EP2126904B1 - Tonkodierverfahren und vorrichtung - Google Patents

Tonkodierverfahren und vorrichtung Download PDF

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EP2126904B1
EP2126904B1 EP07866270A EP07866270A EP2126904B1 EP 2126904 B1 EP2126904 B1 EP 2126904B1 EP 07866270 A EP07866270 A EP 07866270A EP 07866270 A EP07866270 A EP 07866270A EP 2126904 B1 EP2126904 B1 EP 2126904B1
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signal
filter
limited
original
frequency
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EP2126904A1 (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
    • G10L19/04—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 and decoding 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 .
  • the document US2003 / 0158726 A1 describes an example of a spectral reconstruction method comprising during the encoding phase a step of obtaining a frequency-limited audio signal, and a step of generating a temporal filter making it possible to recover a signal that is spectrally close to the original signal.
  • This document also describes a decoding method implementing a temporal filter for generating a reconstructed signal.
  • the filter 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 signal original and secondly 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 a choice among 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 is extended to a collection of predetermined time filters.
  • the frequency-limited signal being encoded for transmission is made from the signal obtained by decoding and broadening the spectrum of the encoded limited signal and the original signal.
  • 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.
  • the choice to use a reduced size filter in place of the generated filter is according to the capabilities of the decoder.
  • the invention also relates to a signal comprising a frequency-limited audio signal representing a frequency limited version of an original audio signal, characterized in that it furthermore comprises generation data of a temporal filter allowing the reconstruction of a signal close to the original signal when 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 may, for example, be achieved by subsampling the signal 101 previously filtered by a low-pass filter. Subsampling consists of keeping only one sample on a set of samples and to remove 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 operation of decoding the limited signal and encoded it will receive.
  • the limited signal thus decoded is then restored to the original spectral extent by a frequency-widening module 103.
  • This frequency widening may, for example, consist of a simple oversampling of the input signal by the insertion of zero-valued samples between the samples of the input signal. Any other method of broadening the signal spectrum can also be used.
  • This extended frequency signal from the frequency-widening 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-widening module 103, to shape it to approach 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 signal limited in encoded frequency, called Slb, and the coefficients of the filter F, contained in the transmitted signal.
  • the signal Slb 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, to compare the signal obtained by application of each filter to the extended signal portion and the original signal which is to approach closer. This method of filter generation is therefore not limited to choosing a type of filter determined for the whole signal but allows to change the type of filter according to the characteristics of each portion of signal.
  • a signal sampled at a given frequency 301 for example 32 kHz
  • Slb the signal limited to its low frequencies
  • Another aim is to determine a filter F making it possible to shape the signal obtained by extending the signal Slb in 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 extended in frequency by inserting n-1 zeros between each sample of the time signal in the module 303. In this way, a signal of the same spectral extent 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 working windows of given and variable sizes. These sizes are typically 128, 256, 512 samples but can be of any size, even if powers of two will be used to simplify the calculations.
  • 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 the 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 spectra 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 necessary 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 305 and corresponding to the filters generated for different window sizes by division of the Fourier transform modules 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. For example, it is possible to relate the spectrum obtained by applying the filter to the signal from the module 303 and the spectrum of the same portion of the original signal.
  • This portion of the signal will have to be larger than the largest window used for calculating the filters. It will be possible to use typically a working window size of 512 samples. 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 from the collection of filters of different sizes generated by analysis of this portion of the signal, or from the collection also comprises a series of determined, typically linear, filters enabling reconstruction, which may be chosen if they prove 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 filters provided by the module 308, as well as possible parameters of the filter.
  • the coefficients of these determined filters are not calculated according to the portion of signal to which we want to apply them, it is unnecessary to carry these coefficients that can be known to the decoder.
  • 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 Slb is then decoded by the module 404 and then oversampled by a factor n by the insertion of n-1 zero samples 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 power or sound output capabilities.
  • 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 allows to model the part of the spectrum not transmitted 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.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)

Claims (11)

  1. Verfahren zur Kodierung eines gesamten oder wenigstens eines Teils eines Audiosignals umfassend zumindest die folgenden Schritte:
    - einen Schritt, des Erhaltens eines Signals, das hinsichtlich der Frequenz begrenzt ist, wobei die Reduzierung der Frequenz des Originalsignals durch ein Unterdrücken der hohen Frequenzen erreicht wurde,
    dadurch gekennzeichnet, dass
    das Verfahren weiterhin umfasst
    - einen Schritt des Erzeugens eines zeitlichen Filters, der es ermöglicht, ein Signal wieder aufzufinden, das dem Originalsignal in spektraler Hinsicht nahe kommt, wenn er auf das Gesamtsignal angewendet wird, welches auf den ursprünglichen Spektralbereich zurückgeführt wurde, indem das Spektrum des begrenzten Signals erweitert wurde.
  2. Verfahren gemäß Anspruch 1,
    wobei man den Filter für einen Abschnitt des vorgegebenen Originalsignals erhält, indem man die Koeffizientenfunktionen einer zum einen auf den Abschnitt des Originalsignals und zum anderen auf den entsprechenden Abschnitt des durch Frequenzerweiterung des beschränkten Signals erhaltenen Signals angewandten FourierTransformation Glied für Glied dividiert.
  3. Verfahren gemäß Anspruch 2,
    wobei Fourier-Transformationen unterschiedlicher Größen dazu verwendet werden, mehrere, der jeweils verwendeten Größe entsprechenden Filter zu erhalten, wobei der erzeugte Filter einer Auswahl aus den mehreren, erhaltenen Filtern entspricht, wobei die Auswahl durch einen Vergleich des Originalsignals mit dem durch Anwendung des Filters auf das durch Frequenzerweiterung des beschränkten Signals erhaltene Signal erfolgt.
  4. Verfahren nach einem der Ansprüche 1 bis 3,
    wobei die Auswahl auf eine Sammlung an vorgegebenen zeitlichen Filtern ausgedehnt wird.
  5. Verfahren nach einem der Ansprüche 1 bis 4,
    wobei das hinsichtlich der Frequenz begrenzte Signal angesichts seiner Übermittlung kodiert wird und die Erzeugung des Filters auf Basis des durch eine Entkodierung und spektraler Erweiterung des kodierten, beschränkten Signals erhaltenen Signals sowie des Originalsignals erfolgt.
  6. Verfahren zum Dekodieren eines gesamten oder eines Teils eines Audiosignals, das zumindest die folgenden Schritte umfasst
    - einen Schritt des Empfangens eines übermittelten Signals
    dadurch gekennzeichnet, dass
    das Verfahren weiterhin umfasst
    - einen Schritt des Empfangens eines auf das empfangene Signal bezogenen zeitlichen Filters,
    - einen Schritt des Erhaltens eines dekodierten Signals mittels einer Dekodierung des empfangenen Signals,
    - einen Schritt des Erhaltens eines erweiterten Signals, das durch eine Erweiterung des Spektrums des dekodierten Signals auf den ursprünglichen spektralen Bereich zurückgeführt wurde,
    - einen Schritt des Erhaltens eines rekonstruierten Signals durch Faltung des gesamten erweiterten Signals mit dem erhaltenen zeitlichen Filter.
  7. Verfahren nach Anspruch 6,
    wobei in dem Schritt des Erhaltens eines rekonstruierten Signals anstelle des erzeugten Filters ein ausgehen von dem erzeugten Filter hinsichtlich seiner Größe reduzierter Filter verwendet wird.
  8. Verfahren nach Anspruch 7,
    wobei die Entscheidung der Verwendung eines Filters mit reduzierter Größe anstelle des erzeugten Filters in Abhängigkeit der Kapazitäten des Dekodierers gefällt wird.
  9. Vorrichtung zur Kodierung eines Audiosignals, wenigstens umfassend
    - Mittel zum Erhalten eines hinsichtlich der Frequenz begrenzten Signals, wobei die Reduzierung des Spektrums des Originalsignals durch eine Unterdrückung der hohen Frequenzen erreicht wurde,
    - Mittel zum Erhalten eines kodierten, begrenzten Signals, indem das hinsichtlich der Frequenz begrenzte Signal kodiert wird,
    dadurch gekennzeichnet, dass
    die Vorrichtung weiterhin umfasst
    - Mittel zum Erzeugen eines zeitlichen Filters, der es ermöglicht, ein Signal wieder aufzufinden, das dem Originalsignal nahe kommt, wenn er auf das Gesamtsignal angewendet wird, das durch Dekodierung und Erweiterung des Spektrums des begrenzten Signals wieder auf den ursprünglichen Spektralbereich zurückgeführt wurde.
  10. Vorrichtung zur Dekodierung eines Audiosignals, wenigstens umfassend die folgenden Mittel
    - Mittel zum Empfangen eines übermittelten Signals,
    dadurch gekennzeichnet, dass
    die Vorrichtung weiterhin umfasst
    - Mittel zum Empfangen eines auf das empfangene Signal bezogenen zeitlichen Filters,
    - Mittel zum Erhalten eines dekodierten Signals, indem das empfangene Signal dekodiert wird,
    - Mittel zum Erhalten eines erweiterten Signals, das auf den ursprünglichen Spektralbereich zurückgeführt wurde, indem das Spektrum des dekodierten Signals erweitert wurde,
    - Mittel zum Erhalten eines rekonstruierten Signals, indem das gesamte erweiterte Signal mit dem erhaltenen zeitlichen Filter gefaltet wird.
  11. Signal, umfassend ein hinsichtlich der Frequenz begrenztes, kodiertes Audiosignal, das eine hinsichtlich der Frequenz begrenzte Version eines Originalaudiosignals darstellt,
    dadurch gekennzeichnet, dass
    es weiterhin Daten zur Erzeugung eines zeitlichen Filters umfasst, der die Rekonstruktion eines Signals ermöglicht, das dem Originalsignal nahe kommt, wenn er auf die Gesamtheit einer hinsichtlich der Frequenz erweiterten Version des in dem Signal enthaltenen, hinsichtlich der Frequenz beschränkten Audiosignals, das auf den ursprünglichen Spektralbereich zurückgeführt wurde, angewendet wird.
EP07866270A 2006-12-28 2007-12-27 Tonkodierverfahren und vorrichtung Active EP2126904B1 (de)

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FR0611481A FR2911031B1 (fr) 2006-12-28 2006-12-28 Procede et dispositif de codage audio
PCT/EP2007/011433 WO2008080605A1 (fr) 2006-12-28 2007-12-27 Procede et dispositif de codage audio

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JP (1) JP5491193B2 (de)
AT (1) ATE470933T1 (de)
DE (1) DE602007007125D1 (de)
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FR2911020B1 (fr) 2006-12-28 2009-05-01 Actimagine Soc Par Actions Sim Procede et dispositif de codage audio

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ATE470933T1 (de) 2010-06-15
JP2010515090A (ja) 2010-05-06
FR2911031B1 (fr) 2009-04-10
US20100094640A1 (en) 2010-04-15
JP5491193B2 (ja) 2014-05-14
WO2008080605A1 (fr) 2008-07-10
FR2911031A1 (fr) 2008-07-04
US8595017B2 (en) 2013-11-26
DE602007007125D1 (de) 2010-07-22
EP2126904A1 (de) 2009-12-02

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