EP1979901B1 - Procede et dispositifs pour le codage de signaux audio - Google Patents

Procede et dispositifs pour le codage de signaux audio Download PDF

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Publication number
EP1979901B1
EP1979901B1 EP06706508.6A EP06706508A EP1979901B1 EP 1979901 B1 EP1979901 B1 EP 1979901B1 EP 06706508 A EP06706508 A EP 06706508A EP 1979901 B1 EP1979901 B1 EP 1979901B1
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Prior art keywords
audio signal
audio
fundamental period
signal
subband
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EP1979901A1 (fr
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Martin Gartner
Bernd Geiser
Peter Jax
Stefan Schandl
Herve Taddei
Peter Vary
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Unify GmbH and Co KG
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Unify GmbH and Co KG
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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
    • G10L19/0204—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 subband decomposition
    • G10L19/0208—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
    • 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

Definitions

  • the invention relates to a method and arrangements for audio signal coding.
  • the invention relates to a method and an audio signal decoder for forming an audio signal and an audio signal encoder.
  • Efficient compression of audio signals is also an important consideration in the context of storage or archival of audio signals.
  • Coding methods in which an audio signal synthesized by an audio synthesis filter is timed to an audio signal to be transmitted prove to be particularly efficient is adjusted by optimizing filter parameters. Such a procedure is often referred to as Analysisby Synthesis.
  • the audio synthesis filter is excited by a preferably also to be optimized excitation signal. Filtering is often referred to as formant synthesis.
  • LPC coefficients LPC: Linear Predictive Coding
  • / or parameters specifying a spectral and / or temporal envelope of the audio signal can be used as filter parameters.
  • the optimized filter parameters and parameters specifying the excitation signal are then transmitted to the receiver on a timely basis in order to form a synthetic audio signal there by means of an audio signal decoder provided on the receiver side, which is as similar as possible to the original audio signal with regard to the subjective auditory impression.
  • Such an audio coding method is known from ITU-T Recommendation G.729.
  • a real-time audio signal with a bandwidth of 4 kHz can be reduced to a transmission rate of 8 kbit / s.
  • the achievable transmission bandwidth and audio synthesis quality depend significantly on the generation of a suitable excitation signal.
  • a bandwidth-expanding excitation signal u hb (k) in a high subband can be formed as a spectral copy of the narrow-band excitation signal u nb (k) , (Through the index k, samples of the excitation signal or of other signals are indicated here and below.)
  • the copy can hereby be formed by spectral translation or by spectral reflection of the narrow-band excitation signal u nb (k).
  • spectral translation or reflection distorts the spectrum of the excitation signal inharmonic and / or it causes a significant audible phase error in the spectrum.
  • the document D1 discloses a voice bandwidth extender and a voice bandwidth extension method.
  • the device comprises inter alia a demultiplexer which divides a received signal into multiplexed parameters such as voice information, ie an index.
  • This index includes and outputs a gain code vector, an adaptive codebook delay index, information about a sound source signal and an index of a sound source codevector, and an index of a spectrum parameter.
  • An adder uses a sound source signal which is expanded in the frequency bandwidth, and adds this signal to a signal from the conversion of the reproduction speech signal at a sampling frequency representing a higher frequency component and outputting a voice signal extended by a frequency bandwidth.
  • frequency components of the audio signal attributable to a first subband are formed by means of a subband decoder on the basis of supplied basic period values each indicating a fundamental period of the audio signal.
  • a second subband attributable frequency components of the audio signal are formed by exciting an audio synthesis filter by means of an excitation signal specific for the second subband.
  • a basic period parameter is derived from the basic period values by an excitation signal generator.
  • pulses with a pulse shape dependent on the basic period parameter are formed by the excitation signal generator in a time interval determined by the basic period parameter and mixed with a noise signal.
  • frequency components of the audio signal attributable to a further second subband can be synthesized on the basis of basic period values which have already been made available for a subband decoder specific to the first subband. Since no additional audio parameters are generally required for the generation of the noise signal, the generation of the excitation signal generally requires no additional transmission bandwidth.
  • the audio quality of the audio signal can be considerably improved, in particular since a harmonic harmonic content determined by the basic period values can be reproduced in the second subband.
  • the basic period parameter may specify the fundamental period of the audio signal except for a fraction of a first sampling interval associated with the subband decoder.
  • the pulses can be spaced with a relation to the sub-band decoder higher accuracy, whereby a harmonic spectrum of the audio signal in the second sub-band can be modeled finer.
  • the pulse shape of a respective pulse can be selected from different pulse forms stored in a look-up table, depending on a portion of the fundamental period parameter which is not integral in units of the first sampling interval. From the look-up table, very different pulse shapes can be retrieved in real-time by simple retrieval with low switching, processing or computational effort.
  • the pulse shapes to be stored can be optimized in advance in terms of a lifelike audio playback. In fact, the cumulative effects or the cumulative impulse response of several filters, decimators and / or modulators can be calculated in advance and stored in the look-up table as a correspondingly shaped pulse.
  • a decimator in this context is a converter which multiplies a sampling interval of a signal by a decimation factor m by rejecting all samples except for every m-th sample.
  • a modulator is a filter that multiplies individual samples of a signal by predetermined individual factors and outputs the respective product.
  • the time interval of the pulses can be determined by an integral part of the basic period parameter in units of the first sampling interval.
  • the pulses from a predetermined pulse shape can be formed by samples having a second sampling distance which is smaller by a bandwidth expansion factor than the first sampling distance.
  • the time interval of the pulses can then be in units of the second sample pitch are determined by the basic period parameter multiplied by the bandwidth expansion factor.
  • the bandwidth expansion factor it is preferable to select the inverse N of the fraction 1 / N corresponding to the accuracy of the basic period parameter in units of the first sampling pitch.
  • the pulses may be formed by a pulse shaping filter having filter coefficients set at the second sampling pitch.
  • the pulses can be filtered before or after admixing of the noise signal by at least one high, low and / or bandpass and / or decimated by at least one decimator.
  • the basic period parameter can be derived from one or more basic period values on a timely basis.
  • the basic period parameter can be derived from fluctuation-compensating, preferably non-linear, associated basic period values of several time frames. In this way it can be avoided that fluctuations or jumps in the fundamental period values, e.g. may result from spurious noise measurements of an audio background frequency, adversely affect the basic period parameters.
  • a relative deviation of a current base period value from a previous base period value or from a quantity derived therefrom can be determined and attenuated as part of the derivation of the basic period parameter.
  • a mixing ratio between the pulses and the noise signal by at least one mixing parameter certainly.
  • This time frame can be derived from a subband decoder existing in the level ratio between a tonal and atonal audio signal component of the first subband.
  • an overtone-to-noise ratio in the first subband level parameter related to the formation of the audio signal components in the second sub-band can be used.
  • the level ratio can be converted in such a way that, when the atonal audio signal component predominates, the tonal audio signal component is further lowered. Since, in the case of natural audio sources, an atonal audio signal component in higher frequency bands, in particular from 6 kHz, increasingly prevails, the quality of reproduction can generally be improved by such a reduction.
  • FIG. 1 shows a schematic representation of an audio signal decoder which generates a synthetic audio signal SAS from a supplied data stream of coded audio data AD.
  • the generation of the synthetic audio signal SAS is divided into different subbands.
  • frequency components of the synthetic audio signal SAS attributable to a first, low subband are generated separately from frequency components of the synthetic audio signal SAS attributable to a second, high subband.
  • the deep subband is also referred to below as narrowband.
  • the supplied audio data AD is decoded by a deep subband-specific low-band decoder LBD, ie a decoder having a bandwidth substantially only the low subband.
  • LBD deep subband-specific low-band decoder
  • a synthetic excitation signal u (k) is formed by a high-band excitation signal generator HBG on the basis of the side information g FIX , g LTP and ⁇ LTP extracted by the low-band decoder LBD.
  • the variable k here and in the following denotes an index by which digital samples of the excitation signal or other signals are indexed.
  • an audio signal encoder can also be realized in a simple manner.
  • the synthesized audio signal SAS is to be forwarded to a comparison device (not shown), which compares the synthesized audio signal SAS with an audio signal to be encoded.
  • the synthesized audio signal SAS is then adjusted to the audio signal to be encoded.
  • the invention may be used to advantage for general audio coding, subband audio synthesis and artificial bandwidth expansion of audio signals.
  • the latter can be interpreted as a special case of a subband audio synthesis in which information about a particular subband is used to reconstruct or estimate missing frequency components of another subband.
  • the aforementioned applications are based on a suitably formed excitation signal u (k).
  • the excitation signal u (k) which represents a spectral fine structure of an audio signal, can be converted by the audio synthesis filter ASYN in different ways, eg by shaping its time and / or frequency response.
  • the synthetic excitation signal u (k) is preferably to be generated so that an overtone-to-noise ratio, i. An energy or intensity ratio of the tonal and atonal components of the original audio signal is reproduced as accurately as possible.
  • a broadband noise component is generally added to the harmonics of the audio fundamental frequency F 0 .
  • This noise component is often dominant at higher frequencies, in particular from 6 kHz.
  • the excitation signal u (k) is considered to be on at a predetermined sampling rate of e.g. 16 kHz or 8 kHz sampled subband signal generated.
  • This subband signal u (k) represents the frequency components of the high subband of 4-8 kHz, by which the bandwidth of the narrowband audio signal NAS is to be extended.
  • the narrowband audio signal NAS extends over a frequency range of 0-4 kHz and is sampled at a sampling rate of 8 kHz.
  • the formed excitation signal u (k) excites the audio synthesis filter ASYN and is thereby formed into the high band audio signal HAS.
  • the synthetic broadband audio signal SAS is finally synthesized by combining the shaped high-band audio signal HAS and the narrow-band audio signal NAS with a higher sampling rate of e.g. 16 kHz generated.
  • excitation signal u (k) is based on an audio generation model in which tonal, i. voiced sounds through a sequence of pulses and atonal, i. unvoiced sounds are excited by preferably white noise.
  • tonal i. voiced sounds through a sequence of pulses
  • atonal i. unvoiced sounds are excited by preferably white noise.
  • Various modifications are contemplated to allow for mixed excitation forms that can provide improved hearing.
  • the generation of the tonal components of the excitation signal u (k) is based on two audio parameters of the audio generation model, namely the audio base frequency F 0 and the energy / intensity ratio ⁇ between the tonal and atonal audio components in the deep subband.
  • the latter is often referred to as the overtone-to-noise ratio or "harmonics to Noise Ratio ", HNR for short
  • the audio base frequency F 0 is also called” fundamental speech frequency ".
  • Both audio parameters F 0 and ⁇ can be extracted at the receiver of a transmitted audio signal; preferably (eg in the case of a bandwidth extension) directly from the low frequency band of the audio signal or (eg in the case of subband audio synthesis) from the low band decoder of an underlying lowband audio codec, where such audio parameters are usually available.
  • the audio base frequency F 0 is often represented by a basic period value divided by the sampling rate divided by the audio basic frequency F 0 .
  • the base period value is often referred to as "pitch lag".
  • the base period value is an audio parameter generally transmitted to standard audio codecs, such as the G.729 recommendation, for purposes of so-called “long-term prediction", LTP for short. If such a standard deep subband audio codec is used, the audio base frequency F 0 may be determined or estimated from the LPT audio parameters provided by this audio codec.
  • an LTP fundamental period value is transmitted at a temporal resolution, ie, accuracy that is a fractional 1 / N of the sampling distance used by this audio codec.
  • the basic LTP period value is provided with an accuracy of 1/3 of the sampling distance. In units of this sampling distance, the basic period value can also be accept non-integer values.
  • accuracy can be achieved by the relevant audio encoder, for example, by a sequence of so-called “open-loop” and "closed-loop” searches. The audio encoder attempts to find that fundamental period value at which the intensity or energy of an LTP residual signal is minimized.
  • an LTP basic period value determined in this way can deviate from the basic period value corresponding to the actual audio basic frequency F 0 of the tonal audio components, particularly in the case of strong background noises, and thus impair accurate reproduction of these tonal audio components.
  • Typical deviations include period-doubling errors and period bisecting errors. That is, the frequency corresponding to the departing LPT basic period value is one-half, or twice, the actual audio basic frequency F 0 of the tonal audio components.
  • the function round maps its argument to the nearest integer.
  • a moving average is formed over the fundamental period values ⁇ post ( ⁇ ) for further smoothing.
  • the moving average corresponds to a type of low-pass filtering.
  • tonal mixing parameters g v ( ⁇ ) and atonal mixing parameters g uv ( ⁇ ) for mixing corresponding tonal and atonal components of the excitation signal u (k) in the high subband time-frame from the subband specific mixing parameters g LTP ( ⁇ ) and g FIX ( ⁇ ) of the low-band decoder LBD are derived.
  • the low-band decoder LBD is a so-called CELP decoder (CELP: Codebook Excited Linear Prediction), which has a so-called adaptive or LTP codebook and a so-called fixed codebook.
  • the intensity ratio between tonal and atonal signal components can be reconstructed from the mixing parameters LTP g and g FIX of the low-band decoder LBD.
  • Both mixing parameters g LTP , g FIX can be extracted on a timely basis from the low-band decoder LBD.
  • an instantaneous intensity ratio between the contributions of the adaptive and fixed codebooks ie the overtone-to-noise ratio ⁇ , can be determined by dividing the energy contributions of the adaptive and fixed codebooks.
  • the mixing parameter g LTP ( ⁇ ) indicates a gain for the adaptive codebook signals
  • the mixing parameter g FIX ( ⁇ ) indicates a gain for the fixed codebook signals.
  • This "Wiener” filtering further lowers a small ⁇ (atonal audio segment) while barely changing large values of ⁇ (tonally dominated audio segment). Such a reduction better approximates natural audio signals.
  • both mixing parameters g v ( ⁇ ) and g uv ( ⁇ ) generally have (at the same time) a non-vanishing one Value.
  • the above calculation rule ensures that the sum of the squares of the mixing parameters g v and g uv , ie a total energy of the mixed excitation signal u (k), is substantially constant.
  • the generation of the excitation signal u (k) is explained in more detail below on the basis of the audio parameters g v , g uv and ⁇ p derived from the low-band decoder LBD using the example of two embodiment variants of the excitation signal generator HBG.
  • the following statements are of course readily generalizable to any values of N.
  • a first embodiment of the excitation signal generator HBG is in FIG. 2 shown schematically.
  • the noise generator NOISE preferably generates white noise.
  • the pulse generator PG1 in turn comprises a rectangular pulse generator SPG and a pulse shaping filter SF with a predetermined filter coefficient set p (k) of finite length. While the noise generator NOISE serves to generate the atonal components of the excitation signal u (k), the pulse generator PG1 contributes to the generation of the tonal components of the excitation signal u (k).
  • the audio parameters g v , g uv and ⁇ p are time-frame continuous in audio sequence of the low-band decoder LBD or derived and adapted by means of a suitable audio parameter extraction block.
  • the filter operations are designed for a fractional basic period parameter ⁇ p with an accuracy of 1 / (2N), equal to 1/6, in units of the sampling rate of the low-band decoder LBD and for a target bandwidth corresponding to the bandwidth of the low-band decoder LBD ,
  • Pulse generator PG1 provide a sampling rate of at least 8 kHz.
  • the rectangular pulse generator SPG generates individual rectangular pulses in a time interval given by 6 * ⁇ p in units of the sampling interval 1/48000 s of the rectangular pulse generator SPG.
  • the individual rule pulses have an amplitude of 6 * ⁇ p . such that the average energy of a long pulse sequence is substantially constant equal to 1.
  • the rectangular pulses generated by the rectangular pulse generator SPG are multiplied by the "tonal" mixing parameter g v and fed to the pulse shaping filter SF.
  • the rectangular pulses are smeared to a certain extent by folding or correlation with the filter coefficients p (k) .
  • This filtering can considerably reduce the so-called crest factor, ie a ratio of peak to average samples, and considerably improve the audio quality of the synthesized audio signal SAS
  • the rectangular pulses may advantageously be spectrally shaped by the pulse shaping filter SF
  • the pulse shaping filter SF may have a bandpass characteristic with a transition region around 4 kHz and a substantially uniform increase in attenuation in the direction of higher and lower frequencies can be achieved that higher frequencies of the excitation signal u (k) have less harmonic components and thus the noise component increases with increasing frequency.
  • FIGS. 3a and 3b An exemplary choice of the filter coefficients p (k) is in the FIGS. 3a and 3b shown schematically. While FIG. 3a the filter coefficient p (k) plotted against its sample index k is in FIG. 3b the energy spectrum of the filter coefficients p (k) plotted against the frequency. For the target frequency range relevant in the present exemplary embodiment, essentially only the spectral range of 4-8 kHz is relevant for the filter coefficients p (k). This frequency range is in FIG. 3b indicated by a widened line.
  • the rectangular pulses "blurred" by the pulse shaping filter SF are added to a noise signal generated by the noise generator NOISE and multiplied by the "atonal" mixing parameter g uv , and the resultant sum signal is fed to the low-pass filter LP.
  • the generated excitation signal u (k) contains the frequency components required for bandwidth expansion. However, these are available as spectrum mirrored around the 4 kHz frequency. In order to invert the spectrum, the excitation signal u (k) can be modulated with modulation factors (-1) k .
  • the tonal and atonal components of the excitation signal u (k) can be treated independently of each other.
  • filtering and decimation operations for the tonal audio components are also summarized in a single processing block.
  • the impulse response of all in FIG. 2 provided filtering, decimation and modulation operations for the tonal audio components are calculated in advance and stored in a look-up table in a suitable form.
  • FIG. 4 Such a configured, second embodiment of the excitation signal generator HBG is in FIG. 4 schematically illustrated and will be explained below.
  • the pulse generator PG2 in turn comprises a pulse positioner PP and a look-up table LOOKUP, in which predetermined pulse shapes v j (k) are stored.
  • the noise generator NOISE serves to generate the atonal components of the excitation signal u (k)
  • the pulse generator PG2 contributes to the generation of the tonal components of the excitation signal u (k).
  • the derivation of the audio parameters g v , g uv and ⁇ p has already been explained above.
  • the fractional basic period parameter ⁇ p is given as above with an accuracy of 1 / (2N), here equal to 1/6, in units of the sampling rate of the low-band decoder LBD.
  • the impulse response of all can FIG. 2 Filtering, decimation and modulation operations are calculated in advance and stored in the form of certain pulse shapes v j (k) in the look-up table LOOKUP.
  • several pulse shapes v j (k) in the lookup table LOOKUP are kept.
  • the number of pulse shapes v j (k) to be provided is preferably given by the inverse of the accuracy of the basic period parameter ⁇ p , ie here by 2N.
  • the index j runs from 0 to 2N-1.
  • the look-up table LOOKUP is supplied with the fractional component ⁇ p - ⁇ ⁇ p ⁇ of the respective basic period parameter ⁇ p .
  • the parenthesis ⁇ ⁇ denotes an integer part of a rational or real number.
  • a pulse shape is selected from the stored pulse shapes v j (k) and a correspondingly shaped pulse is output from the look-up table LOOKUP.
  • ⁇ p - ⁇ ⁇ p ⁇ can assume the values 0, 1/6, 2/6, 3/6, 4/6 and 5/6.
  • that impulse form v j (k) is selected whose index j corresponds to the respective counter of the relevant fraction.
  • Each of the stored pulse forms v j (k) corresponds to an impulse response of the in FIG. 2 shown chain from the filters SF, LP, D3, HP and D2 (and optionally a modulator) for a particular fractional fraction ⁇ p - ⁇ ⁇ p ⁇ the basic period parameter ⁇ p .
  • the illustrated pulse shapes v j (k) are for a fractional resolution of ⁇ p of 1/6 (at a sampling rate of 8 kHz) and plotted against its scan index k.
  • An assignment of a respective pulse shape v j (k) to the associated fractional fraction ⁇ p - ⁇ ⁇ p ⁇ is the legend of FIG. 5 refer to.
  • the pulse output by the LOOKUP look-up table which has a pulse shape selected on the basis of the fractional fraction ⁇ p- ⁇ ⁇ p ⁇ , is multiplied by the "tonal" mixing parameter g v and fed to the pulse positioner PP.
  • the supplied pulses are temporally positioned depending on the integer part ⁇ ⁇ p ⁇ of the basic period parameter ⁇ p .
  • the pulses are output by the pulse positioning device PP at a time interval which corresponds to the integer component ⁇ ⁇ p ⁇ of the basic period parameter ⁇ p.
  • the pulses may be modulated by a respective sign of the pulse shapes v j (k) or of the respective pulses, either for even values of ⁇ ⁇ p ⁇ or for odd values of ⁇ ⁇ is inverted p ⁇ .
  • the noise signal of the noise generator NOISE multiplied by the "atonal" mixture parameter g uv is finally added to obtain the excitation signal u (k).
  • FIG. 4 illustrated embodiment can be generally with less effort than in FIG. 2 implement illustrated embodiment variant.
  • an excitation signal generator according to FIG. 4 by specifying suitable pulse shapes v j (k) effectively the same excitation signals u (k) as with an excitation signal generator according to FIG. 2 to generate. Since the output pulses have a relatively large distance (typically 20-134 sampling distances), the computational effort for an inventive excitation signal generator according to FIG. 4 relatively low.
  • the invention can be implemented by means of a low-cost digital signal processor with relatively low memory and computing power requirements.

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Claims (15)

  1. Procédé de formation d'un signal audio (SAS) à partir d'un flux de données audio (AD) codées, avec lequel
    a) des composantes fréquentielles (NAS) du signal audio incluses dans une première sous-bande basse sont formées au moyen d'un décodeur de sous-bande (LBD) à l'aide de valeurs de période fondamentale (λLTP) contenues dans les données audio (AD) et indiquant chacune une période fondamentale du signal audio (SAS),
    b) des composantes fréquentielles (HAS) du signal audio incluses dans une seconde sous-bande haute sont formées par excitation d'un filtre de synthèse audio (ASYN) au moyen d'un signal d'excitation (u(k)) spécifique à la seconde sous-bande, et
    c) pour la génération du signal d'excitation (u(k)) par un générateur de signal d'excitation (HBG)
    - un paramètre de période fondamentale (λp) est déduit des valeurs de période fondamentale (λLTP) ainsi que
    - des impulsions ayant une forme d'impulsion dépendante du paramètre de période fondamentale (λp) sont formées dans un intervalle de temps défini par le paramètre de période fondamentale (λp) et mélangées à un signal de bruit.
  2. Procédé selon la revendication 1,
    caractérisé
    en ce qu'un premier intervalle d'échantillonnage spécifique à la première sous-bande est attribué au décodeur de sous-bande (LBD) et
    en ce que le paramètre de période fondamentale (λp) indique la période fondamentale du signal audio (SAS) jusqu'à une fraction du premier intervalle d'échantillonnage.
  3. Procédé selon la revendication 2,
    caractérisé en ce que
    la forme d'impulsion (vj(k)) d'une impulsion respective est sélectionnée en fonction d'une partie non entière (λp - └λp┘) du paramètre de période fondamentale (λp) selon des unités du premier intervalle d'échantillonnage parmi différentes formes d'impulsion (vj(k)) prédéfinies, enregistrées dans une table de consultation.
  4. Procédé selon la revendication 2 ou 3,
    caractérisé en ce que l'intervalle de temps des impulsions est défini par une partie entière (└λp┘) du paramètre de période fondamentale (λp) selon des unités du premier intervalle d'échantillonnage.
  5. Procédé selon la revendication 2 ou 3,
    caractérisé
    en ce que les impulsions sont formées, à partir d'une forme d'impulsion prédéfinie, par des valeurs d'échantillonnage présentant un second intervalle d'échantillonnage, le second intervalle d'échantillonnage étant plus petit d'un facteur d'extension de largeur de bande (N) que le premier intervalle d'échantillonnage, et
    en ce que l'intervalle de temps des impulsions est défini selon des unités du second intervalle d'échantillonnage par le paramètre de période fondamentale (λp) multiplié par le facteur d'extension de largeur de bande (N).
  6. Procédé selon la revendication 5,
    caractérisé en ce que
    les impulsions sont formées par un filtre de formage d'impulsions (SF) ayant des coefficients de filtre (p(k)) prédéfinis dans le second intervalle d'échantillonnage.
  7. Procédé selon la revendication 5 ou 6,
    caractérisé en ce que,
    avant ou après le mélange du signal de bruit, les impulsions sont soumises à une décimation par au moins un décimateur (D2, D3).
  8. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que
    les impulsions sont filtrées, avant ou après le mélange du signal de bruit, par au moins un filtre passe-haut, passe-bas ou passe-bande.
  9. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que
    le paramètre de période fondamentale (λp) est déduit, selon des laps de temps, d'après une ou plusieurs valeurs de période fondamentale (λLTP).
  10. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que
    le paramètre de période fondamentale (λp) est déduit d'après des valeurs de période fondamentale (λLTP) de plusieurs laps de temps combinées pour compenser des variations.
  11. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que
    une différence relative (e) d'une valeur de période fondamentale (λLTP) réelle à une valeur de période fondamentale antérieure ou à une grandeur (λpost) déduite de celle-ci est déterminée et est atténuée dans le cadre de la déduction du paramètre de période fondamentale (λp).
  12. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que
    un taux de mélange entre les impulsions et le signal de bruit est défini par au moins un paramètre de mélange (gv, guv) qui est déduit, selon des laps de temps, d'un rapport de niveau (γ), présent dans le décodeur de sous-bande (LBD), entre une partie tonale et une partie atonale du signal audio de la première sous-bande.
  13. Procédé selon la revendication 12,
    caractérisé en ce que,
    dans le cadre de la déduction du paramètre de mélange (gv, guv), le rapport de niveau (γ) est déplacé de manière telle que, en cas de prépondérance de la partie atonale du signal audio, la partie tonale du signal audio soit réduite.
  14. Décodeur de signaux audio pour la formation d'un signal audio (SAS) à partir d'un flux de données audio (AD) codées, comportant
    a) un décodeur de sous-bande (LBD) pour la formation des composantes fréquentielles (NAS) du signal audio incluses dans une première sous-bande basse à l'aide de valeurs de période fondamentale (λLTP) contenues dans les données audio (AD) et indiquant chacune une période fondamentale du signal audio (SAS),
    b) un filtre de synthèse audio (ASYN) et
    c) un générateur de signal d'excitation (HBG) destiné à la génération d'un signal d'excitation (u(k)) pour la formation de composantes fréquentielles (HAS) du signal audio comprises dans une seconde sous-bande haute par l'excitation du filtre de synthèse audio, le générateur de signal d'excitation (HBG) comportant :
    - un dispositif de déduction pour la déduction d'un paramètre de période fondamentale (λp) à partir des valeurs de période fondamentale (λLTP)
    - un générateur de bruit (NOISE) pour la formation d'un signal de bruit,
    - un générateur d'impulsions (PG1, PG2) pour la formation d'impulsions ayant une forme d'impulsion dépendante du paramètre de période fondamentale (λp) dans un intervalle de temps défini par le paramètre de période fondamentale (λp), ainsi que
    - un système de mélange pour le mélange des impulsions avec le signal de bruit.
  15. Codeur de signaux audio avec un décodeur de signaux audio selon la revendication 14 ainsi qu'avec un système de comparaison pour l'harmonisation d'un signal audio synthétisé (SAS) par le décodeur de signaux audio à un signal audio à coder, le signal audio synthétisé (SAS) étant harmonisé au signal audio à coder par une variation des données audio (AD), en particulier des informations secondaires (GFIX, gLTP, λLTP) contenues dans celles-ci.
EP06706508.6A 2006-01-31 2006-01-31 Procede et dispositifs pour le codage de signaux audio Active EP1979901B1 (fr)

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EP1979901A1 (fr) 2008-10-15
WO2007087824A1 (fr) 2007-08-09
US20090024399A1 (en) 2009-01-22
CN101336451A (zh) 2008-12-31
US8612216B2 (en) 2013-12-17

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