EP2126905B1 - Verfahren und Vorrichtung zur Kodierung und Dekodierung von Audiosignalen, kodiertes Audiosignal - Google Patents

Verfahren und Vorrichtung zur Kodierung und Dekodierung von Audiosignalen, kodiertes Audiosignal Download PDF

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EP2126905B1
EP2126905B1 EP07866272A EP07866272A EP2126905B1 EP 2126905 B1 EP2126905 B1 EP 2126905B1 EP 07866272 A EP07866272 A EP 07866272A EP 07866272 A EP07866272 A EP 07866272A EP 2126905 B1 EP2126905 B1 EP 2126905B1
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signal
channel
filter
audio stream
channels
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EP2126905A1 (de
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Alexandre Delattre
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Nintendo European Research and Development SAS
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Mobiclip SAS
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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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
    • 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
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • 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
    • G10L19/04—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 predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Definitions

  • the present invention relates to a method and an audio coding device. It applies, in particular, coding enriched all or part of the audio spectrum, especially for transmission over a computer network, for example Internet, or storage on a digital information carrier.
  • This method and device can be integrated into any system for compressing and decompressing an audio signal on all hardware platforms.
  • the bit rate is often reduced by limiting the bandwidth of the audio signal.
  • low frequencies are retained because the human ear has a better resolution and spectral sensitivity in low frequency than in high frequency.
  • the low frequencies of the signal are kept, so that the data rate to be transferred is even lower.
  • some methods of the state of the art attempt, starting from the signal limited to the low frequencies, to extract harmonics which make it possible to recreate the high frequencies artificially . These methods are usually based on a spectral enrichment of recreating a high frequency spectrum by transposition of the low frequency spectrum, this high frequency spectrum being spectrally re-formed.
  • the resulting signal therefore consists, for the low frequency part, of the received low frequency signal and for the high frequency portion of the reformed enrichment.
  • the patent application US2006 / 0235678 describes the encoding of an audio signal with for each channel the provision of parameters of a spectral band replication ("SBR") spectral band extension method.
  • SBR spectral band replication
  • the invention relates to a method for encoding all or part of a multi-channel audio stream comprising a step of obtaining a composite signal obtained by composing the signals corresponding to each channel of the multi-channel audio stream; a step of obtaining a frequency-limited compound signal, the reduction of the frequency of the original composite signal being obtained by suppressing the high frequencies and a step of generating a temporal filter per channel making it possible to recover a signal that is spectrally close to the original signal of the corresponding channel when applied to the signal obtained by broadening the spectrum of the limited compound signal.
  • the filter corresponding to this channel is obtained by dividing member to member of a function of the coefficients of a Fourier transform applied on the one hand to the portion of the original signal and on the other hand to the corresponding portion of the signal obtained by broadening the spectrum of the limited signal.
  • Fourier transforms of different sizes are used to obtain a plurality of filters corresponding to each size used, the generated filter corresponding to one of the plurality of filters obtained by comparing the original signal and the signal obtained by applying the filter to the signal obtained by broadening the spectrum of the limited signal.
  • the choice of time filter can be made in a collection of predetermined time filters.
  • the filter is generated from the signal obtained by decoding and broadening the spectrum of the encoded limited compound signal and the signal. original.
  • the method further comprises a step of defining one of the channels of the multichannel audio stream as a reference channel; a time correlation step of each of the other channels on said reference channel defining for each channel an offset value and the step of composing the signals of each channel is performed with the signal of the reference channel and the signals correlated temporally for the other channels.
  • the offset value defined by the time correlation of the channel is associated with the generated filter.
  • the method further comprises a step of defining one of the channels of the multichannel audio stream as a reference channel; a step of equalizing each of the other channels on said reference channel defining for each channel an amplification value and the step of composing the signals of each channel is performed with the signal of the reference channel and the signals equalized for the other channels.
  • the amplification value defined by the temporal correlation of the channel is associated with the generated filter.
  • the invention also relates to a method for decoding all or part of a multi-channel audio stream comprising at least one step of receiving a transmitted signal; a step of receiving a time filter relating to the received signal for each channel of the multichannel audio stream; a step of obtaining a decoded signal by decoding the received signal; a step of obtaining an extended signal by enlargement the spectrum of the decoded signal and a step of obtaining a reconstructed signal by convolution of the extended signal with the received time filter for each channel of the multichannel audio stream.
  • a filter reduced in size from the generated filter is used in place of this filter generated in the step of obtaining a reconstructed signal for each channel.
  • the choice to use a reduced-size filter in place of the filter generated for each channel is according to the capabilities of the decoder.
  • one of the channels of the multichannel stream being defined as a reference channel, an offset value being associated with each received filter for the channels other than the reference channel
  • the method comprises in addition, a step of shifting the signal corresponding to each channel other than the reference channel making it possible to generate a temporal phase shift similar to the temporal phase shift between each channel and the reference channel in the original multi-channel audio stream.
  • the method further comprises a step of smoothing the offset values at the boundaries between the working windows so as to avoid a sudden change in the offset value for each channel other than the channel. reference.
  • one of the channels of the multichannel stream being defined as a reference channel, an amplification value being associated with each filter received for the channels other than the reference channel
  • the method further comprises a step of amplifying the signal corresponding to each channel other than the reference channel making it possible to generate a gain difference similar to the difference in gain between each channel and the reference channel in the original multi-channel audio stream .
  • the invention also relates to a device for encoding a multi-channel audio stream comprising at least means for obtaining a composite signal obtained by composing the signals corresponding to each channel of the multi-channel audio stream; means for obtaining a frequency-limited composite signal, the reduction of the spectrum of the original composite signal being obtained by suppressing high frequencies and means for generating a temporal filter per channel making it possible to recover a signal spectrally close to the original signal of the corresponding channel when it is applied to the signal obtained by broadening the spectrum of the limited signal.
  • the invention also relates to a device for decoding a multi-channel audio stream comprising at least the following means means for receiving a transmitted signal; means for receiving a time filter relating to the received signal for each channel of the multichannel audio stream; means for obtaining a decoded signal by decoding the received signal; means for obtaining an extended signal by broadening the spectrum of the decoded signal and means for obtaining a reconstructed signal by convolution of the extended signal with the time filter received for each channel of the multi-channel audio stream.
  • the invention also relates to a signal comprising a frequency-limited audio signal representing a frequency-limited version of an original audio signal resulting from the composition of the different channels of a multi-channel audio stream, characterized in that it also comprises channel time filter generation data for reconstructing a signal near the original signal of each channel when it is applied to an extended frequency version of the frequency limited audio signal contained in the signal.
  • the Fig. 1 represents the encoding method generally.
  • the signal 101 is the source signal to be encoded, this signal is then the original signal not limited in frequency.
  • Step 102 represents a frequency limiting step of the signal 101.
  • This frequency limitation can, for example, be achieved by subsampling of the signal 101 previously filtered by a low-pass filter. Subsampling consists of keeping only one sample on a set of samples and removing the other samples from the signal. A sub-sampling of a factor "n" where a sample is kept on n makes it possible to obtain a signal whose width of the spectrum will be divided by n. n is here a natural integer.
  • the signal limited in frequency and encoded at the output of the compression module 106 is also provided at the input of a decoding module 107.
  • This module performs the inverse operation of the encoding module 106 and makes it possible to construct a limited version of the signal. frequency identical to the version to which the decoder will have access when it also performs this decoded operation of the limited signal and encoded it will receive.
  • the limited signal thus decoded is then restored to the original spectral range by a frequency enrichment module 103.
  • This frequency enrichment may, for example, consist of a simple over-sampling of the input signal by the insertion of zero-valued samples between the samples of the input signal. Any other method of enriching the signal spectrum can also be used.
  • This extended frequency signal derived from the frequency enrichment module 103, is then supplied to a filter generation module 104.
  • This filter generation module 104 also receives the original signal 101 and calculates a time filter allowing, when it is applied to the extended signal from the frequency enrichment module 103, to shape it to get closer to the original signal.
  • the filter thus calculated is then supplied to the multiplexer 108 after an optional compression step 105.
  • the Fig. 2 represents generally the corresponding decoding method.
  • the decoder therefore receives the signal from the multiplexer 108 of the encoder. It demultiplexes it to obtain on the one hand the encoded frequency limited signal, called S1b, and the filter coefficients F, contained in the transmitted signal.
  • the signal S1b is then decoded by a decoding or decompression module 202 that is functionally equivalent to the module 107 of the Fig. 1 .
  • the signal is extended in frequency by the module 203 functionally equivalent to the module 103 of the Fig. 1 . A decoded and extended version of the signal is thus obtained.
  • the coefficients of the filter F are decoded if they had been encoded or compressed by a decompression module 201, and the filter obtained is applied to the extended time signal in a signal conditioning module 204. then an output signal close to the original signal.
  • This treatment is simple to implement because of the temporal nature of the filter to be applied to the signal for shaping.
  • the transmitted filter and thus applied during the reconstruction of the signal, is transmitted periodically and changes over time.
  • This filter is adapted to a portion of the signal to which it applies. It is thus possible to calculate for each signal portion a time filter particularly adapted according to the dynamic spectral characteristics of this portion of signal. In particular, it is possible to have several types of time filter generators and to select for each portion of signal the filter giving the best result for this portion. This is possible because the filter generation module has on the one hand the original signal and on the other hand the extended signal as it will be reconstructed by the decoder, so it is able, in the case where it is generated by several different filters, compare the signal obtained by applying each filter to the extended signal portion and the original signal that is to be approached as closely as possible. This method of filter generation is therefore not limited to choosing a type of filter determined for the entire signal but allows to change the type of filter according to the characteristics of each portion of the signal.
  • a signal sampled at a given frequency 301 for example 32 kHz
  • S1b the signal limited to its low frequencies
  • a filter F for shaping the signal obtained by extending the signal S1b frequency.
  • This signal is then encoded, for example by a method of the PCM ("Pulse Code Modulation") type, by the module 311 which will then be compressed, for example by an ADPCM module 312. This gives the subsampled signal containing the low frequencies of the original signal 301. This signal is sent to the multiplexer 314 to be transmitted to the decoder.
  • PCM Pese Code Modulation
  • this signal is transmitted to a decoding module 313.
  • This signal that the decoder will obtain from the signal sent to it is simulated.
  • This signal that will be used for the generation of the filter F will therefore allow to take into account the artifacts resulting from these coding and decoding phases, compression and decompression.
  • This signal is then frequency-expanded by inserting n-1 zero between each sample of the time signal in the module 303. In this way, a signal of the same spectral range as the original signal is reconstructed. According to the Nyquist theorem, we obtain a n-order spectrum folding.
  • the signal is downsampled from an order 2 to the encoding and oversampled from an order 2 to the decoding.
  • the spectrum is duplicated "mirrored" by axial symmetry in the frequency domain.
  • a Fourier transform is performed on the frequency-extended time signal from the module 303.
  • a sliding fast Fourier transform is performed on work windows of given size and variable. These sizes are typically 128, 256, 512 samples but can be of any size even if one will preferentially use powers of two to simplify calculations.
  • the modules of these transforms applied to these windows are then calculated. A same Fourier transform calculation is performed on the original signal in the module 306.
  • a member-to-member division 305 is then performed between the modules of the Fourier transform coefficients obtained by steps 304 and 306 to generate by inverse Fourier transforms temporal filters of sizes proportional to those of the windows used, ie 128, 256 or 512.
  • This step therefore generates several filters of different sizes among which we will have to choose the filter finally used. It will be seen that this selection step is performed by the module 309. Since the coefficients of the ratio between the windows are real, symmetrical in the frequency space, the equivalent filter F is then, in the time domain, real and symmetrical.
  • This property of symmetry can be used to transmit only half of the coefficients, the other being deduced by symmetry.
  • Obtaining a symmetrical real filter also makes it possible to reduce the number of operations required during the convolution of the received signal extended by the filter in the decoder.
  • Other embodiments make it possible to obtain real unsymmetrical filters. For example, if the temporal signal in a working window is frequency-limited, it is advantageous to iteratively determine the parameters of an infinite impulse response filter, Chebychev low-pass from the spectra from steps 304 and 306. and the cutoff frequency of the window.
  • a module 308 will offer other types of filters.
  • it can offer linear, cubic or other filters.
  • these filters are known to allow oversampling.
  • the module 308 therefore contains an arbitrary number of such filters that can be used.
  • the choice module 309 will therefore have as input a collection of filters. On the one hand, it will have the filters generated by the module 307 and corresponding to the filters generated for different window sizes by dividing the modules of the Fourier transforms applied to the original signal and the reconstructed signal. On the other hand, it will also have in input the original signal 301 and the reconstructed signal coming from the module 303. In this way the module 309 can compare the application of the different filters to the reconstructed signal coming from the module 303 with the original signal to choose the filter giving, on the signal portion considered, the best output signal, that is to say the spectrally closest to the original signal.
  • the filter generating the minimum of a function of the distortion.
  • This portion of the signal called the working window
  • the working window will have to be larger than the largest window used for calculating the filters.
  • the size of this working window may also vary depending on the signal. Indeed, a large working window size can be used for the encoding of a substantially stationary signal portion while a shorter window will be more suitable for a more dynamic signal portion to better take into account the rapid variations. . It is this part that makes it possible to select, for each portion of the signal, the most relevant filter allowing the best reconstruction by the decoder of the signal and to get closer to the original signal.
  • the module 310 will quantify the spectral coefficients of the filter that will be encoded, for example, using a Huffman table to optimize the data to be transmitted.
  • Multiplexer 314 will thus multiplex with each portion of the signal, the most relevant filter for the decoding of this portion of signal.
  • This filter being chosen either in the collection of filters of different sizes generated by analysis of this portion of the signal, or in the collection, also comprises a series of determined, typically linear filters, allowing the reconstruction, which can be chosen if they occur. reveal more interesting for the reconstruction by the decoder of the signal portion.
  • the generated filter is one of the determined filters, it is possible to transmit only an identifier identifying this filter from the collection of determined, typically linear filters, allowing the reconstruction, which can be chosen if they prove more interesting for the reconstruction by the decoder of the portion of signal.
  • the generated filter is one of the determined filters, it is possible to transmit only an identifier identifying this filter from the collection of determined filters provided by the module 308, as well as possible parameters of the filter. Indeed, since the coefficients of these determined filters are not calculated as a function of the portion of the signal to which they are to be applied, it is unnecessary to transport these coefficients which may be known to the decoder. Thus, the bandwidth for transporting information relating to the filter is reduced in this case to a simple identifier of the filter.
  • the Fig. 4 represents the corresponding decoding in the particular embodiment described.
  • the signal is received by the decoder which demultiplexes the signal.
  • the audio signal S1b is then decoded by the module 404 and then oversampled by a factor n by the insertion of n-1 samples at zero between the samples received by the module 405.
  • the spectral coefficients of the filter F are dequantized and decoded according to the Huffman tables by the module 401.
  • the size of the filter can be adapted by the module 402 of the decoder to its computing or memory capacity or any possible hardware limitation.
  • a decoder with few resources can use a subsampled filter which will allow it to reduce the operations during the application of the filter.
  • the subsampled filter can also be generated by the encoder according to the resources of the transmission channel or the resources of the decoder, provided of course that the latter information is held by the encoder.
  • the spectrum of the filter can be reduced to decoding to perform a smaller oversampling (n-1, n-2 etc. ..) depending on the hardware sound output capabilities of the decoder such as the power or the output capabilities its .
  • the module 403 then performs an inverse Fourier transform on the spectral coefficients of the filter to obtain the real filter in the time domain.
  • the filter is moreover symmetrical which makes it possible to reduce the data transported for the transmission of the filter.
  • the module 406 operates the convolution of the oversampled signal from the module 405 with the filter thus reconstituted to obtain the resulting signal.
  • This convolution is particularly greedy in calculation because the oversampling is done by inserting null values.
  • the fact that the filter is real, and even symmetrical in the preferred embodiment also reduces the number of operations required for this convolution.
  • the invention offers the advantage of performing a reshaping, not only of the high part of the spectrum reconstituted from the transmitted lower part but of the whole of the signal thus reconstituted. In this way, it makes it possible to model the part of the non-transmitted spectrum but also to correct artifacts due to the various operations of compression, decompression, encoding and decoding of the transmitted low frequency part.
  • a secondary advantage of the invention is the ability to dynamically adapt the filters used according to the nature of each portion of the signal thanks to the module allowing the choice of the best filter, in terms of sound quality and "machine time” used, among several for each portion of the signal.
  • the encoding method thus described for a single-channel signal can be adapted for a multi-channel signal.
  • a first obvious adaptation is the application of the single-channel solution to each audio channel independently. This solution is nevertheless expensive in that it does not take advantage of the strong correlation between the different channels of a multi-channel audio stream.
  • the proposed solution is to compose a single channel from the different channels of the stream. A treatment similar to that described previously in the case of a single-channel signal is then performed on this compound stream.
  • a filter is determined for each channel so as to reproduce the considered channel when it is applied to the composite stream.
  • the process will now be described more precisely using the Figures 5 and 6 in the case of stereophony.
  • the stereophonic embodiment naturally extends to a stream composed of more than two channels such as a stream 5.1 for home theater for example.
  • the Fig. 5 represents the architecture of a stereophonic encoder according to an embodiment of the invention.
  • the audio stream to be encoded consists of a left channel "L" referenced 501 and a right channel “R” referenced 502.
  • a composition module 503 composes these two signals to generate a composite signal.
  • This composition may, for example, be an average of the two channels, the composite signal is then equal to L + R / 2.
  • This compound signal then undergoes the same processing as the single-channel signal described above. It is sub-sampled by a factor n by the subsampling module 504.
  • the subsampled signal is then coded by an encoder 505 to be encoded by an encoder 506.
  • These modules are the same as the modules already described 311 and 312 of the Fig. 3 .
  • the subsampled and encoded composite signal is transmitted to the recipient of the stream. It is also decoded by a decoding module 507 corresponding to the module 313 of the Fig. 3 . Then, it is oversampled by the sampling module 508 corresponding to the module 303. The signal is then processed by two filter generation modules 509 and 510. Each of these modules corresponds to the modules 304, 305, 306, 308, 309. and 310 of the Fig. 3 . The first, 509, generates a filter F R which, when applied to the compound stream coming from the module 508, generates a signal close to the right channel R. This module takes as input the composite signal from the module 508 and the signal of the original R right channel 502.
  • the second, 510 generates a filter F L which, when applied to the compound stream coming from the module 508, generates a signal close to the left channel L.
  • This module takes as input the composed signal from the module 508 and the original left channel signal L 501. These filters, or an identifier of these filters, are then multiplexed with the subsampled stream and encoded from the encoding module 506 to be sent to the receiver .
  • the different channels of a multi-channel signal have a strong correlation but exhibit a temporal phase shift.
  • a slight time shift occurs between the signals of the different channels.
  • this offset tends to generate noise.
  • This registration is performed by temporal correlation between the channel to be recalibrated and the reference channel.
  • This correlation defines an offset value on the working window chosen for the correlation.
  • This working window is advantageously chosen equal to the working window used for generating the filter.
  • the value of the offset can then be associated with the generated filter to be transmitted in addition to the filters so as to allow reconstruction of the original inter-channel phase shift during the reproduction of the audio stream.
  • a step of equalizing the gains of the signals of the different channels can be used to homogenize the powers of the signals corresponding to the different channels.
  • This equalization defines an amplification value to be applied to the signal on the working window.
  • This amplification value can be introduced into the calculated filter allowing the reconstruction of the signal at decoding.
  • the calculation of this Amplification value is for each channel except one chosen as the reference channel. The introduction of the amplification value makes it possible to reconstruct, at decoding, the differences in gains between the channels in the original signal.
  • the generation calculation of a filter as well as that of the phase shift is done on a portion of signal called work window ( frame in English).
  • work window frame in English
  • the passage from one window to another will therefore cause a change in phase shift between the channels. This change may cause noise during playback.
  • the Fig. 6 represents the architecture of a stereophonic embodiment of the decoder. This figure is the stereophonic counterpart of Fig. 4 .
  • the received audio stream is demultiplexed to obtain the encoded low frequency composite stream called S 1b and the filters F R and F L.
  • the compound stream is then decoded by the decoding module 601 corresponding to the module 404 of the Fig. 4 . Its spectrum is then widened in frequency by the sampling module 602 corresponding to the module 405 of the Fig. 4 .
  • the signal thus obtained is then convoluted by the filters F R and F L decompressed by the modules 603 and 605 to restore the right and left channels S R and S L.
  • phase shift information is introduced into the stream, the channel that does not serve as a reference channel for the phase shift is recalibrated using this information to generate the phase shift of the original channels.
  • This phase shift information may, for example, take the form of an offset value associated with each of the filters for the channels other than the channel defined as reference channel.
  • this offset is smoothed, for example linearly, between the different work windows.

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Claims (18)

  1. Verfahren zur Codierung der Gesamtheit oder eines Teils eines Mehrkanal-Audiodatenflusses, enthaltend mindestens die folgenden Schritte:
    - einen Schritt zum Erhalten eines zusammengesetzten Signals, das durch die Zusammensetzung der Signale erhalten wird, die jedem Kanal des Mehrkanal-Audiodatenflusses entsprechen;
    - einen Schritt zum Erhalten eines zusammengesetzten frequenzbegrenzten Signals, wobei die Reduzierung der Frequenz des ursprünglichen zusammengesetzten Signals durch Unterdrückung der hohen Frequenzen erreicht wird;
    dadurch gekennzeichnet, dass es des Weiteren enthält:
    - einen Schritt zum Generieren eines Zeitfilters pro Kanal, das das Wiedergewinnen eines Signals ermöglicht, das spektral nahe am ursprünglichen Signal des entsprechenden Kanals liegt, wenn es an das Signal angelegt wird, das durch Erweiterung des Spektrums des zusammengesetzten begrenzten Signals erhalten wird.
  2. Verfahren nach Anspruch 1, bei welchem für einen Anteil des ursprünglich gegebenen Signals für einen gegebenen Kanal das Filter, das diesem Kanal entspricht, durch elementweise Teilung einer Funktion von Koeffizienten einer Fourier-Transformierten erhalten wird, die einerseits an den Anteil des ursprünglichen Signals und andererseits an den entsprechenden Anteil des durch Erweiterung des Spektrums des begrenzten Signals erhaltenen Signals angelegt wird.
  3. Verfahren nach Anspruch 2, bei welchem Fourier-Transformierte mit unterschiedlichen Größen verwendet werden, um eine Vielzahl von Filtern zu erhalten, die jeder verwendeten Größe entsprechen, wobei das generierte Filter einer Auswahl aus der Vielzahl von Filtern entspricht, die durch Vergleich des ursprünglichen Signals und des Signals erhalten werden, das durch Anlegen des Filters an dem durch Erweiterung des Spektrums des begrenzten Signals erhaltenen Signal erhalten wird.
  4. Verfahren nach einem der Ansprüche 1 bis 3, bei welchem die Auswahl des Zeitfilters aus einer Sammlung von vorbestimmten Zeitfiltern erfolgen kann.
  5. Verfahren nach einem der Ansprüche 1 bis 4, bei welchem, da das zusammengesetzte frequenzbegrenzte Signal im Hinblick auf seine Übertragung codiert wird, die Generierung des Filters ausgehend von dem durch Decodierung und Erweiterung des Spektrums des zusammengesetzten begrenzten codierten Signals und des ursprünglichen Signals erhaltenen Signal erfolgt.
  6. Verfahren nach einem der Ansprüche 1 bis 5, dadurch gekennzeichnet, dass es des Weiteren enthält:
    - einen Schritt zum Definieren eines der Kanäle des Mehrkanal-Audiodatenflusses als Referenzkanal;
    - einen Schritt zum zeitlichen Korrelieren jedes der anderen Kanäle mit dem Referenzkanal, wobei für jeden Kanal ein Phasenverschiebungswert definiert wird;
    - wobei der Schritt zum Zusammensetzen der Signale jedes Kanals mithilfe des Signals des Referenzkanals und der zeitlich korrelierten Signale für die anderen Kanäle ausgeführt wird.
  7. Verfahren nach Anspruch 6, dadurch gekennzeichnet, dass für jeden Kanal mit Ausnahme des Referenzkanals der durch die zeitliche Korrelation des Kanals definierte Phasenverschiebungswert mit dem generierten Filter verbunden wird.
  8. Verfahren nach einem der Ansprüche 1 bis 5, dadurch gekennzeichnet, dass es des Weiteren enthält:
    - einen Schritt zum Definieren eines der Kanäle des Mehrkanal-Audiodatenflusses als Referenzkanal;
    - einen Schritt zum Abgleichen jedes der anderen Kanäle mit dem Referenzkanal, wobei für jeden Kanal ein Verstärkungswert definiert wird;
    wobei der Schritt zum Zusammensetzen der Signale jedes Kanals mithilfe des Signals des Referenzkanals und der abgeglichenen Signale für die anderen Kanäle ausgeführt wird.
  9. Verfahren nach Anspruch 8, dadurch gekennzeichnet, dass für jeden Kanal mit Ausnahme des Referenzkanals der durch die zeitliche Korrelation des Kanals definierte Verstärkungswert mit dem generierten Filter verbunden wird.
  10. Verfahren zum Decodieren der Gesamtheit oder eines Teils eines Mehrkanal-Audiodatenflusses, enthaltend mindestens die folgenden Schritte:
    - einen Schritt zum Empfangen eines übertragenen Signals;
    dadurch gekennzeichnet, dass es des Weiteren enthält:
    - einen Schritt zum Empfangen eines Zeitfilters in Bezug auf das empfangene Signal für jeden Kanal des Mehrkanal-Audiodatenflusses;
    - einen Schritt zum Erhalten eines decodierten Signals mittels Decodieren des empfangenen Signals;
    - einen Schritt zum Erhalten eines durch Erweiterung des Spektrums des decodierten Signals erweiterten Signals;
    - einen Schritt zum Erhalten eines Signals, das durch Konvolution des erweiterten Signals mit dem empfangenen Zeitfilter für jeden Kanal des Mehrkanal-Audiodatenflusses rekonstruiert wird.
  11. Verfahren nach Anspruch 10, bei welchem ein ausgehend vom generierten Filter in der Größe reduziertes Filter an Stelle dieses generierten Filters in dem Schritt zum Erhalten eines rekonstruierten Signals für jeden Kanal verwendet wird.
  12. Verfahren nach Anspruch 11, bei welchem die Auswahl für die Verwendung eines in der Größe reduzierten Filters an Stelle des für jeden Kanal generierten Filters abhängig von der Leistungsfähigkeit der Decodiereinrichtung erfolgt.
  13. Verfahren nach einem der Ansprüche 10 bis 12, dadurch gekennzeichnet, dass, da einer der Kanäle des Mehrkanal-Datenflusses als Referenzkanal definiert ist, ein Phasenverschiebungswert mit jedem empfangenen Filter für die Kanäle mit Ausnahme des Referenzkanals verbunden ist, das Verfahren des Weiteren enthält:
    - einen Schritt zur Phasenverschiebung des Signals, das jedem Kanal mit Ausnahme des Referenzkanals entspricht, die es gestattet, eine zeitliche Phasenverschiebung zu generieren, die der zeitlichen Phasenverschiebung zwischen jedem Kanal und dem Referenzkanal in dem ursprünglichen Mehrkanal-Audiodatenfluss ähnlich ist.
  14. Verfahren nach Anspruch 13, dadurch gekennzeichnet, dass es des Weiteren enthält:
    - einen Schritt zum Glätten der Phasenverschiebungswerte an den Grenzen zwischen den Arbeitsfenstern, so dass eine abrupte Änderung des Phasenverschiebungswerts für jeden Kanal mit Ausnahme des Referenzkanals vermieden wird.
  15. Verfahren nach einem der Ansprüche 10 bis 12, dadurch gekennzeichnet, dass, da einer der Kanäle des Mehrkanal-Datenflusses als Referenzkanal definiert ist, ein Verstärkungswert mit jedem empfangenen Filter für die Kanäle mit Ausnahme des Referenzkanals verbunden ist, das Verfahren des Weiteren enthält:
    - einen Schritt zum Verstärken des Signals, das jedem Kanal mit Ausnahme des Referenzkanals entspricht, der es gestattet, eine Verstärkungsdifferenz zu generieren, die der Verstärkungsdifferenz zwischen jedem Kanal und dem Referenzkanal in dem ursprünglichen Mehrkanal-Audiodatenfluss ähnlich ist.
  16. Vorrichtung zur Codierung eines Mehrkanal-Audiodatenflusses, die mindestens enthält:
    - Mittel zum Erhalten eines zusammengesetzten Signals, das durch die Zusammensetzung der jedem Kanal des Mehrkanal-Audiodatenflusses entsprechenden Signale erhalten wird;
    - Mittel zum Erhalten eines zusammengesetzten frequenzbegrenzten Signals, wobei die Reduzierung des Spektrums des ursprünglichen zusammengesetzten Signals durch Unterdrückung der hohen Frequenzen erreicht wird;
    dadurch gekennzeichnet, dass sie des Weiteren enthält:
    - Mittel zum Generieren eines Zeitfilters pro Kanal, die das Wiedergewinnen eines Signals ermöglichen, das spektral nahe am ursprünglichen Signal des entsprechenden Kanals liegt, wenn es an dem Signal angelegt wird, das durch Erweiterung des Spektrums des begrenzten Signals erhalten wird.
  17. Vorrichtung zur Decodierung eines Mehrkanal-Audiodatenflusses, die mindestens folgende Mittel enthält:
    - Mittel zum Empfangen eines übertragenen Signals;
    dadurch gekennzeichnet, dass sie des Weiteren enthält:
    - Mittel zum Empfangen eines Zeitfilters in Bezug auf das empfangene Signal für jeden Kanal des Mehrkanal-Audiodatenflusses;
    - Mittel zum Erhalten eines decodierten Signals mittels Decodierung des empfangenen Signals;
    - Mittel zum Erhalten eines erweiterten Signals mittels Erweiterung des Spektrums des decodierten Signals;
    - Mittel zum Erhalten eines mittels Konvolution des erweiterten Signals mit dem Zeitfilters, der für jeden Kanal des Mehrkanal-Audiodatenflusses erhalten wird, rekonstruierten Signals.
  18. Signal, das ein frequenzbegrenztes Audiosignal umfasst, das eine frequenzbegrenzte Version eines ursprünglichen Audiosignals darstellt, das aus der Zusammensetzung der verschiedenen Kanäle eines Mehrkanal-Audiodatenflusses resultiert, dadurch gekennzeichnet, dass es des Weiteren Daten zum Generieren eines Zeitfilters pro Kanal umfasst, das die Rekonstruktion eines Signals ermöglicht, das dem ursprünglichen Signal jedes Kanals nahe ist, wenn es an eine frequenzerweiterte Version des frequenzbegrenzten Audiosignals angelegt wird, die im Signal enthalten ist.
EP07866272A 2006-12-28 2007-12-28 Verfahren und Vorrichtung zur Kodierung und Dekodierung von Audiosignalen, kodiertes Audiosignal Active EP2126905B1 (de)

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FR0611481A FR2911031B1 (fr) 2006-12-28 2006-12-28 Procede et dispositif de codage audio
FR0708067A FR2911020B1 (fr) 2006-12-28 2007-11-16 Procede et dispositif de codage audio
PCT/EP2007/011442 WO2008080609A1 (fr) 2006-12-28 2007-12-28 Procede et dispositif de codage audio

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EP2126905A1 (de) 2009-12-02
FR2911020B1 (fr) 2009-05-01
FR2911020A1 (fr) 2008-07-04
US8340305B2 (en) 2012-12-25
US20100046760A1 (en) 2010-02-25

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