EP1854334B1 - Dispositif et procede de production d'un signal stereo code d'un morceau audio ou d'un flux de donnees audio - Google Patents

Dispositif et procede de production d'un signal stereo code d'un morceau audio ou d'un flux de donnees audio Download PDF

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EP1854334B1
EP1854334B1 EP06707184A EP06707184A EP1854334B1 EP 1854334 B1 EP1854334 B1 EP 1854334B1 EP 06707184 A EP06707184 A EP 06707184A EP 06707184 A EP06707184 A EP 06707184A EP 1854334 B1 EP1854334 B1 EP 1854334B1
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Prior art keywords
channel
stereo
uncoded
channels
signal
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German (de)
English (en)
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EP1854334A1 (fr
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Jan Plogsties
Harald Mundt
Harald Popp
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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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Priority to PL06707184T priority Critical patent/PL1854334T3/pl
Priority to EP09006142.5A priority patent/EP2094031A3/fr
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S3/00Systems employing more than two channels, e.g. quadraphonic
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S3/00Systems employing more than two channels, e.g. quadraphonic
    • H04S3/002Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution
    • H04S3/004For headphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00Stereophonic arrangements
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S5/00Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2400/00Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/01Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2420/00Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/03Application of parametric coding in stereophonic audio systems

Definitions

  • the present invention relates to multi-channel audio technology, and more particularly to multi-channel audio applications in conjunction with headphone techniques.
  • the two international patent applications WO 99/49574 and WO 99/14983 disclose audio signal processing techniques for driving a pair of oppositely located headphone speakers to provide a user with spatial perception of the audio scene through the two headphones, which is not just a stereo representation but a multi-channel presentation.
  • the listener receives via his or her headphones a spatial perception of an audio piece that is equal to its spatial perception at best, if the user would sit in a playback room, which is equipped for example with a 5.1 audio system.
  • each channel of the multi-channel audio track or multi-channel audio stream, as shown in Fig. 2 is shown, fed to its own filter, after which then the respective filtered belonging together channels are added, as shown below.
  • Fig. 10 shows a playback room 200 in which a so-called 5.1 audio system is arranged.
  • the 5.1 audio system includes a center speaker 201, a front-left speaker 202, a front-right speaker 203, a rear-left speaker 204, and a rear-right speaker 205.
  • a 5.1 audio system has an additional subwoofer 206, which is also referred to as a low-frequency enhancement channel.
  • a listener 207 wearing a headset 208 having a left headphone speaker 209 and a right headphone speaker 210.
  • Fig. 2 Processing means shown is now adapted to each channel 1, 2, 3 of the multi-channel inputs 20 with a filter H iL , the sound channel from the speaker to the left speaker 209 in Fig. 10 describes to filter and to further filter the same channel with a filter H iR representing the sound from one of the five speakers to the right ear 210 to the right speaker 210 of the headphone 208.
  • the filter H iL would represent the channel indicated by a dashed line 212
  • the filter H 1R would represent the channel represented by a dashed line 213.
  • the left earphone speaker 209 receives not only the direct sound, but also early reflections at an edge of the playback room and, of course, late reflections that are expressed in a diffuse reverberation.
  • FIG. 11 Such a filter representation is in Fig. 11 shown.
  • Fig. 11 a schematic example of an impulse response of a filter, for example, the filter H 1L of Fig. 2
  • the direct sound coming through line 212 in FIG Fig. 11 is represented by a peak at the beginning of the filter, whereas early reflections, such as those shown by 214 in FIG Fig. 10 are represented by a central region with a plurality of (discrete) smaller peaks in Fig. 11 be reproduced.
  • the diffuse reverberation is then typically not more resolved to individual peaks, since the sound of the speaker 202 is reflected in principle as often as desired, the energy of course, with each reflection and additional propagation distance continues to decrease, as by the decreasing energy in the rear section, with "diffuse reverberation" in Fig. 11 is indicated, is shown.
  • the filter shown comprises a filter impulse response, which has approximately a course, as shown by the schematic impulse response representation in Fig. 11 is reproduced.
  • each channel is filtered with a corresponding filter for the left ear to then simply add up the signals output from the filters, which are all for the left ear, to obtain the headphone output signal for the left ear L.
  • addition is made by the right ear adder 23 and the right headphone speaker 210 in FIG Fig. 10 in order to obtain the headphone output signal for the right ear by superimposing all the loudspeaker signals filtered with a corresponding filter for the right ear.
  • Headphone systems for generating a multi-channel headphone sounds are therefore complicated, bulky and expensive, due to the high computing power, the high power requirements for the necessary high computing power and high memory requirements for the ratings to be performed with the impulse response and the associated large volume or expensive blocks for the player is.
  • Such applications are therefore tied to home PC sound cards or laptop sound cards or home stereo systems.
  • the ever-expanding market for mobile players remains the multi-channel headphone sound closed because the computational requirements for filtering the multi-channels with z.
  • B. 12 different filters both in terms of processor resources as well as in terms of power consumption of the typically battery-powered devices are not feasible in the price segment. This is about a price segment at the lower (low) end of the scale. Especially this price segment is economically very interesting because of the large quantities.
  • the post-published EP 1 768 451 A1 discloses an apparatus for coding an acoustic signal and an apparatus for decoding an acoustic signal.
  • a coefficient table holds coefficients in the form of a matrix of two rows and n columns, the coefficients simulating head-related transfer characteristics to be used when a signal is to be reproduced.
  • a first signal output unit an N-channel frequency domain signal is converted into a two-channel downmix signal according to the coefficient table.
  • a second signal output unit exists for generating auxiliary information to be used to reconstruct only the channel signals based on a two-channel downmix signal, thereby allowing the downmix signal to be filtered according to a desired transfer function.
  • a decoder reproduces the original multi-channel signal by reproducing the original multi-channel signal space information, which is then used to reconstruct the multi-channel signal.
  • the object of the present invention is to provide an efficient signal processing concept that enables multi-channel headphone reproduction even on simple playback devices.
  • This object is achieved by a device for generating a coded stereo signal according to claim 1 or by a method for generating a coded stereo signal according to claim 10 or a computer program according to claim 11.
  • the present invention is based on the finding that the high-quality and attractive multi-channel headphone sound can be made available to all available playback devices, such as CD players or hardware players, by providing a multi-channel display of an audio track or audio data stream So, for example, a 5.1 representation of an audio piece outside of a hardware player, so z. B. in a high-power computer of a provider of a headphone signal processing.
  • the result of earphone signal processing is not simply played back but fed to a typical audio stereo coder, which then generates a coded stereo signal from the left earphone channel and the right earphone channel.
  • This coded stereo signal can then, like any other coded stereo signal, which has no multi-channel representation, the hardware player or z.
  • the result of the headphone signal processing ie the left and the right headphone signal is not reproduced in a headphone, as in the prior art, but coded and output as coded stereo data.
  • Such an output may be a store, a transfer, or anything like that.
  • Such a file with coded stereo data can then be readily supplied to any player designed for stereo playback without the user having to make any changes to his device.
  • the concept according to the invention of generating a coded stereo signal from the result of the headphone signal processing thus makes it possible for the multichannel presentation, which provides a much better and more lifelike quality for a user, to also be simple and very widely used in the future even more widespread hardware players can be used.
  • a coded multi-channel representation that is, a parameter representation having one or typically two base channels, and further comprising parameter data to on the basis of the base channels and the parameter data to the multi-channels of the multi-channel representation produce.
  • the headphone signal processing according to the invention is performed not in the time domain by convolution of the time signal with the impulse response, but in the frequency domain by multiplication with the filter transfer function.
  • the present invention is used as a multi-channel representation of a BCC representation with one or preferably two base channels.
  • the BCC method works in the frequency domain, the multi-channels are not transformed into the time domain after their synthesis, as is usual with the BCC decoder. Instead, the spectral representation of the multichannels is used in blocks and subjected to the headphone signal processing.
  • the transfer functions of the filters are used, that is, the Fourier transforms the impulse responses to perform a multiplication between the spectral representation of the multi-channels and the filter transfer functions.
  • a block-wise filter processing is preferred in which the impulse responses of the filters in the time domain be separated and be transformed block by block to then perform necessary for such measures weights of the spectra, as described for example in the WO 94/01933 are disclosed.
  • Fig. 1 shows a schematic block diagram of an inventive device for generating a coded stereo signal of an audio track or audio data stream.
  • the stereo signal comprises in uncoded form an uncoded first stereo channel 10a and an uncoded second stereo channel 10b, and is generated from a multi-channel representation of the audio piece or audio data stream, the multi-channel representation comprising information on more than two multi-channels.
  • the multi-channel representation may be in uncoded or encoded form. If the multi-channel representation is present in uncoded form, it includes three or more multi-channels. In a preferred application scenario, the multichannel presentation includes five channels and a subwoofer channel.
  • this encoded form typically includes one or more base channels and parameters for synthesizing the three or more multi-channels from the one or both base channels.
  • a multi-channel decoder 11 is therefore one example of a means for providing the more than two multi-channels from the multi-channel representation.
  • the multichannel presentation already in uncoded form ie z. B. in the form of 5 + 1 PCM channels, the means for providing an input terminal for a device 12 for Performing headphone signal processing to produce the uncoded stereo signal with the uncoded first stereo channel 10a and the uncoded second stereo channel 10b.
  • the headphone signal processing means 12 is arranged to evaluate the multi-channels of the multi-channel display respectively with a first filter function for the first stereo channel and a second filter function for the second stereo channel and weighted multi-channels in each case to obtain the uncoded first stereo channel and the uncoded second stereo channel, as determined by Fig. 2 has been shown.
  • the device 12 for performing the headphone signal processing is followed by a stereo encoder 13, which is configured to encode the first uncoded stereo channel 10a and the second uncoded stereo channel 10b to the encoded stereo signal at an output 14th of the stereo encoder 13.
  • the stereo encoder performs data rate reduction so that a data rate necessary to transmit the encoded stereo signal is less than a data rate necessary to transmit the uncoded stereo signal.
  • inventive concept allows the computationally expensive steps to multi-channel decode and perform headphone signal processing not be performed in the player itself, but are performed externally.
  • the result of the inventive concept is a coded stereo file, which is for example an MP3 file, an AAC file, a HE AAC file or any other stereo file.
  • the multi-channel decoding, the headphone signal processing and the stereo encoding can be performed on different devices, since the output data or input data of the individual blocks are easily portable and standardized generated and stored.
  • Fig. 7 a preferred embodiment of the present invention is shown, in which the multi-channel decoder 11 has a filter bank or an FFT function, such that the multi-channel representation is provided in the frequency domain.
  • the individual multi-channels are generated separately as blocks of spectral values for each channel.
  • the headphone signal processing is then not performed in the time domain by folding the temporal channels with the filter impulse responses, but it is performed a multiplication of the frequency domain representation of the multi-channels with a spectral representation of the filter impulse response.
  • an uncoded stereo signal is reached, which is not present in the time domain, but comprising a left and a right stereo channel, wherein such a stereo channel is given as a result of blocks of spectral values, each block of spectral values represents a short-term spectrum of the stereo channel.
  • the headphone signal processing block 12 is supplied on the input side either with time domain or frequency domain data.
  • the uncoded stereo channels in the frequency domain ie again generated as a sequence of blocks of spectral values.
  • a stereo coder which is transformation-based, that processes spectral values, without a frequency / time conversion and a subsequent time between the headphone signal processor 12 and the stereo coder 13 is preferred as the stereo coder 13 Frequency conversion is required.
  • the stereo encoder 13 On the output side, the stereo encoder 13 then outputs a file with the coded stereo signal, which in addition to page information comprises a coded form of spectral values.
  • a continuous frequency domain processing performed without a conversion into the time domain and, if appropriate, again has to be implemented in the frequency domain. If an MP3 coder or an AAC coder is used as the stereo coder, it is preferable to convert the Fourier spectrum at the output of the headphone signal processing block into an MDCT spectrum.
  • phase information which is required exactly for the convolution / evaluation of the channels in the headphone signal processing block is converted into the MDCT representation which does not operate in phase-correct manner, so that for the stereo encoder, in contrast to a normal MP3 encoder or a normal AAC encoder no means for converting time domain in the frequency domain, ie in the MDCT spectrum is needed.
  • Fig. 9 shows a general block diagram for a preferred stereo encoder.
  • the stereo encoder comprises on the input side a joint stereo module 15, which preferably adaptively determines whether a common stereo coding, for example in the form of a middle / side encoding, provides a higher coding gain than a separate processing from left and right channel.
  • the joint stereo module 15 may further be configured to perform intensity stereo coding, wherein intensity stereo coding provides a significant coding gain, especially at higher frequencies, without audible artifacts occurring.
  • the output of the joint stereo module 15 is then further processed using various other redundancy-reducing measures, such as TNS filtering, noise substitution, etc., and then applying the results to a quantizer 16 which quantizes using a psychoacoustic masking threshold of the spectral values.
  • the quantizer step size is chosen such that the noise introduced by the quantization remains below the psychoacoustic marker threshold, so that a data rate reduction is achieved without the distortions introduced by the lossy quantization becoming audible.
  • the quantizer 16 is finally followed by an entropy coder 17, which performs a lossless entropy coding of the quantized spectral values.
  • the encoded stereo signal is then present, which comprises side information necessary for decoding in addition to the entropy-coded spectral values.
  • Fig. 3 shows a joint stereo device 60.
  • This device may be a device implementing, for example, the intensity stereo (IS) technique or the binaural cue coding technique (BCC).
  • IS intensity stereo
  • BCC binaural cue coding technique
  • Such a device usually receives as input at least two Channels CH1, CH2, .... CHn, and outputs a single carrier channel and multi-channel parametric information.
  • the parametric data is defined so that an approximation of an original channel (CH1, CH2, ..., CHn) can be calculated in a decoder.
  • the carrier channel will include subband samples, spectral coefficients, time domain samples, etc. that provide a relatively fine representation of the underlying signal, while the parametric data does not include such samples or spectral coefficients, but control parameters for controlling a particular reconstruction algorithm, such as multiplying by weighting. by time shifting, by frequency shifting, etc.
  • the parametric multi-channel information therefore comprises a relatively rough representation of the signal or the associated channel.
  • the amount of data needed by a carrier channel is about 60 to 70 kbps, while the amount of data required by one channel parametric page information is in the range of 1.5 to 2.5 kbps.
  • the above figures apply to compressed data.
  • a non-compressed CD channel requires data rates on the order of about ten times.
  • An example of parametric data is the known scale factors, intensity stereo information, or BCC parameters, as set forth below.
  • the reconstructed signals differ in their amplitude, but they are identical in terms of their phase information.
  • the energy-time envelopes of both original audio channels are maintained by the selective scaling operation, which typically operates in a frequency-selective manner. This corresponds to the human perception of sound at high frequencies, where the dominant spatial information is determined by the energy envelopes.
  • the transmitted signal i. H. the carrier channel is generated from the sum signal of the left channel and the right channel instead of the rotation of both components.
  • this processing i. H. generating intensity-stereo parameters to perform the scaling operations in a frequency-selective manner, i. H. independent for each scale factor band, d. H. for each encoder frequency partition.
  • both channels are combined to form a combined or "carrier" channel and, in addition to the combined channel, the intensity stereo information.
  • the intensity stereo information depends on the energy of the first channel, the energy of the second channel or the energy of the combined channel.
  • the BCC technique is described in the AES convention paper 5574 " Binaural Cue Coding Applied to Stereo and Multichannel Audio Compression ", T. Faller, F. Baumgarte, May 2002, Münch en, described.
  • BCC coding a number of audio input channels are converted into a spectral representation, using a DFT-based transformation with overlapping windows. The resulting spectrum is divided into non-overlapping sections, each of which has an index. Each partition has a bandwidth proportional to the equivalent rectangular bandwidth (ERB).
  • the Inter Channel Level Differences (ICLD) and the Inter Channel Time Differences (ICTD) are determined for each partition and for each frame k. The ICLD and ICTD are quantized and encoded to eventually arrive as page information in a BCC bitstream.
  • the inter-channel level differences and the inter-channel time differences are given for each channel relative to a reference channel. Then, the parameters are calculated according to predetermined formulas that depend on the particular partitions of the signal to be processed.
  • the decoder On the decoder side, the decoder typically receives a mono signal and the BCC bit stream.
  • the mono signal is transformed into the frequency domain and input to a spatial synthesis block which also receives decoded ICLD and ICTD values.
  • the BCC parameters ICLD and ICTD are used to perform a mono signal weighting operation to synthesize the multichannel signals representing, after a frequency / time conversion, a reconstruction of the original multichannel audio signal.
  • the joint stereo module 60 operates to output the channel-side information such that the parametric channel data is quantized and coded ICLD or ICTD parameters using one of the original channels as the reference channel for encoding the channel side information.
  • the carrier signal is formed from the sum of the participating source channels.
  • the above techniques provide only a monodic representation for a decoder that can only process the carrier channel, but is unable to process the parametric data to produce one or more approximations of more than one input channel.
  • Fig. 5 shows such a BCC scheme for encoding / transmission of multi-channel audio signals.
  • the multi-channel audio input signal at an input 110 of a BCC encoder 112 is down-mixed in a so-called downmix block 114.
  • the original multi-channel signal at the input 110 is a 5-channel surround signal having a front left channel, a front right channel, a left surround channel, a right surround channel and a center channel.
  • the downmix block 114 generates a sum signal by simply adding these five channels into a mono signal.
  • This single channel is output on a sum signal line 115.
  • Page information provided by the BCC analysis block 116 is output on a page information line 117.
  • inter-channel level differences ICLD
  • inter-channel time differences ICTD
  • the BCC analysis block 116 is also capable of calculating inter-channel correlation (ICC) values.
  • the sum signal and the page information are transmitted in a quantized and encoded format to a BCC decoder 120.
  • the BCC decoder decomposes the transmitted sum signal into a number of subbands and performs scaling, delays and other processing to provide the subbands of the multichannel audio channels to be output. This processing is performed so that the ICLD, ICTD and ICC parameters (cues) of a reconstructed multichannel signal at output 121 match the corresponding cues for the original multichannel signal at input 110 in BCC encoder 112.
  • the BCC decoder 120 includes a BCC synthesis block 122 and a page information revision block 123.
  • the sum signal on line 115 is fed to a time / frequency conversion unit or filter bank FB 125.
  • FB 125 At the output of the block 125 there exists a number N of subband signals or, in an extreme case, a block of spectral coefficients, when the audio filter bank 125 performs a 1: 1 transform, ie a transform producing N spectral coefficients from N time domain samples.
  • the BCC synthesis block 122 further includes a delay stage 126, a level modification stage 127, a correlation processing stage 128, and an inverse filter bank stage IFB 129.
  • stage 129 At the output of stage 129, the reconstructed Multi-channel audio signal with, for example, five channels in the case of a 5-channel surround system are output to a set of speakers 124 as shown in FIG Fig. 5 or Fig. 4 are shown.
  • the input signal sn is converted into the frequency domain or the filter bank region by means of the element 125.
  • the signal output by element 125 is copied so as to obtain multiple versions of the same signal, as represented by copy node 130.
  • the number of versions of the original signal is equal to the number of output channels in the output signal.
  • each version of the original signal at node 130 undergoes a certain delay d 1 , d 2 , ..., d i , ... d N.
  • the delay parameters are determined by the page information processing block 123 in FIG Fig. 5 and from the inter-channel time differences as determined by the BCC analysis block 116 of FIG Fig. 5 have been calculated derived.
  • the multiplication parameters a 1 , a 2 ,..., A i ,..., A N which are also calculated by the page information processing block 123 based on the interchannel level differences as calculated by the BCC analysis block 116. be calculated.
  • the ICC parameters calculated by BCC analysis block 116 are used to control the functionality of block 128 so that certain correlations between the delayed and level manipulated signals are obtained at the outputs of block 128. It should be noted here that the order of stages 126, 127, 128 is different from the one in FIG Fig. 6 may differ.
  • the BCC analysis is performed frame by frame, so temporally variable, and that further a frequency-wise BCC analysis is obtained as indicated by the filter bank partitioning Fig. 6 is apparent.
  • the BCC parameters are obtained for each spectral band.
  • the audio filter bank 125 decomposes the input signal into, for example, 32 bandpass signals
  • the BCC analysis block obtains a set of BCC parameters for each of the 32 bands.
  • the BCC synthesis block 122 performs Fig. 5 who is detailed in Fig. 6 is a reconstruction, which is based on the exemplified 32 bands.
  • Fig. 4 presented a scenario that is used to determine individual BCC parameters. Normally the ICLD, ICTD and ICC parameters can be defined between channel pairs. However, it is preferred to determine the ICLD and ICTD parameters between a reference channel and each other channel. This is in Fig. 4A shown.
  • ICC parameters can be defined in several ways. Generally speaking, one can determine ICC parameters in the encoder between all possible channel pairs, as shown in FIG Fig. 4B is shown. However, it has been proposed to calculate only ICC parameters between the strongest two channels at a time, as in Fig. 4C where an example is shown in which one ICC parameter between channels 1 and 2 is calculated one at a time, and at another time an ICC parameter between channels 1 and 5 is calculated.
  • the decoder then synthesizes the inter-channel correlation between the strongest channels in the decoder and uses certain heuristic rules to compute and synthesize the inter-channel coherence for the remaining channel pairs.
  • the multiplication parameters a 1 , a N based on the transmitted ICLD parameters
  • the ICLD parameters represent an energy distribution of an original multichannel signal. Without loss of generality, it is preferred as shown in FIG Fig. 4A shown to take four ICLD parameters representing the energy difference between the respective channels and the front left channel.
  • the multiplication parameters a 1 , ..., a N are derived from the ICLD parameters such that the total energy of all reconstructed output channels is the same (or proportional to the energy of the transmitted sum signal).
  • Fig. 7 The embodiment shown is based on the frequency-time conversion, which by the inverse filter banks IFB 129 of Fig. 6 be achieved, waived. Instead, the spectral representations of the individual channels are used at the input of these inverse filter banks and the headphone signal processing device of Fig. 7 supplied to perform the evaluation of the individual multi-channels with the two filters per multi-channel without an additional frequency / time transformation.
  • the multi-channel decoder so z. B. the filter bank 125 of Fig. 6 and the stereo encoder should have the same time / frequency resolution. Furthermore, it is preferred to use one and the same filter bank, which is particularly advantageous in that for the entire processing, as shown in FIG Fig. 1 represents, only a single filter bank is needed. In this case, a particularly efficient processing, since the transformation in the multi-channel decoder and in the stereo encoder do not need to be calculated.
  • the input data or output data in the inventive concept are therefore preferably coded in the frequency domain by means of transformation / filter bank and are coded according to psychoacoustic specifications taking advantage of masking effects, in which case a spectral representation of the signals should be present in particular in the decoder.
  • Examples include MP3 files, AAC files or AC3 files.
  • the input data or output data can also be encoded by summing and subtraction, as is the case with so-called matrixed methods. Examples are Dolby ProLogic, Logic7 or Circle Surround.
  • the data, in particular the multichannel representation can additionally be coded with parametric methods, as is the case with MP3 surround, this method being based on the BCC technique.
  • the inventive method for generating can be implemented in hardware or in software.
  • the implementation may be on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals, which may interact with a programmable computer system such that the method is performed.
  • the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out a method according to the invention, when the computer program product runs on a computer.
  • the invention can thus be realized as a computer program with a program code for carrying out the method when the computer program runs on a computer.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)
  • Stereo-Broadcasting Methods (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
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Claims (11)

  1. Dispositif de génération d'un signal audio codé d'un morceau audio ou d'un flux de données audio avec un premier canal stéréo et un deuxième canal stéréo à partir d'une représentation multicanal du morceau audio ou du flux de données audio présentant des informations sur plus de deux canaux multiples, aux caractéristiques suivantes:
    un moyen (11) destiné à préparer les plus de deux canaux multiples à partir de la représentation multicanal;
    un moyen (12) destiné à effectuer un traitement de signal d'écouteur, pour générer un signal stéréo non codé avec un premier canal stéréo non codé (10a) et un deuxième canal stéréo non codé (10b), le moyen (12) destiné à effectuer étant réalisé,
    pour évaluer chaque canal multiple avec une première fonction de filtre (HiL), dérivée d'une position virtuelle d'un haut-parleur pour la reproduction du canal multiple et d'une première position d'oreille virtuelle d'un auditeur, pour le premier canal stéréo et avec une deuxième fonction de filtre (HiR), dérivée d'une position virtuelle du haut-parleur et d'une deuxième position d'oreille virtuelle de l'auditeur, pour le deuxième canal stéréo, pour générer, pour chaque canal multiple, un premier canal évalué et un deuxième canal évalué, les deux positions d'oreille virtuelles de l'auditeur étant différentes,
    pour additionner (22) les premiers canaux évalués, pour obtenir le premier canal stéréo non codé (10a), et
    pour additionner (23) les deuxièmes canaux évalués, pour obtenir le deuxième canal stéréo non codé (10b);
    pour utiliser la première fonction de filtre (HiL) qui tient compte du bruit direct, des réflexions et de la réverbération diffuse et la deuxième fonction de filtre (HiR) qui tient compte du bruit direct, des réflexions et de la réverbération diffuse, et
    un codeur stéréo (13) destiné à coder le premier canal stéréo non codé (10a) et le deuxième canal stéréo non codé (10b), pour obtenir le signal stéréo codé (14), le codeur stéréo étant réalisé de sorte qu'un débit de données requis pour la transmission du signal stéréo codé soit inférieur à un débit de données requis pour la transmission du signal stéréo non codé.
  2. Dispositif selon la revendication 1, dans lequel la première et la deuxième fonction de filtre correspondent à une réponse impulsionnelle de filtre présentant un pic à une faible valeur de temps représentant le bruit direct, plusieurs pics plus petits à des valeurs de temps moyennes représentant les réflexions, et une zone continue qui n'est plus résolue selon des pics individuels et représentant la réverbération diffuse.
  3. Dispositif selon l'une des revendications précédentes,
    dans lequel la représentation multicanal présente un ou plusieurs canaux de base ainsi que des informations de paramètres pour le calcul des canaux multiples à partir d'un ou de plusieurs canaux de base, et
    dans lequel le moyen (11) destiné à préparer est réalisé de manière à calculer, à partir de l'un ou des plusieurs canaux de base et des informations de paramètres, les au moins trois canaux multiples.
  4. Dispositif selon la revendication 3,
    dans lequel le moyen (11) destiné à préparer est réalisé de manière à fournir du côté de la sortie une représentation de plage de fréquences par bloc pour chaque canal multiple, et
    dans lequel le moyen (12) destiné à effectuer est réalisé de manière à évaluer la représentation de plage de fréquences par bloc avec une représentation de plage de fréquences de la première et de la deuxième fonction de filtre.
  5. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (12) destiné à effectué est réalisé de manière à fournir une représentation de plage de fréquences par bloc du premier canal stéréo non codé et du deuxième canal stéréo non codé, et
    dans lequel le codeur stéréo (13) est un codeur à base de transformation et est par ailleurs réalisé de manière à traiter la représentation de plage de fréquences par bloc du premier canal stéréo non codé et du deuxième canal stéréo non codé sans transformation de la représentation de plage de fréquences en une représentation dans le temps.
  6. Dispositif selon l'une des revendications précédentes,
    dans lequel le codeur stéréo (13) est réalisé de manière à effectuer un codage stéréo commun (15) du premier et du deuxième canal stéréo.
  7. Dispositif selon l'une des revendications précédentes,
    dans lequel le codeur stéréo (13) est réalisé de manière à quantifier (16) un bloc de valeurs spectrales à l'aide d'un seuil de masquage psycho-acoustique et à le soumettre à un codage entropique (17), pour obtenir le signal stéréo codé.
  8. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (11) destiné à préparer est réalisé sous forme de décodeur BCC.
  9. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (11) destiné à préparer est réalisé sous forme de décodeur multicanal présentant un banc de filtres avec plusieurs sorties,
    dans lequel le moyen (12) destiné à effectuer est réalisé de manière à évaluer les signaux aux sorties du banc de filtres par la première et la deuxième fonction de filtre, et
    dans lequel le codeur stéréo (13) est réalisé de manière à quantifier le premier canal stéréo non codé présent dans la plage de fréquences et le deuxième canal stéréo non codé présent dans la plage de fréquences (16) et à les soumettre à un codage entropique (17), pour obtenir le signal stéréo codé.
  10. Procédé de génération d'un signal audio codé d'un morceau audio ou d'un flux de données audio avec un premier canal stéréo et un deuxième canal stéréo à partir d'une représentation multicanal du morceau audio ou du flux de données audio présentant des informations sur plus de deux canaux multiples, aux étapes suivantes consistant à:
    préparer (11) les plus de deux canaux multiples à partir de la représentation multicanal;
    effectuer (12) un traitement de signal d'écouteur, pour générer un signal stéréo non codé avec un premier canal stéréo non codé (10a) et un deuxième canal stéréo non codé (10b), l'étape (12) consistant à effectuer (12) présentant les étapes suivantes consistant à:
    évaluer chaque canal multiple avec une première fonction de filtre (HiL), dérivée d'une position virtuelle d'un haut-parleur pour la reproduction du canal multiple et d'une première position d'oreille virtuelle d'un auditeur, pour le premier canal stéréo et avec une deuxième fonction de filtre (HiR), dérivée d'une position virtuelle du haut-parleur et d'une deuxième position d'oreille virtuelle de l'auditeur, pour le deuxième canal stéréo, pour générer, pour chaque canal multiple, un premier canal évalué et un deuxième canal évalué, les deux positions d'oreille virtuelles de l'auditeur étant différentes,
    additionner (22) les premiers canaux évalués, pour obtenir le premier canal stéréo non codé (10a), et
    additionner (23) les deuxièmes canaux évalués, pour obtenir le deuxième canal stéréo non codé (10b);
    utiliser la première fonction de filtre (HiL) qui tient compte du bruit direct, des réflexions et de la réverbération diffuse et la deuxième fonction de filtre (HiR) qui tient compte du bruit direct, des réflexions et de la réverbération diffuse; et
    coder stéréo (13) le premier canal stéréo non codé (10a) et le deuxième canal stéréo non codé (10b), pour obtenir le signal stéréo codé (14), l'étape de codage stéréo étant réalisée de sorte qu'un débit de données requis pour la transmission du signal stéréo codé soit inférieur à un débit de données requis pour la transmission du signal stéréo non codé.
  11. Programme d'ordinateur avec un code de programme pour réaliser le procédé pour générer un signal stéréo codé selon la revendication 10 lorsque le programme d'ordinateur est exécuté sur un ordinateur.
EP06707184A 2005-03-04 2006-02-22 Dispositif et procede de production d'un signal stereo code d'un morceau audio ou d'un flux de donnees audio Active EP1854334B1 (fr)

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EP09006142.5A EP2094031A3 (fr) 2005-03-04 2006-02-22 Dispositif et procédé de production d'un signal stéréo codé d'un élément audio ou d'un courant de données audio

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DE102005010057A DE102005010057A1 (de) 2005-03-04 2005-03-04 Vorrichtung und Verfahren zum Erzeugen eines codierten Stereo-Signals eines Audiostücks oder Audiodatenstroms
PCT/EP2006/001622 WO2006094635A1 (fr) 2005-03-04 2006-02-22 Dispositif et procede de production d'un signal stereo code d'un morceau audio ou d'un flux de donnees audio

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PL1854334T3 (pl) 2010-09-30
EP1854334A1 (fr) 2007-11-14
CA2599969C (fr) 2012-10-02
RU2007136792A (ru) 2009-04-10

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