders are coders MPEG-4 AAC Scalable, MPEG-4 AAC Low Delay, MPEG-4 BSAC, MPEG-4 Twin VQ, Dolby AC-3 etc.
  • the inventive Concept is backwards compatible.
  • the hearse fitted Encoder or decoder is not changed, only extended. Additional information for the lossless components can be backward compatible in listening coded bitstream transmitted, for example, MPEG-2 AAC in the field "Ancillary Data".
  • MPEG-2 AAC in the field "Ancillary Data”.
  • the addition to the previous one Hearing-matched decoder, which is shown in dashed lines in Fig. 7 is, these additional data can evaluate and together with the quantized MDCT spectrum from the listener-matched decoder reconstruct the IntMDCT spectrum without loss.
  • scalable data streams have different scaling layers include, at least the lowest Scaling layer independent of the higher scaling layers can be transmitted and decoded. Further Scaling layers or enhancement layers are used in a Scalable processing of data from the first scaling layer or base layer added.
  • a fully equipped Encoder can produce a scaled data stream which has a first scaling layer and which in principle Any number of other scaling layers has.
  • An advantage of the scaling concept is that in In the case where a broadband transmission channel for Available is the scaled data stream generated by the encoder complete, including all scaling layers be transmitted over the broadband transmission channel can.
  • the coded signal can nevertheless transmitted over the transmission channel, but only in the form of the first scaling layer or a specific one Number of further scaling layers, where the certain number smaller than the whole of the encoder generated number of scaling layers.
  • the encoder already adapted to the channel, to which he is connected, the basic scaling layer or first scaling layer and one dependent on the channel Create number of further scaling layers.
  • the scalable concept also has the Advantage that it is backwards compatible. This means, a decoder that is only capable of the first scaling layer to process, just the second and more Scaling layers in the data stream are ignored and on can produce useful output signal. Is the decoder on the other hand, a typically more modern decoder that has multiple Scaling layers from the scaled data stream can process this encoder with the same data stream how a basic decoder is addressed.
  • the basic Scalability in that the quantization block, so the Output of the bitstream encoder 60a into a first scaling layer 81 of FIG. 8 which, when FIG. 6, psychoacoustically encoded data e.g. For includes a frame.
  • both scaling layers 81 and 82 are transmitted to the decoder.
  • the transmission channel is a narrow-band transmission channel, in the only the first scaling layer "fits", so just imagine the second scaling layer the transmission are removed from the data stream, so that a Decoder addressed only with the first scaling layer becomes.
  • the decoder side can be a "basic decoder", which only can process the psychoacoustically encoded data that simply omit the second scaling layer 82 if it does receive them over a broadband transmission channel Has.
  • the decoder is a full-featured decoder, Both a psychoacoustic decoding algorithm and an integer decoding algorithm, so this fully featured decoder can handle both the first scaling layer as well as the second scaling layer to decode to a lossless coded and to generate decoded output again.
  • Fig. 8a is again psychoacoustically in a first scaling layer encoded data for one frame.
  • the second scaling layer of Fig. 8a is now scaled finer, so that from this second scaling layer in FIG. 8a, several scaling layers arise, such as e.g. a (smaller) second scaling layer, a third scaling layer, a fourth scaling layer, etc.
  • Fig. 9 schematically illustrates binary coded spectral values Each line 90 in FIG. 9 represents a binary coded one Difference spectral value.
  • Fig. 9 are the difference spectral values sorted by the frequency as it passes through an arrow 91 is indicated. So has a difference spectral value 92 is a higher frequency than the difference spectral value 90.
  • the first column of the panel in FIG. 9 represents the most significant bit of a difference spectral value in front.
  • the second digit represents the bit with the significance MSB-1.
  • the third column represents a bit of significance MSB-2.
  • the third to last column is one bit valence LSB + 2.
  • the penultimate column is set Finally, the bit with the significance LSB + 1 represents last column one bit with the significance LSB, ie the least significant one Bit of a difference spectral value.
  • precision scaling is performed by taking, for example, the 16 most significant bits of a difference spectral value as the second scaling layer to then be entropy coded by the entropy coder 60b, if desired.
  • a decoder using the second scaling layer receives on the output side difference spectral values with an accuracy of 16 bits, so that the second scaling layer together with the first scaling layer delivers a losslessly decoded CD-quality audio signal. It is known that CD-quality audio samples have a width of 16 bits.
  • the encoder may further generate a third scale layer comprising the last eight bits of a difference spectral value and also entropy as needed is encoded (means 60 of Fig. 6).
  • a full-featured decoder that uses the data stream the first scaling layer, the second scaling layer (16 most significant bits of the difference spectral values) and the third scaling layer (8 least significant bits a difference spectral value), so the decoder using all three scale layers lossless coded / decoded audio signal in studio quality, So with one present at the output of the decoder Word width of a sample of 24 bits.
  • Word lengths of the samples are common than in the consumer sector.
  • the word width is 16 bits in an audio CD, while in the studio area 24 bit or 20 Bit be used.
  • the transmitted values are preferably scaled back to the original range, for example 24 bits, by multiplying them by 2 8, for example.
  • An inverse IntMDCT is then applied to the corresponding scaled-back values.
  • the redundancy in the LSBs has an audio signal, for example in the upper frequency range very little energy, so expresses this also applies to the IntMDCT spectrum in very small values, for example, much smaller than the z. B. at 8 bits are possible values (-128, ..., 127). This expresses in compressibility of the LSB values of the IntMDCT spectrum. It should also be noted that at very small difference spectral values typically a number of bits from MSB to MSB-n are equal to zero, and then that first with a bit with a weight MSB-n-1 the first, leading 1 in a binary coded difference spectral value occurs. In such a case, if a difference spectral value only zeros in the second scaling layer includes an entropy coding is suitable for further Data compression especially good.
  • Invention is for the second scaling layer 82 of Fig. 8a prefers sample rate scalability.
  • a sample rate scalability is achieved by that in the second scaling layer, as shown in Fig. 9 right is, the difference spectral values up to a first Limit frequency are included, while in a third scaling layer the difference spectral values with one frequency between the first cutoff frequency and the maximum Frequency are included.
  • the invention includes the second scaling layer in FIG. 9 Differential spectral values up to a frequency of 24 kHz, which corresponds to a sampling rate of 48 kHz.
  • the third scaling layer then contains the difference spectral values from 24 kHz to 48 kHz, which corresponds to a sampling rate of 96 kHz.
  • the second scaling layer and the third scale layer not necessarily all bits of a difference spectral value must be coded. So could in one another form of combined scalability the second Scaling layer, the bits MSB to MSB-x of the difference spectral values up to a certain cutoff frequency.
  • a third scaling layer could then be the bits MSB to MSB-x of the difference spectral values from the first one Limit frequency up to the maximum frequency include.
  • a fourth scaling layer could then be the remaining bits for the difference spectral values up to the cutoff frequency.
  • the last scaling layer could then be the rest Bits of the difference spectral values for the upper frequencies include. This concept becomes a division of the panel in Fig. 9 in four quadrants, each Quadrant represents a scaling layer.
  • the frequency scalability is at a preferred Embodiment of the present invention a scalability between 48 kHz and 96 kHz sampling rate described.
  • the 96 kHz sample signal is in the IntMDCT range in the lossless extension layer first only up to half coded and transmitted. If the upper Part is not transmitted in addition, it is in the decoder assumed to be zero. In the case of the inverse IntMDCT (same Length as in the encoder) then creates a 96 kHz signal, the in the upper frequency range contains no energy and therefore be scanned to 48 kHz without loss of quality can.
  • the accuracy scaling can be similar in a sense be softened. So can the first scaling layer also spectral values with z. B. have more than 16 bits, where next scaling layer then has the difference. Generally speaking, the second scaling layer thus the difference spectral values with lower accuracy, while in the next scaling layer the rest, so the difference between the complete spectral values and the spectral values contained in the second scaling layer is transmitted. This is a variable accuracy reduction reached.
  • the inventive method for encoding or decoding is preferably on a digital storage medium such.
  • a digital storage medium such.
  • the control signals so with a programmable computer system can work together, that the coding and / or decoding method are carried out can / can.
  • that is a computer program product with on a machine readable Carrier stored program code for performing the Coding method and / or the decoding method, if the program product runs on a computer.
  • the invention So procedures can be in a computer program with a program code for carrying out the invention Procedure if the program is on a computer expires, be realized.
  • Fig. 1 shows an overview diagram for the invention preferred device for processing discrete-time Samples that represent an audio signal to integer ones Obtaining values based on the int MDCT integer transform algorithm is working.
  • the discrete-time ones Samples are passed through the device shown in FIG fenestrated and optionally in a spectral representation implemented.
  • the discrete-time samples that are connected to a Input 10 are fed into the device with a window w of length 2N discrete time Samples corresponds windowed to an output 12th to achieve integer windowed samples which are suitable to, by means of a transformation and in particular, the device 14 for executing an integer DCT converted into a spectral representation too become.
  • the integer DCT is designed to consist of N input values N output values to produce what, in contrast to the MDCT function 408 of Fig. 4a, which is windowed from 2N Samples due to the MDCT equation only N spectral values generated.
  • the discrete-time samples are first in a device 16, two discrete-time samples are selected, together form a vector of time-discrete samples represent.
  • a time-discrete sample the is selected by means 16, lies in the first Quarter of the window.
  • the other discrete-time sample lies in the second quarter of the window as it is based of Fig. 3 is executed in more detail.
  • the through the device 16 generated vector is now with a Rotary matrix of dimension 2 x 2 applied, this Operation is not performed immediately, but by means of several so-called lifting matrices.
  • a lifting matrix has the property of being only one Element, which depends on the window w and unequal Is "1" or "0".
  • Each of the three lifting matrices to the right of the equals sign has the value "1" as main diagonal elements. Furthermore, in each lifting matrix is a secondary diagonal element equal to 0, and a minor diagonal element from the rotational angle ⁇ dependent.
  • the vector is now filled with the third lifting matrix, i. H. the lifting matrix on the far right in the above equation, multiplied, to get a first result vector.
  • This is illustrated in FIG. 1 by a device 18.
  • the first result vector with any Rounding function which is the set of real numbers in the set of integer numbers, rounded, as it is in Fig. 1 is shown by a device 20.
  • the device 20 becomes a rounded first result vector receive.
  • the rounded first result vector becomes now in a device 22 for multiplying the same with the middle, d. H. second, fed to the lifting matrix, to obtain a second result vector which is in a device 24 is in turn rounded to a rounded to obtain the second result vector.
  • the rounded one second result vector is now in a device 26th fed, to multiply the same with the listed on the left in the above equation, d. H. first, Lifting matrix to obtain a third result vector finally rounded by means 28 is finally fenced at the output 12 integer Get samples, which now, if one spectral representation of the same is desired by the Device 14 needs to be processed to work on one Spectral output 30 to obtain integer spectral values.
  • the device 14 is an integer DCT or Integer DCT executed.
  • the coefficients of the DCT-IV form an orthonormal N x N Matrix.
  • Each orthogonal N x N matrix can be transformed into N (N-1) / 2 Givens rotations be decomposed, as in the technical publication P. P. Vaidyanathan, "Multirate Systems And Filter Banks”, Prentice Hall, Englewood Cliffs, 1993 is. It should be noted that also further decompositions exist.
  • DCT-IV which prefers here is comprised of non-symmetric basis functions, i. H. a cosine quarter-wave, a cosine 3/4 wave, a cosine 5/4 wave, a cosine 7/4 wave, etc.
  • DCT-II type II
  • the 0th basic function has a DC component
  • the first basic function is a half cosine wave
  • the second basis function is a whole cosine wave, etc. Due to the fact that the DCT-II takes the DC component into account, it is used in video coding, but not in audio coding, as opposed to audio coding for video coding the DC component is not relevant is.
  • An MDCT with a window length of 2N can be reduced into a discrete cosine transformation of type IV with a length N. This is accomplished by explicitly performing the TDAC operation in the time domain and then applying the DCT-IV. For a 50% overlap, the left half of the window for a block t overlaps the right half of the previous block, ie, the block t-1.
  • the overlapping part of two consecutive blocks t-1 and t is preprocessed in the time domain, ie before the transformation, as follows, ie processed between the input 10 and the output 12 of FIG. 1:
  • the values denoted by the tilde are the values at the output 12 of FIG. 1, while those without tilde in the above Equation x values denote the values at input 10 or behind the device 16 for selection.
  • the running index k runs from 0 to N / 2-1, while w represents the window function.
  • this preprocessing may be written in the time domain as a Givens rotation, as it has been executed.
  • window functions w can be used as long as they have this TDAC condition fulfill.
  • the discrete-time samples x (0) to x (2N-1), which share a window "windowed”, are initially so through the device 16 of FIG. 1, that the sample x (0) and the sample x (N-1), d. H. a sample from the first quarter of the window and a sample from the second quarter of the window, be selected to the Vector at the output of the device 16 to form.
  • Which Crossing arrows represent schematically the lifting multiplications and subsequent rounding of the facilities 18, 20 and 22, 24 and 26, 28, respectively, at the entrance the DCT-IV blocks the integer windowed samples to obtain.
  • 2N windowed windows at the exit 12 integer samples that now look like it is shown in Fig. 2, in a DCT-IV transformation be fed.
  • the windowed integer samples the first quarter of the window will be in one precedent DCT-IV along with the windowed integer ones Samples of the fourth quarter of the previous one Window processed. Similarly, the fourth quarter the windowed integer samples in Fig. 2 with the first quarter of the next window together into a DCT-IV transformation fed.
  • the middle one shown in FIG integer DCT-IV transformation 32 now provides N integer spectral values y (0) to y (N-1). This integer Spectral values can now be simple, for example Entropy-coded without any intervening Quantization is required because the fenestration and Transformation provides integer output values.
  • a decoder In the right half of Fig. 2, a decoder is shown.
  • the decoder consisting of inverse transformation and "inverse windowing" works inverse to the encoder. It is it is known that an inverse of the reverse transformation of a DCT-IV DCT-IV can be used as shown in FIG is.
  • the output values of the decoder DCT-IV 34 become now, as shown in Fig. 2, with the corresponding Values of the previous transformation or the subsequent transformation is inversely processed to the integer windowed samples at the output of the Device 34 or the preceding and following Transformation again discrete-time audio samples x (0) to produce x (2N-1).
  • the output side operation is done by inverse Givens rotation, ie, such that the blocks 26, 28, 22, 24 and 18, 20, respectively, are traversed in the opposite direction. This is illustrated in more detail with reference to the second lifting matrix of Equation 1. If (in the encoder) the second result vector is formed by multiplying the rounded first result vector by the second lifting matrix (means 22), the following expression results: ( x . y ) ⁇ ( x . y + x sin ⁇ )
  • Equation 6 The values x, y on the right side of Equation 6 are integers. However, this does not apply to the value x sin ⁇ .
  • the rounding function r must be introduced, as in the following equation ( x . y ) ⁇ ( x . y + r ( x sin .alpha)) is shown. This operation executes the device 24.
  • the inverse mapping (in the decoder) is defined as follows: ( x ' y ') ⁇ ( x ' y '- r ( x 'Sin .alpha))
  • the output values therefore always remain integer, whereby it it is preferred to also use integer input values.
  • PCM samples as stored on a CD are, are integer numerical values whose value range varies depending on the bit width, d. H. depending on whether the time-discrete digital input values 16-bit values or 24-bit values are. Still, that's how it's been done is, the entire process is invertible by the inverse Rotations are performed in reverse order. It Thus there exists an integer approximation of the MDCT perfect reconstruction, ie a lossless transformation.
  • the transformation shown provides integer output values instead of floating-point values. It provides a perfect reconstruction, so no mistake is introduced if one Forward and then a reverse transformation executed become.
  • the transformation is according to a preferred embodiment the present invention is a substitute for the modified discrete cosine transformation. Others too However, transformation techniques can be performed in integers be as long as a decomposition into rotations and one Disassembly of the rotations in lifting steps possible is.
  • the integer MDCT has the most favorable features the MDCT. It has an overlapping structure, which means a better frequency selectivity than non-overlapping ones Block transformations is obtained. by virtue of the TDAC function already in the window before the transformation is taken into account, becomes a critical scan maintained so that the total number of spectral values, which represent an audio signal, equal to the total number of input samples.
  • the floating-point samples shows, shows in the described preferred integer transformation that only in the Spectral range in which there is little signal level, the noise compared to normal MDCT is increased while up this Rauscherhöhung not at significant signal levels makes noticeable.
  • This is the integer processing for an efficient hardware implementation, since only multiplication steps are used without more in move-add steps (shift / add steps) can be disassembled, which is easy and fast can be implemented in hardware.
  • move-add steps shift / add steps
  • the integer transformation provides a good spectral Presentation of the audio signal and still remains in the range the whole numbers. When referring to tonal parts of an audio signal is applied, this results in a good E-nergiekonzentri für.
  • This can be an efficient lossless Coding scheme can be constructed by simply placing the in Fig. 1 shown fenestration / transformation with a Entropy coder is cascaded.
  • a stacked Coding (Stacked Coding) Using Escape Values as it is used in MPEG AAC, is favorable. It is preferable to set all values by a certain power of Shorten two until they are in a desired code table fit, and then the omitted low-order ones Code bits in addition.
  • An almost lossless Encoder could also be obtained by: simply omit certain of the least significant bits become.
  • the open-loop predictor is called TNS.
  • the quantization after the prediction leads to an adaptation of the resulting quantization noise to the temporal structure of the audio signal and therefore prevents Vorechos in psychoacoustic audio encoders.
  • For a lossless Audio coding is the second alternative, d. H. with a closed loop predictor, more appropriate, because closed-loop prediction is accurate Reconstruction of the input signal allowed. If those Technique applied to a generated spectrum must be Rounding step after each step of the prediction filter be performed to stay in the range of integers. By using the inverse filter and the same Rounding function can accurately restore the original spectrum getting produced.

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Claims (33)

  1. Dispositif pour coder un signal audio discret dans le temps, pour obtenir des données audio codées, aux caractéristiques suivantes :
    un dispositif (52) destiné à fournir un bloc de quantification de valeurs spectrales du signal audio discret dans le temps quantifiées à l'aide d'un modèle psycho-acoustique (54) ;
    un dispositif (58) destiné à quantifier inversement le bloc de quantification et à arrondir les valeurs spectrales quantifiées inversement, pour obtenir un bloc d'arrondi de valeurs spectrales quantifiées inversement arrondies ;
    un dispositif (56) destiné à générer un bloc de nombre entier de valeurs spectrales de nombre entier à l'aide d'un algorithme de transformation de nombre entier qui est réalisé de manière à générer, à partir d'un bloc de valeurs de balayage discrètes dans le temps de nombre entier, le bloc de nombre entier de valeurs spectrales ;
    un dispositif de combinaison (58) destiné à former un bloc de différence qui dépend d'une différence en valeurs spectrales entre le bloc d'arrondi et le bloc de nombre entier, pour obtenir un bloc de différence avec des valeurs spectrales de différence ;
    un dispositif (60) destiné à traiter le bloc de quantification et le bloc de différence, pour générer des données audio codées comportant des informations sur le bloc de quantification et des informations sur le bloc de différence.
  2. Dispositif selon la revendication 1, dans lequel le dispositif (52) destiné à fournir est réalisé
       pour générer, à partir d'un bloc de temps de valeurs de signal audio dans le temps, au moyen d'une MDCT, un bloc MDCT de valeurs spectrales MDCT, et
       pour quantifier le bloc MDCT à l'aide d'un modèle psycho-acoustique, pour générer le bloc de quantification qui présente des valeurs spectrales MDCT quantifiées.
  3. Dispositif selon la revendication 2,
       dans lequel le dispositif (56) destiné à générer le bloc de nombre entier est réalisé de manière d'effectuer une IntMDCT sur le bloc de temps, pour générer le bloc de nombre entier présentant des valeurs spectrales IntMDCT.
  4. Dispositif selon l'une des revendications précédentes,
       dans lequel le dispositif (52) destiné à fournir est réalisé de manière à calculer, à l'aide d'un algorithme de transformation à virgule flottante, le bloc de quantification.
  5. Dispositif selon l'une des revendications 1 à 3,
       dans lequel le dispositif (52) destiné à fournir est réalisé de manière à calculer le bloc de quantification à l'aide du dispositif (56) destiné à générer le bloc de nombre entier généré.
  6. Dispositif selon l'une des revendications précédentes,
       dans lequel le dispositif (60) destiné à traiter est réalisé de manière à soumettre le bloc de quantification à un codage entropique (60a), pour obtenir un bloc de quantification codé de manière entropique,
       à soumettre le bloc d'arrondi à un codage entropique (60b), pour obtenir un bloc d'arrondi codé de manière entropique, et
       à transmettre le bloc de quantification codé de manière entropique dans une première couche de modulation d'un flux de données modulé qui représente les données audio codées, et à transmettre le bloc d'arrondi codé de manière entropique dans une deuxième couche de modulation du flux de données modulé.
  7. Dispositif selon la revendication 6,
       dans lequel le dispositif (60) destiné à traiter est réalisé, par ailleurs, de manière à utiliser, pour le codage entropique du bloc de quantification, l'un parmi une pluralité de tableaux de code en fonction des valeurs spectrales quantifiées, et
       dans lequel le dispositif (60) destiné à traiter est réalisé, en outre, de manière à sélectionner, pour le codage entropique du bloc de différence, l'un parmi une pluralité de tableaux de code en fonction d'une propriété d'un dispositif de quantification pouvant être utilisé lors d'une quantification pour générer le bloc de quantification.
  8. Dispositif selon l'une des revendications précédentes,
       dans lequel le dispositif (52) destiné à fournir est réalisé de manière à utiliser, en fonction d'une qualité du signal audio, l'une parmi une pluralité de fenêtres, pour la division en fenêtres d'un bloc dans le temps des valeurs de signal audio, et
       dans lequel le dispositif (56) destiné à générer est réalisé de manière à effectuer, pour l'algorithme de transformation de nombre entier, la même sélection de fenêtre.
  9. Dispositif selon l'une des revendications 1 à 8,
       dans lequel le dispositif destiné à générer est réalisé de manière à utiliser un algorithme de transformation de nombre entier, présentant les étapes suivantes consistant à :
    diviser en fenêtres les valeurs de balayage discrètes dans le temps avec une fenêtre (w) d'une longueur correspondant à 2N valeurs de balayage discrètes dans le temps, pour fournir des valeurs de balayage discrètes dans le temp divisées en fenêtres pour une conversion des valeurs de balayage discrètes dans le temps en une représentation spectrale au moyen d'une transformation pouvant générer, à partir de N valeurs d'entrée, N valeurs de sortie, la division en fenêtres présentant les étapes partielles suivantes :
    sélectionner (16) une valeur de balayage discrète dans le temps d'un quart de la fenêtre et une valeur de balayage discrète dans le temps d'un autre quart de la fenêtre, pour obtenir un vecteur de valeurs de balayage discrètes dans le temps ;
    soumettre le vecteur à une matrice de rotation carrée dont la dimension coïncide avec la dimension du vecteur, la matrice de rotation pouvant être représentée par une pluralité de matrices d'élévation, une matrice d'élévation ne présentant qu'un seul élément qui est fonction de la fenêtre (w) et est différent de 1 ou 0, l'étape partielle de la soumission présentant les sous-étapes suivantes consistant à :
    multiplier (18) le vecteur par une matrice d'élévation, pour obtenir un premier vecteur de résultat ;
    arrondir (20) une composante du premier vecteur de résultat par une fonction d'arrondi (r) qui reproduit un nombre réel sur un nombre entier, pour obtenir un premier vecteur de résultat arrondi ; et
    réaliser de manière séquentielle les étapes de multiplication (22) et d'arrondi (24) par une autre matrice d'élévation, jusqu'à ce que toutes les matrices d'élévation soient traitées, pour obtenir un vecteur tourné qui présente une valeur de balayage divisée en fenêtres de nombre entier du quart de la fenêtre et une valeur de balayage divisée en fenêtres de nombre entier de l'autre quart de la fenêtre, et
    réaliser l'étape de division en fenêtres pour toutes les valeurs de balayage discrètes dans le temps des quarts restants de la fenêtre, pour obtenir 1N valeurs de balayage de nombre entier filtrées ; et
    convertir (14) N valeurs de balayage de nombre entier divisées en fenêtres en une représentation spectrale par une DCT de nombre entier pour des valeurs avec les valeurs de balayage de nombre entier filtrées du deuxième quart et du troisième quart de la fenêtre, pour obtenir N valeurs spectrales de nombre entier.
  10. Dispositif selon l'une des revendications précédentes,
       dans lequel le dispositif (52) destiné à fournir le bloc de quantification est réalisé de manière à réaliser, avant une étape de quantification (52b), une prédiction de valeurs spectrales sur la fréquence, à l'aide d'un filtre de prédiction, pour obtenir des valeurs spectrales résiduelles de prédiction qui représentent, après une quantification, le bloc de quantification ;
       dans lequel est, par ailleurs, prévu un dispositif de prédiction qui est réaliser de manière à effectuer une prédiction sur la fréquence des valeurs spectrales de nombre entier du bloc de nombre entier, par ailleurs étant prévu un dispositif d'arrondi, pour arrondir les valeurs spectrales résiduelles de prédiction sur base des valeurs spectrales de nombre entier représentant le bloc d'arrondi.
  11. Procédé selon l'une des revendications précédentes,
       dans lequel le signal audio discret dans le temps présente au moins deux canaux,
       dans lequel le dispositif (52) destiné à fournir est réalisé de manière à effectuer un codage de centre/côté avec des valeurs spectrales du signal audio discret dans le temps, pour obtenir, après une quantification de valeurs spectrales de centre/côté, le bloc de quantification, et
       dans lequel le dispositif (56) destiné à générer le bloc de nombre entier est réalisé de manière à effectuer également un codage centre/côté correspondant au codage de centre/côté du dispositif (52) destiné à fournir.
  12. Dispositif selon l'une des revendications précédentes,
       dans lequel le dispositif (60) destiné à traiter est réalisé de manière à générer un flux de données MPEG-2-AAC, dans un champ Ancilliary Data étant introduits des informations auxiliaires pour l'algorithme de transformation de nombre entier.
  13. Dispositif selon l'une des revendications précédentes,
       dans lequel le dispositif destiné à traiter (60) est réalisé de manière à sortir les données audio codées comme flux de données avec une pluralité de couches de modulation.
  14. Dispositif selon la revendications 13,
       dans lequel le dispositif destiné à traiter (60) est réalisé de manière à introduire dans une première couche de modulation (81) des informations sur le bloc de quantification et à introduire dans une deuxième couche de modulation (82) des informations sur le bloc de différence.
  15. Dispositif selon la revendication 13,
       dans lequel le dispositif destiné à traiter (60) est réalisé de manière à introduire dans une première couche de modulation des informations sur le bloc de quantification et à introduire les informations sur le bloc de différence dans au moins une deuxième et une troisième couche de modulation.
  16. Dispositif selon la revendication 15,
       dans lequel sont contenues, dans la deuxième couche de modulation, des valeurs spectrales de différence à précision réduite et, dans une ou plusieurs couches de modulation supérieures, une partie restante des valeurs spectrales de différence.
  17. Dispositif selon la revendication 15 ou 16,
       dans lequel les informations sur le bloc de différence comportent des valeurs spectrales de différence codées de manière binaire,
       dans lequel la deuxième couche de modulation comporte, pour les valeurs spectrales de différence, un certain de bits d'un bit à la valeur la plus élevée (MSB) à un bit de valeur inférieure (MSB-x) pour une valeur spectrales de différence, et
       dans lequel la troisième couche de modulation comporte un nombre de bits partant d'un bit à la valeur la plus basse (MSB-x-1) à un bit à la valeur la plus élevée (LSB).
  18. Dispositif selon la revendication 17,
       dans lequel le signal audio discret dans le temps est présent sous forme de valeurs de balayage avec une largeur de 24 bits, et
       dans lequel le dispositif destiné à traiter (60) est réalisé de manière à introduire 16 bits de valeur supérieure des valeurs spectrales de différence dans la deuxième couche de modulation, et à introduire les 8 bits restants d'une valeur spectrale de différence dans la troisième couche de modulation, de sorte qu'un décodeur atteigne, à l'aide de la deuxième couche de modulation, une qualité CD, un décodeur atteignant, à l'aide également de la troisième couche de modulation, une qualité de studio.
  19. Dispositif selon la revendication 15,
       dans lequel le dispositif (60) destiné à traiter est réalisé de manière à introduire dans une deuxième couche de modulation au moins une partie des valeurs spectrales de différence pour la représentation d'un signal filtré passe-bas et à introduire dans au moins une autre couche de modulation une différence entre les valeurs spectrales de différence dans la deuxième couche de modulation et les valeurs spectrales de différence originales.
  20. Dispositif selon la revendication 15 ou 19,
       dans lequel le dispositif (60) destiné à traiter est réalisé de manière à introduire dans une deuxième couche de modulation au moins une partie des valeurs spectrales de différence jusqu'à une fréquence limite déterminée, et à introduire dans une troisième couche de modulation au moins une partie des valeurs spectrales de différence de la fréquence limite déterminée à une fréquence supérieure.
  21. Procédé pour coder un signal audio discret dans le temps, pour obtenir des données audio codées, aux étapes suivantes consistant à :
    fournir (52) un bloc de quantification de valeurs spectrales du signal discret dans le temps quantifiées à l'aide d'un modèle psycho-acoustique (54) ;
    quantifier inversement (58) le bloc de quantification et arrondir les valeurs spectrales quantifiées inversement, pour obtenir un bloc d'arrondi de valeurs spectrales quantifiées inversement arrondies ;
    générer (56) un bloc de nombre entier de valeurs spectrales de nombre entier à l'aide d'un algorithme de transformation de nombre entier réalisé de manière à générer, à partir d'un bloc de valeurs de balayage discrètes dans le temps de nombre entier, le bloc de nombre entier de valeurs spectrales ;
    former (58) un bloc de différence qui est fonction d'une différence en valeurs spectrales entre le bloc d'arrondi et le bloc de nombre entier, pour obtenir un bloc de différence avec des valeurs spectrales de différence ; et
    traiter (60) le bloc de quantification et le bloc de différence, pour générer des données audio codées comportant les informations sur le bloc de quantification et des informations sur le bloc de différence.
  22. Dispositif pour décoder des données audio codées qui ont été générées à partir d'un signal audio discret dans le temps en fournissant (52) un bloc de quantification de valeurs spectrales du signal audio discret dans le temps quantifiées à l'aide d'un modèle psycho-acoustique (54), en quantifiant inversement (58) le bloc de quantification et en arrondissant les valeurs spectrales quantifiées inversement, pour obtenir un bloc d'arrondi de valeurs spectrales quantifiées inversement arrondies, en générant (56) un bloc de nombre entier de valeurs spectrales de nombre entier à l'aide d'un algorithme de transformation de nombre entier réalisé de manière à générer, à partir d'un bloc de valeurs de balayage discrètes dans le temps de nombre entier, le bloc de nombre entier de valeurs spectrales, et en formant (58) un bloc de différence qui est fonction d'une différence en valeurs spectrales entre le bloc d'arrondi et le bloc de nombre entier, pour obtenir un bloc de différence à valeurs spectrales de différence, aux caractéristiques suivantes :
    un dispositif (70) destiné à traiter les données audio codées, pour obtenir un bloc de quantification et un bloc de différence ;
    un dispositif (74) destiné à quantifier inversement et à arrondir le bloc de quantification, pour obtenir un bloc de quantification quantifié inversement de nombre entier ;
    un dispositif (78) destiné à combiner en valeurs spectrales le bloc de quantification de nombre entier et le bloc de différence, pour obtenir un bloc de combinaison ; et
    un dispositif (82) destiné à générer une représentation dans le temps du signal audio discret dans le temps à l'aide du bloc de combinaison et à l'aide d'un algorithme de transformation inverse de nombre entier inverse à l'algorithme de transformation de nombre entier.
  23. Dispositif pour décoder selon la condition 22,
       dans lequel les données audio codées sont modulées et comportent une pluralité de couches de modulation,
       dans lequel le dispositif (70) destiné à traiter les données audio codées est réalisé de manière à déterminer, comme première couche de modulation, le bloc de quantification à partir des données audio codées, et à déterminer, comme deuxième couche de modulation, le bloc de différence à partir des données audio codées.
  24. Dispositif selon la revendication 22,
       dans lequel les informations sur le bloc de différence comportent des couvertures valeurs spectrales de différence codées de manière binaires,
       dans lequel les données audio codées sont modulées et comportent une pluralité de couches de modulation,
       dans lequel le dispositif (70) destiné à traiter les données audio codées est réalisé de manière à déterminer, comme première couche de mesurage, le bloc de quantification à partir des données audio codées et à extraire, comme deuxième couche de modulation, une représentation des valeurs spectrales de différence à précision réduite.
  25. Dispositif selon la revendication 24,
       dans lequel le dispositif (70) destiné à traiter les données audio codées est réalisé de manière à extraire, comme deuxième couche de modulation, un nombre de bits partant d'un bit à la valeur la plus élevée jusqu'à un bit de valeur inférieure qui est d'une valeur supérieure à un bit à la valeur la plus basse d'une valeur spectrale de différence, et
       dans lequel le dispositif (82) destiné à générer une représentation dans le temps du signal audio discret dans le temps est réalisé de manière à générer synthétiquement, avant une utilisation de l'algorithme de transformation de nombre entier, des bits manquants pour une valeur spectrale de différence.
  26. Dispositif selon la revendication 25,
       dans lequel le dispositif (82) est réalisé de manière à effectuer, pour la génération synthétique, une modulation vers le haut de la deuxième couche de modulation, lors de la modulation vers le haut étant utilisé un facteur de modulation qui est égal à 2n, n étant égal au nombre de bits de valeur basse qui ne sont pas contenus dans la deuxième couche de modulation, ou à utiliser, pour la génération synthétique, un algorithme de Dithering.
  27. Dispositif selon la revendication 22,
       dans lequel les données audio codées sont modulées et comportent une pluralité de couches de modulation, et
       dans lequel le dispositif (70) destiné à traiter les données audio codées est réalisé de manière à déterminer, comme première couche de mesurage, le bloc de quantification à partir des données audio codées et à déterminer, comme deuxième couche de modulation, des valeurs spectrales de différence filtrées passe-bas.
  28. Dispositif selon la revendication 22 ou 27,
       dans lequel les données audio codées sont modulées et comportent une pluralité de couches de modulation, et
       dans lequel le dispositif (70) destiné à traiter les données audio codées est réalisé de manière à déterminer, comme première couche de modulation, le bloc de quantification à partir des données audio codées, et à déterminer, comme deuxième couche de modulation, des valeurs spectrales de différence jusqu'à une première fréquence limite, la première fréquence limite étant inférieure à une fréquence maximale d'une valeur spectrale de différence pouvant être générée dans un codeur.
  29. Dispositif selon la revendication 28,
       dans lequel le dispositif (82) destiné à générer une représentation dans le temps est réalisé de manière à régler des valeurs d'entrée dans un algorithme de transformation de nombre de pleine longueur qui se situent au-dessus de la fréquence limite de la deuxième couche de modulation à une valeur prédéterminée, et à balayer vers le bas la représentation dans le temps du signal audio discret dans le temps, après l'utilisation de l'algorithme de transformation de nombre entier inverse, d'un facteur qui est choisi en fonction d'un rapport entre une fréquence maximale d'une valeur spectrale de différence pouvant être générée par un codeur et la fréquence limite.
  30. Dispositif selon la revendication 29,
       dans lequel la valeur prédéterminée est, pour toutes les valeurs d'entrée au-dessus de la fréquence limite, égale à zéro.
  31. Procédé pour décoder des données audio codées qui ont été générées à partir d'un signal audio discret dans le temps par fourniture, quantification inverse, génération, formation et traitement, aux étapes suivantes consistant à :
    traiter (70) les données audio codées, pour obtenir un bloc de quantification et un bloc de différence ;
    quantifier inversement (74) le bloc de quantification et arrondir, pour obtenir un bloc de quantification quantifié inversement de nombre entier ;
    combiner en valeurs spectrales (78) le bloc de quantification de nombre entier et le bloc de différence, pour obtenir un bloc de combinaison ;
    générer (82) une représentation dans le temps du signal audio discret dans le temps à l'aide du bloc de combinaison et à l'aide d'un d'algorithme de transformation de nombre entier inverse à l'algorithme de transformation de nombre entier.
  32. Programme d'ordinateur avec un code de programme pour la mise en oeuvre du procédé pour coder selon la revendication 21, lorsque le programme est exécuté sur un ordinateur.
  33. Programme d'ordinateur avec un code de programme pour la mise en oeuvre du procédé pour décoder selon la revendication 31, lorsque le programme est exécuté sur un ordinateur.
EP02792858A 2002-04-18 2002-12-02 Dispositif et procede pour coder un signal audio a temps discret et dispositif et procede pour decoder des donnees audio codees Expired - Lifetime EP1495464B1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE10217297A DE10217297A1 (de) 2002-04-18 2002-04-18 Vorrichtung und Verfahren zum Codieren eines zeitdiskreten Audiosignals und Vorrichtung und Verfahren zum Decodieren von codierten Audiodaten
DE10217297 2002-04-18
PCT/EP2002/013623 WO2003088212A1 (fr) 2002-04-18 2002-12-02 Dispositif et procede pour coder un signal audio a temps discret et dispositif et procede pour decoder des donnees audio codees

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JP (1) JP4081447B2 (fr)
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EP2264699A2 (fr) 2006-11-02 2010-12-22 Fraunhofer-Gesellschaft zur Förderung der Angewandten Forschung e.V. Dispositif et procédé pour le post-traitement de valeurs spectrales et codeur et décodeur pour signaux audio
CN101553870B (zh) * 2006-11-02 2012-07-18 弗劳恩霍夫应用研究促进协会 后处理谱值的设备和方法及音频信号的编码器和解码器
EP2264699A3 (fr) * 2006-11-02 2012-10-10 Fraunhofer-Gesellschaft zur Förderung der Angewandten Forschung e.V. Dispositif et procédé pour le post-traitement de valeurs spectrales et codeur et décodeur pour signaux audio
US8321207B2 (en) 2006-11-02 2012-11-27 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Device and method for postprocessing spectral values and encoder and decoder for audio signals

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ATE305655T1 (de) 2005-10-15
CN1258172C (zh) 2006-05-31
WO2003088212A1 (fr) 2003-10-23
CA2482427A1 (fr) 2003-10-23
DE10217297A1 (de) 2003-11-06
KR20050007312A (ko) 2005-01-17
CA2482427C (fr) 2010-01-19
JP4081447B2 (ja) 2008-04-23
DE50204426D1 (de) 2005-11-03
HK1077391A1 (en) 2006-02-10
KR100892152B1 (ko) 2009-04-10
JP2005527851A (ja) 2005-09-15
AU2002358578A1 (en) 2003-10-27
CN1625768A (zh) 2005-06-08
EP1495464A1 (fr) 2005-01-12

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