EP1230827B1 - Procede et dispositif pour traiter un signal audio stereo - Google Patents

Procede et dispositif pour traiter un signal audio stereo Download PDF

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Publication number
EP1230827B1
EP1230827B1 EP00985148A EP00985148A EP1230827B1 EP 1230827 B1 EP1230827 B1 EP 1230827B1 EP 00985148 A EP00985148 A EP 00985148A EP 00985148 A EP00985148 A EP 00985148A EP 1230827 B1 EP1230827 B1 EP 1230827B1
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Prior art keywords
channel
signal
modified
audio signal
sum
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English (en)
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EP1230827A2 (fr
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Bodo Teichmann
Oliver Kunz
Jürgen HERRE
Klaus Peichl
Michael Beer
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • 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

Definitions

  • the present invention relates generally to coding of audio signals and in particular processing of stereo signals.
  • a stereo signal comprises at least two channels, i.e. H. one left channel and a right channel.
  • a stereo signal has five different channels, d. H. a front left Channel, a front center channel and a front right channel and a left rear and a rear right channel.
  • M / S method A known method for processing stereo signals in order to achieve more efficient coding is referred to as the center / side method (M / S method).
  • the M / S method combines the first and second channels to create a center channel and a side channel.
  • L channel left channel
  • R channel right channel
  • the center channel is equal to the sum of the left channel L and the right channel R multiplied by a factor of 0.5
  • the side channel is the difference between the left channel L and the right channel R. , multiplied by a factor of z. B. 0.5 (other factors are also possible).
  • M 0.5 ⁇ (L + R)
  • S 0.5 ⁇ (L - R)
  • a listener will see the similarity of the left and right channels perceive that, in the case of identical channels, a speaker or orchestra right in the middle between is perceived by the two speakers. on the other hand a listener will perceive dissimilar channels in that he has a pronounced stereo effect, d. H. that a speaker, an orchestra or individual instruments of an orchestra can be located exactly to the left and / or right can.
  • a listener will perceive dissimilar channels in that he has a pronounced stereo effect, d. H. that a speaker, an orchestra or individual instruments of an orchestra can be located exactly to the left and / or right can.
  • the side channel will also be about the same as the left Channel.
  • the coding required Bit size not reduced due to the M / S coding, but in the borderline case it even doubles, if of that it is assumed that the left channel L a certain amount of energy comprises, while the right channel R is 0.
  • the left channel L a certain amount of energy comprises, while the right channel R is 0.
  • no M / S processing but only L / R processing is therefore sufficient in one Extreme case from a saving of 50% to another Extreme case, which requires a doubling of the coding Bits.
  • an audio signal for example is in the form of PCM samples as they are z.
  • B. outputs a CD player by means of a time-frequency transformation or a filter bank in a spectral representation transferred.
  • a block with a certain number of samples also called a "frame” is used to make a block of complex Generate spectral values that a short-term spectrum of Form frames of audio samples.
  • Block formation is made using transformation windows reached, which are, for example, 1024 samples long.
  • a mid / side processing can either be done before the transformation be performed in the spectral range, d. H. using the digital time discrete samples.
  • center / side processing can also be used after the transformation, d. H. with the complex spectral values be performed.
  • the latter alternative offers furthermore the advantage that a middle / side processing not, as in the time domain, used for the entire spectrum but also for certain frequency bands, if certain spectral values of a middle / side processing undergo and others do not.
  • Audio coders are usually designed in such a way that they a constant bit rate, i.e. H. a certain number of bits per second.
  • the quantization noise introduced by quantization if possible, be chosen such that its Energy below the psychoacoustic masking threshold or Monitoring threshold of the audio signal is.
  • the basic process, the quantization noise in the frequency domain to adjust is to "shape" the noise below Use of scale factors.
  • the spectrum into several groups of spectral coefficients divided, which are called scale factor bands, to which a single scale factor is assigned.
  • On Scale factor represents a multiplication value that uses is the amplitude of all spectral coefficients in to change this scale factor band.
  • That mechanism is used to map the quantization noise in the spectral range that is generated by the quantizer, so that the energy in each scale factor band of quantization noise under the psychoacoustic Masking threshold in this scale factor band. It is it can be seen that neither quantization nor entropy coding Are processes that favor a constant bit rate. It should be noted that - on the contrary - both Process favor a variable bit rate. For transmission applications however, it is often required that the Encoder has a constant bit rate at the output. To one Delivering a constant bit rate is usually a so-called Bit reservoir used.
  • bits are assigned to the bit reservoir in order to In the case of an audio signal section that has more bits for coding needed to be able to give more bits, so that Bit reservoir is emptied again.
  • the "inner bit rate" of the encoder is higher than that required on the output side constant bit rate. This case will occur if that Audio signal is difficult to encode, i. H. if the encoder have to spend many bits to encode the audio signal, which is also referred to as the "high load” of the encoder can be.
  • the notion that it encodes tonal pieces relatively efficiently can, however, be noisy signals that are relative have high energies, and also a relative one have a complex spectrum, such as language or drum or drum music, relatively little compressed can be.
  • signals that are transient, i. H. who have an irregular timing can only be relative be encoded at great expense if there are no coding artifacts should be generated. In the case of transient signals already when windowing from long windows to shorter ones Window switched to better temporal resolution too reach, or to achieve that the quantization noise only over a smaller number of audio samples "Smeared". In the case of short windows fall much more page information.
  • An encoder that determines that the output bit rate is not is enough, and which has already “emptied” the bit reservoir has now several options to its internal bit rate “violent” to reduce the criterion of constant Output bit rate to meet.
  • One way is to avoid switching to short windows. This however leads to audible coding artifacts.
  • Another option is the audio bandwidth to decrease, d. H. no longer the full audio bandwidth encode but from a certain of the output bit rate dependent spectral frequency the overlying spectral values to be set to 0 in order to reduce the output bit rate.
  • This method does not lead to audible quantization disturbances, however, leads to a loss of highs in the audio signal. However, this loss is often perceived less strongly as an audible quantization noise.
  • a particular problem when encoding stereo signals consists of the effect called "stereo unmasking", which is briefly outlined below.
  • stereo unmasking a normal L / R coding is used, so both the left channel and the right channel itself also transformed, quantized and encoded so that in the left channel and right channel introduced quantization noise independently for data reduction from the other channel. That means that Left channel quantization noise and quantization noise are not correlated in the right channel.
  • the "stereo unmasking" effect is that because of the fact that the quantization noise in the two channels is not correlated, the quantization noise of the left Left channel and the quantization noise of the right channel is perceived on the right. A high masking of the noise but only takes place in the middle, where that too Useful signal is, but not left and right.
  • the M / S coding therefore has, besides its data rate reducing Effect with special signals also the advantage that the quantization noise in both the left channel as well in the right channel with the quantization noise of the other channel is correlated, so that the quantization noise takes place in the middle and there of that Useful signal essentially completely or much better than is covered up in the uncorrelated case. It is different Case where the left and right channels are relatively dissimilar are. If M / S coding is used here, then due to the stereo effect, the useful signal is either on the left or be right, while due to the M / S coding the Quantization noise is correlated and more in the middle lies. Stereo unmasking also takes place here, so to speak instead of.
  • Scalable audio encoders are arranged that its output bitstream has at least a first one and has a second scaling layer.
  • a decoder which is simply designed, is made from the scaled bit stream just take the first scaling layer, for example an encoded audio signal with reduced bandwidth has or a with a simple coding algorithm encoded audio signal.
  • Another decoder the is fully designed, both the first scaling layer as well as take the second scaling layer out of the bit stream, around the first scaling layer with a first decoder to decode, and then the second scaling layer also decode that alone or together an audio signal with the decoded first scaling layer with full bandwidth.
  • Scalable encoders are particularly useful in the area of stereo signals desirable, because here as the first scaling layer Mono signal, i.e. H. the middle channel, can be used while as the second scaling layer z. B. the side channel can be taken.
  • a simple decoder or a decoder, which is designed for fast operation is only deliver the mono signal while a better decoder or a decoder, in which the speed of the Transfer is not the most crucial criterion, besides take the side layer from the mono or middle layer to a full stereo signal at the output of the decoder to create.
  • the first scaling layer can from the second scaling layer or from any one Number of further scaling layers in the audio coding process itself, in the audio bandwidth, in the audio quality, regarding Mono / Stereo and or a combination of the above Distinguish quality criteria or other conceivable criteria.
  • the aim is that the second scaling layer has the smallest possible number has bits, or that a decoder that the decoded second scaling layer, as extensively as possible also uses the first scaling layer. If a scalable Encoder for stereo signals is considered, the as the first scaling layer, the middle signal, i. H.
  • the Mono signal and the second layer is the side channel supplies
  • the M / S coding is used, the better.
  • this requirement applies to certain stereo signals contrary to bit efficiency, namely with stereo signals, that have high stereo channel separation.
  • the object of the present invention is a Device and method for processing a stereo audio signal to create that to less audible interference leads.
  • This task is accomplished by a processing device a stereo audio signal according to claim 1 and by a method of processing a stereo audio signal after Claim 18 solved.
  • the present invention is based on the finding that that it is often cheaper for stereo audio signals a high stereo channel separation to avoid a higher one Audio bandwidth and / or less audible interference in comparison to achieve the case where the stereo channel separation is maintained while the audio bandwidth is reduced or interference introduced by quantization become audible.
  • Audible quantization disorders are generally a foreign body in an audio signal while a Receiver of a stereo signal processed according to the invention does not necessarily know how the stereo channel separation of the Output signal was and thus a lower stereo channel separation is not perceived as a coding artifact.
  • a reduction in stereo channel separation is thus used the encoder bit rate in general decrease, or reduce to a predetermined value.
  • a device for processing a stereo signal, that a first channel and a second channel comprises a device for analyzing the stereo audio signal, to get a measure of a lot of bits which is needed by an encoder to get the stereo audio signal to encode using an encoding algorithm and means for modifying the first and of the second channel to a modified first and one to obtain modified second channel, the device to modify to the device for analysis responds to be effective when the measure of the amount of Bits exceed a predetermined amount, and being set up is designed to modify such that a Sum signal from the first and second modified channel at least according to a characteristic of the signal that is similar to the energy of the signal changes, essentially equal to the characteristic of a sum signal from the first and second channel, and that a difference signal from the first and second modified channels compared to Difference signal attenuated from the first and second channels is.
  • the characteristic is similar runs to energy, which can be energy itself, but also z.
  • energy is a characteristic that is similar to energy runs, spoken.
  • Modifying the stereo audio signal i.e. H. reducing the channel separation
  • the first and the second channel e.g. B. the left channel and the right channel, modified such that the volume, i.e. H. the sum signal, compared to the unmodified first and second channels at least in terms of energy and preferably even in terms of signal remains essentially the same while the difference signal is subdued.
  • the preprocessing of the stereo signal according to the invention is always use when it is determined that the quantity bits needed to encode the stereo audio signal gets too high.
  • the measure of the amount of bits used for Coding of the stereo audio signal may be needed the stereo audio signal by analyzing it for various Ways are derived.
  • the center and side channels of the stereo audio signal to be considered due to an energy ratio or a difference in the logarithms of the Energys to determine how many bits are needed. Without having to determine the exact number of bits, it is reasonable to conclude that in the case of a small Energy ratio between the center and side channel, d. H. in the case of channels of approximately the same size, a high number of bits will be needed. The lower the energy ratio between the middle and the side channel, so more attenuation of the side channel will be necessary to get one to achieve certain output bit rate.
  • a small energy ratio lies between the middle and the side channel before when the original audio signal has a high stereo channel separation has, for example if the left channel a lot Has energy while the right channel is essentially noise Has.
  • the amount of bits is independent of the nature the middle channel and the side channel in it To look at the encoder itself.
  • a measure of that of one The number of bits required by the encoder is the so-called perceptual Entropy (PE), which is equal to the energy ratio between the useful audio signal and that calculated for the useful audio signal psychoacoustic listening threshold is. If the PE is big, can be concluded that the audio signal is a relative has low concealability. On the other hand, if the PE is small, d. H. the energy of the useful signal is only slightly above that psychoacoustic listening threshold, so the useful signal only be quantized relatively roughly, and the quantization noise is still below the psychoacoustic listening threshold "hidden".
  • the sum of the PE, preferably averaged over a certain time of the left channel and that, also preferably over one averaged over time, PE for the right channel over is a predetermined value, the Side channel attenuated to the required number of bits to reduce.
  • This alternative aspect of the present Invention is therefore not concerned with the individual Appearance of the center and side channels, but with the Stereo audio signal itself, which is not regarding its M / S coding ability is assessed, but its general Audio coding capability, d. H. the difficulty of doing the same encode to achieve a specific target bit rate.
  • a generalization of the second aspect is some other size as a measure of the amount of bits too use, which indicates the "load" of the encoder.
  • a variable can also be a signal, for example, which indicates due to transient properties of the audio signal, that an audio encoder use short windows for windows must, since the fact is that short windows are not most recently due to the increased number of page information require a higher bit rate.
  • the present invention can thus cover the entire range of Control variables of an audio encoder are used to get a Measure to find that or how much the side channel is damped must be set to the encoder output bit rate reduce.
  • Preferred embodiments of the present invention lead an increasing or decreasing in time Attenuation of the side channel through to prevent a Listener immediately perceives the decreasing stereo channel separation, but that the reduction in stereo channel separation gradually occurs or the stereo channel separation increases gradually increases to the encoder side To mask manipulation of the stereo audio signal as well as possible.
  • Fig. 1 shows a block diagram of the device according to the invention to process a stereo audio signal that is on an input 10 is fed into the device and has a first channel L and a second channel R.
  • the Stereo audio signal in the form of the first channel L and the second Channel R is on the one hand in a device 12 for analysis of the stereo audio signal, and on the other hand also in a device 14 for modifying the first and second channels fed to at an output 16 shows a modified first channel L 'and a modified one to obtain second channel R '.
  • the modified first channel L 'and the modified second Channel R 'at output 16 from the unmodified first channel L and from the unmodified second channel R at input 10 differ in that the applied to output 16 modified Stereo audio signal has a lower channel separation than have the unmodified stereo audio signal at input 10 becomes.
  • the device 12 for analyzing the stereo audio signal determines a measure of an amount of bits that are in one Fig. 1 encoder, not shown, is required to the stereo audio signal using one by the encoder to code the given coding algorithm.
  • the measure of that Bit amount is determined by the device 12 for analysis a signal path 18 of the device 14 for modification fed.
  • the modification of the first and second channels becomes like this performed that the energy of the sum of the modified Stereo audio signal at output 16 in a predetermined ratio and preferably substantially equal to the energy of the unmodified stereo audio signal at input 10 is, however, the difference signal, which apart from the factor of z. B. 0.5 corresponds to the side channel, in modified stereo audio signal at output 16 compared to unmodified stereo audio signal attenuated at input 10 is.
  • Fig. 1 are two ways of feeding the device 12 shown for analysis, individually or in Combination can be used.
  • the first option is represented by a left arrow 15a, so to speak represents a feed forward, d. H.
  • the establishment to analyze the stereo audio signal is not with the modified signal L, R fed.
  • the other possibility consists of the device 12 for analysis with the modified signal L ', R' to feed.
  • the damping is dependent from the current unmodified signal or from a the last processing blocks of the modified signal is controlled to a certain extent in terms of feedback. So that is it doesn't matter whether the stereo audio signal itself is direct is analyzed, or indirectly based on a previous one modified signal.
  • the Means 12 for analyzing the unmodified stereo audio signal received at entrance 10 forms and then the relationship of the energies of the middle and of the side channel viewed.
  • the energy ratio between the middle and the side channel is preferably over a certain time averaged, for example in the order of magnitude of 10 audio frames, which is a value of Corresponds to 200 ms if an MPEG-2 AAC encoder is used as the audio encoder is used, which has a frame length of about 20 ms can have.
  • the MPEG-2 AAC encoder is on the standard ISO / IEC 13818-7, in which the individual Function blocks of an audio encoder and an audio decoder and their interaction is described in detail are.
  • the energy ratio or the difference the logarithms smaller than a certain one depending on the Use case is the value to be determined empirically, for example can be chosen to 6 dB, the facility 14 activated for modification in order to dampen the Side channel as it is referring to FIG. 2 will be run in more detail.
  • the inventive device for processing of the stereo audio signal only then the side channel attenuate when the signal is no longer so good MS coding ability because, for example, both channels either energy and / or signal dissimilar to each other are.
  • stereo channel separation becomes whenever reduced to keeping the original Stereo channel separation at too high an output bit rate would result, and if stereo channel separation at all was high.
  • the attenuation of the side channel to reduce the output side Encoder bit rate used regardless of whether the stereo audio signal has a certain MS coding capability or not proceeds from this from that even in the case of a small stereo channel separation still further attenuation of the side channel achieved can be set to a predetermined output bit rate of the audio encoder not to exceed. This is independent of the MS coding capability of the audio signal the number of bits estimated that is required to encode the audio signal.
  • the energy ratio or the difference of the logarithms of the Audio signal itself and its psychoacoustic masking threshold, also called Perceptual Entropy (PE) provides a measure of how many bits are to be encoded of the audio signal are required. If the PE is high, so many bits are required because the masking ability of the Audio signal is relatively low and therefore finely quantized must become. On the other hand, if the PE is small, relatively few will be Bits because the audio signal is masked relatively well, and therefore only a relatively rough quantization is required is.
  • the second Aspect of the present invention the measure of the amount of bits determined as follows.
  • the PE values for the individual Scale factor bands are integrated over frequency, d. H. summed up. This will be for both the left performed for right channel as well. Then will the PE sum for the left channel to the PE sum for the right channel summed.
  • This sum PE value from left and the right channel represents the bit requirement for a frame Sum channel PE value is then preferably still above a certain number of frames, such as B. 10, averaged, by an averaged PE value for the stereo audio signal receive. If this averaged PE value is greater than or equal to a predetermined one that is typically to be determined empirically Value, the multiplication facility is activated, to dampen the side channel.
  • any encoder will need, any other control variable be used, which is a measure of the "load" of the encoder represents such.
  • the means 14 for modifying can be interpreted as having a first input 20a for the first channel L and a second input 20b for the second channel R.
  • the device 14 comprises a first multiplier 22a for multiplying the first channel L with a certain factor x, a second Multiplier 22b for multiplying the first channel L by a factor y, a third multiplier for multiplying of the second channel R with the factor x and finally a fourth multiplier 22d for multiplying the second channel R with the factor y.
  • At exit 26a of the first Summierers 24a is the modified first Channel L 'on, and at the output 26b of the second adder 24b is the modified second channel R '.
  • Equation (6) and equation (9) result in equation (10) for x and equation (11) for y.
  • x 0.5 * (1 + exp (0.05 att))
  • y 0.5 * (1 - exp (0.05 att))
  • Attenuation "att" (in dB) is dependent on one of the control variables described. This results in with equations (9) and (10) the factors x and y for the damping matrix represented by FIG is equally reflected in equations (1) and (2).
  • Attenuation value att which has been determined empirically, is used if the measure of the amount of bits is one exceeds the predetermined limit.
  • the damping is not increased suddenly, because a decrease in channel separation that occurs suddenly goes to an audible disturbance or could surprise the listener, for example if a speaker was initially placed on the left and up is perceived once in the middle. Therefore, in the case where which is determined to dampen the side channel, a gradual damping of the side channel, for example using a predetermined increment value, such that vividly speaking the news anchor slowly "wanders" from the left to the center. In the opposite case, it is found that the Measure of the amount of bits again smaller than the predetermined The damping is not abruptly canceled, but slowly returned to 0, so that um in the example, the speaker slowly stays away from the Center to the side "wanders".
  • the device 12 for analyzing it is preferred additionally in the device 12 for analyzing, whether the phase shift of L and R is close of 180 degrees. If this is found, it can be done easily the sign of R can be reversed. Then it goes originally wanted spatial stereo effect lost, however the effect of reduced volume is avoided what will bother a listener less.
  • the M channel could also be in the device for modification or in a downstream Encoder level increased to a certain value such that the energy of the modified M-channel in a predetermined ratio to the energy of the M-channel of the unmodified stereo audio signal.
  • a value of 1 is preferred, but also a certain gain by the modifier means or damping can be performed, but always that Relationship to the unmodified stereo audio signal essentially should be maintained so that a listener does not significant volume fluctuations due to preprocessing will perceive.
  • small ones Volume fluctuations are not so problematic and sometimes even imperceptible. Large volume fluctuations however, a test listener will find it annoying.
  • the processing device according to the invention of a stereo audio signal could therefore also be Time-frequency transformation level of a time / frequency transformation-based Be arranged encoder, such as. B. an MPEG audio encoder. This concept even results the additional possibility of stereo preprocessing can be made frequency-selective, d. H.
  • FIG. 3 a Device for processing a stereo audio signal shown which is in addition to the puncture blocks shown in FIG also an MS encoder 30 and a scalable Encoder 32 includes a scaled output Bitstream BS outputs.
  • the MS encoder 30 includes how it is known in the art to have a summer 30a for summing of the modified left channel L 'and the modified right channel R 'to multiply by a multiplier 30b to which a factor of e.g. B. 0.5 is assigned to generate the multiplied center channel.
  • the MS encoder 30 includes a subtractor 30c and a further multiplier 30d to the modified To generate side channel S 'which is opposite a side signal, that from the unmodified stereo audio signal is formed at the input 10, is damped.
  • the middle channel M ' and the side channel S ' are both scalable Encoder 32 is fed, which preferably has a mono stereo scalability having.
  • the first scaling layer is represent the mono signal M ', and the second scaling layer will include the modified side channel S '.
  • Further Scaling options such as B. that the modified or unmodified mono channel M 'additionally band limited and that in the second scaling layer next to the modified side channel also contain the upper mono band is possible.
  • the effect of scalability on the mono stereo encoder 32 is particularly cheap, if not LR coding but MS coding is used.
  • the invention Stereo signal processing by devices 12 and 14 is therefore especially in connection with the scalable encoder 32 particularly advantageous.
  • a mono stereo scalability to obtain, namely, MS coding can be used if they are actually compared to the LR coding is no longer preferable. This is what it does achieved that the side channel at the entrance of the scalable Encoder 32 damped compared to the unmodified case is.
  • FIG. 3 is a dashed signal path 36 from scalable encoder 32 for device 12 for analysis located.
  • This dashed signal path 36 is intended to symbolize that certain measures to measure the amount to derive bits required by the scalable encoder to encode the stereo audio signal at input 10, do not have to be calculated directly in the device 12, but from the scalable encoder into the device 12 can be output, such as B.
  • Perceptual Entropy PE the reference to the use of short windows, etc. That means that these function blocks are not both in the facility 12 for analysis as well as in the scalable encoder 32 must be present, but that their implementation only in the scalable encoder 32 is sufficient.
  • the means for modifying 14 would to determine the measure 18 for the amount of bits, initially none Carry out modification.
  • the device shown in Fig. 3 would be in a "pre-run mode" where no bit stream is written, but where only the required degree of damping for the side channel determined becomes.
  • the means 14 for modifying with accordingly fixed factors x, y work.
  • the scalable encoder is a time / frequency transform encoder, so would the level of the scalable encoder 32, which performs the time-frequency transformation, be connected upstream of input 10.
  • the facilities 12, 14 and 30 would then be embedded in the scalable encoder 32.
  • the signal paths 36a, 36b illustrate that the modified ones Channels without M / S coding to a scalable encoder can be directed so that it can then determine can determine whether M / S or L / R coding is cheaper.

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Claims (18)

  1. Dispositif pour traiter un signal audio stéréo, comprenant un premier canal (L) et un deuxième canal (R) et présentant les caractéristiques suivantes:
    un dispositif d'analyse (12) du signal audio stéréo ou d'un signal dérivé du signal audio stéréo pour obtenir une mesure pour une quantité de bits nécessitées par un codeur (32) pour coder le signal audio stéréo moyennant l'utilisation d'un algorithme de codage et
    un dispositif (14) pour modifier le premier canal et le deuxième canal (L, R) pour obtenir un premier canal et un deuxième canal modifiés (L', R'),
    le dispositif de modification (14) répondant au dispositif d'analyse (12) pour entrer en action lorsque la mesure (18) pour la quantité de bits dépasse une mesure prédéterminée et
    le dispositif de modification (14) étant configuré de manière telle qu'une caractéristique d'un signal de somme du premier canal et du deuxième canal modifiés (L', R'), dont l'allure est similaire à l'énergie du signal de somme, se trouve dans un rapport prédéterminé avec la caractéristique d'un signal de somme du premier canal et du deuxième canal (L, R) et en ce qu'un signal de différence entre le premier canal et le deuxième canal modifiés (L', R') est atténué par rapport à un signal de différence entre le premier canal et le deuxième canal (L, R).
  2. Dispositif selon la revendication 1, dans lequel le dispositif d'analyse (12) présente les caractéristiques suivantes:
    un dispositif de détermination de la caractéristique de la somme du premier et du deuxième canal pour une durée temporelle prédéterminée;
    un dispositif de détermination de la caractéristique de la différence entre le premier canal et le deuxième canal pour une durée temporelle prédéterminée et
    un dispositif de formation du rapport entre la caractéristique de la somme du premier canal et du deuxième canal et la caractéristique de la différence entre le premier canal et le deuxième canal, le rapport des caractéristiques étant la mesure (18) pour la quantité de bits.
  3. Dispositif selon la revendication 1, dans lequel le dispositif d'analyse (12) présente les caractéristiques suivantes:
    un premier dispositif pour déterminer un premier rapport de caractéristiques entre le premier canal et le seuil de masquage psychoacoustique du premier canal pour un temps prédéterminé;
    un deuxième dispositif pour déterminer un deuxième rapport de caractéristiques entre le deuxième canal et le seuil de masquage psychoacoustique du deuxième canal pour un temps prédéterminé et
    un dispositif pour additionner le premier rapport de caractéristiques et le deuxième rapport de caractéristiques, la somme du premier rapport de caractéristiques et du deuxième rapport de caractéristiques faisant référence à la mesure (18) pour la quantité de bits.
  4. Dispositif selon la revendication 1, dans lequel le codeur (32) est aménagé pour utiliser des fenêtres longues ou courtes pour transformer un signal audio stéréo temporel en un signal audio stéréo spectral en réponse à la structure temporelle du signal audio stéréo et dans lequel le dispositif d'analyse (12) est aménagé pour détecter si des fenêtres courtes ou longues sont mises en oeuvre dans le codeur (32), la mesure pour la quantité de bits étant le fait que des fenêtres courtes sont mises en oeuvre.
  5. Dispositif selon l'une des revendications précédentes, dans lequel le dispositif de modification (14) est aménagé pour entrer en action de manière que le signal de différence entre le premier canal et le deuxième canal est progressivement atténué pour arriver à une atténuation déterminée en partant de la non-atténuation et pour entrer en action de manière telle que l'atténuation est progressivement réduite pour passer d'une atténuation déterminée à la non-atténuation.
  6. Dispositif selon la revendication 5, dans lequel la vitesse de l'atténuation est sélectionnée aussi lente que possible tout en restant tellement rapide qu'un mécanisme de réservoir de bits du codeur (32) est mis à profit pour que le codeur (32) ne réduise pas la largeur de bande audio et ne viole pas un seuil de masquage psychoacoustique lors d'une quantification.
  7. Dispositif selon l'une des revendications précédentes, dans lequel le dispositif de modification (14) est aménagé pour atténuer adaptivement le signal de différence en dépendance de la mesure qui a été déterminée.
  8. Dispositif selon la revendication 2, dans lequel le dispositif de modification (14) est aménagé pour atténuer le signal de différence en dépendance d'un rapport de caractéristiques généré par le dispositif de formation du rapport de caractéristiques de sorte que l'atténuation du signal de différence est élevée lorsque le rapport de caractéristiques est faible et que l'atténuation du signal de différence est faible lorsque le rapport de caractéristiques est élevé.
  9. Dispositif selon la revendication 7 ou la revendication 8, dans lequel le dispositif de modification (14) est aménagé de manière à atténuer le signal de différence adaptivement de manière telle que le rapport de caractéristiques entre le signal de différence et le signal de somme est sensiblement égal à une valeur prédéterminée.
  10. Dispositif selon l'une des revendications précédentes, dans lequel le dispositif de modification (14) présente les caractéristiques suivantes:
    un premier multiplicateur (22a) pour multiplier le premier canal (L) par un premier facteur (x);
    un deuxième multiplicateur (22b) pour multiplier le premier canal (L) par un deuxième facteur (y);
    un troisième multiplicateur (22c) pour multiplier le deuxième canal (R) par le premier facteur (x);
    un quatrième multiplicateur (22d) pour multiplier le deuxième canal (R) par le deuxième facteur (y);
    un premier additionneur (24a) pour additionner le signal de sortie du premier multiplicateur (22a) et le signal de sortie du quatrième multiplicateur (22d) pour générer le premier canal modifié (L') et
    un deuxième additionneur (24b) pour additionner le signal de sortie du troisième multiplicateur (22c) et le signal de sortie du deuxième multiplicateur (22b) pour générer le deuxième canal modifié (R'),
    le premier facteur et le deuxième facteur (x, y) étant sélectionnés de manière telle que le signal de somme du premier canal et du deuxième canal et le signal de somme du premier canal et du deuxième canal modifiés sont sensiblement égaux et que le signal de différence est atténué d'un facteur déterminé.
  11. Dispositif selon l'une des revendications précédentes, dans lequel le dispositif d'analyse (12) présente par ailleurs la caractéristique suivante:
    un dispositif pour déterminer si un angle de phase entre le premier canal et le deuxième canal (L, R) a une valeur proche de 180° ; et
    le dispositif de modification (14) présentant par ailleurs la caractéristique suivante:
    un dispositif pour inverser le signe d'un canal (L, R) au cas où l'angle de phase est proche de 180°.
  12. Dispositif selon l'une des revendications précédentes, dans lequel le premier canal et le deuxième canal (L, R) du signal stéréo sont donnés par des valeurs spectrales qui ont été générées au départ d'un signal stéréo temporel par transformation dans la zone spectrale, le dispositif de modification (14) étant aménagé pour effectuer une atténuation du signal de différence procédant par sélection de fréquences.
  13. Dispositif selon la revendication 12, dans lequel le dispositif de modification est aménagé pour atténuer plus fortement dans une plage de fréquences dans laquelle le repérage directionnel de l'ouïe humaine est réduit que dans une plage de fréquences dans laquelle le repérage directionnel de l'ouïe humaine n'est pas réduit.
  14. Dispositif selon l'une des revendications précédentes, lequel présente par ailleurs les caractéristiques suivantes:
    un dispositif centre / côté (30) pour générer un canal centre (M') équivalant à la moitié de la somme du canal gauche modifié (L') et du canal droit modifié (R'),
    un dispositif côté (30) pour générer un canal côté équivalant à la moitié de la différence du premier canal modifié (L') et du deuxième canal modifié (R') et
    un codeur cadrable (32) aménagé pour coder le canal centre (M') et écrire en tant que première couche de cadrage dans un train de bits (BS) et également aménagé pour coder le canal côté (S') et écrire en tant que deuxième couche de cadrage dans le train de bits (BS).
  15. Dispositif selon la revendication 14, dans lequel le codeur cadrable (32) est aménagé pour utiliser un dispositif réservoir de bits au cas où la mesure pour la quantité de bits dépasse une valeur prédéterminée, pour que la largeur de bande audio ne soit pas réduite et/ou que le seuil de masquage psychoacoustique ne soit pas violé.
  16. Dispositif selon l'une des revendications précédentes, dans lequel la caractéristique dont l'allure est similaire à l'énergie est l'énergie elle-même, la somme d'échantillons élevés au carré pendant une durée temporelle déterminée, la somme de valeurs spectrales élevées au carré dans une plage de fréquences déterminée, la somme de valeurs d'échantillons pendant une durée temporelle déterminée et/ou la somme de valeurs spectrales élevées au carré dans une plage de fréquences déterminée.
  17. Dispositif selon l'une des revendications précédentes, dans lequel le signal audio stéréo est traité par blocs et dans lequel le signal dérivé du signal audio stéréo et utilisé lors de l'analyse est le signal modifié d'un bloc de traitement précédent.
  18. Procédé pour traiter un signal audio stéréo, comprenant un premier canal (L) et un deuxième canal (R) et présentant les étapes suivantes:
    analyse (12) du signal audio stéréo ou d'un signal dérivé du signal audio stéréo pour obtenir une mesure pour une quantité de bits nécessitée par un algorithme de codage pour coder le signal audio stéréo et
    modification (14) du premier canal et du deuxième canal (L, R) pour obtenir un premier canal et un deuxième canal modifiés (L', R'), au cas où est déterminée, à l'étape de l'analyse, une mesure (18) pour la quantité de bits, laquelle dépasse une mesure prédéterminée, la modification devant être effectuée de manière telle qu'une caractéristique d'un signal de somme du premier canal et du deuxième canal modifiés (L', R'), dont l'allure est similaire à l'énergie du signal de somme, se trouve dans un rapport prédéterminé avec une caractéristique d'un signal de somme du premier canal et du deuxième canal (L, R) et en ce qu'un signal de différence entre le premier canal et le deuxième canal modifiés (L', R') est atténué par rapport à un signal de différence entre le premier canal et le deuxième canal (L, R).
EP00985148A 1999-12-08 2000-12-07 Procede et dispositif pour traiter un signal audio stereo Expired - Lifetime EP1230827B1 (fr)

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DE19959156A DE19959156C2 (de) 1999-12-08 1999-12-08 Verfahren und Vorrichtung zum Verarbeiten eines zu codierenden Stereoaudiosignals
DE19959156 1999-12-08
PCT/EP2000/012352 WO2001043503A2 (fr) 1999-12-08 2000-12-07 Procede et dispositif pour traiter un signal audio stereo

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US20030091194A1 (en) 2003-05-15
WO2001043503A3 (fr) 2002-05-10
JP2007316658A (ja) 2007-12-06
EP1230827A2 (fr) 2002-08-14
JP4000261B2 (ja) 2007-10-31
DE19959156A1 (de) 2001-06-28
WO2001043503A2 (fr) 2001-06-14
DE19959156C2 (de) 2002-01-31
DE50003945D1 (de) 2003-11-06
US7260225B2 (en) 2007-08-21
ATE251376T1 (de) 2003-10-15
JP2003516555A (ja) 2003-05-13
JP4579273B2 (ja) 2010-11-10

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