Deprecated: The each() function is deprecated. This message will be suppressed on further calls in /home/zhenxiangba/zhenxiangba.com/public_html/phproxy-improved-master/index.php on line 456
EP0478615B2 - Polyphonische kodierung - Google Patents
[go: Go Back, main page]

EP0478615B2 - Polyphonische kodierung - Google Patents

Polyphonische kodierung Download PDF

Info

Publication number
EP0478615B2
EP0478615B2 EP90909155A EP90909155A EP0478615B2 EP 0478615 B2 EP0478615 B2 EP 0478615B2 EP 90909155 A EP90909155 A EP 90909155A EP 90909155 A EP90909155 A EP 90909155A EP 0478615 B2 EP0478615 B2 EP 0478615B2
Authority
EP
European Patent Office
Prior art keywords
filter
signal
channel
sum
difference
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Lifetime
Application number
EP90909155A
Other languages
English (en)
French (fr)
Other versions
EP0478615A1 (de
EP0478615B1 (de
Inventor
Christopher Ellis Holt
Edward Munday
Barry Michael George Cheetham
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
British Telecommunications PLC
Original Assignee
British Telecommunications PLC
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Family has litigation
First worldwide family litigation filed litigation Critical https://patents.darts-ip.com/?family=10658483&utm_source=google_patent&utm_medium=platform_link&utm_campaign=public_patent_search&patent=EP0478615(B2) "Global patent litigation dataset” by Darts-ip is licensed under a Creative Commons Attribution 4.0 International License.
Application filed by British Telecommunications PLC filed Critical British Telecommunications PLC
Publication of EP0478615A1 publication Critical patent/EP0478615A1/de
Application granted granted Critical
Publication of EP0478615B1 publication Critical patent/EP0478615B1/de
Publication of EP0478615B2 publication Critical patent/EP0478615B2/de
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Images

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04H—BROADCAST COMMUNICATION
    • H04H20/00—Arrangements for broadcast or for distribution combined with broadcast
    • H04H20/86—Arrangements characterised by the broadcast information itself
    • H04H20/88—Stereophonic broadcast 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques

Definitions

  • This invention relates to polyphonic coding techniques, particularly, but not exdusively, for coding speech signals.
  • polyphonic specifically stereophonic
  • sound is more perceptually appealing than monophonic sound.
  • polyphonic sound allows a spatial reconstruction of the original sound field with an image of each sound source being perceived at an identifiable point corresponding to its position in the original conference room. This can eliminate confusion and misunderstandings during audio-conference discussions since each participant may be identified both by the sound of his voice and by his perceived position within the conference room.
  • polyphonic transmissions require an increase in transmission capacity as compared with monophonic transmissions.
  • the conventional approach of transmitting two independent channels thus doubling the required transmission capacity, imposes an unnaceptably high cost penalty in many applications and is not possible in some cases because of the need to use existing channels with fixed transmission capacities.
  • each microphone signal (referred to hereinafter as x L (t) with Laplace transform x L (s) and x R (t) with Laplace transform X R (s) respectively) may be considered to be the superposition of source signals processed by respective acoustic transfer functions. These transfer functions are strongly affected by the distances between the sound sources and each microphone and also by the acoustic properties of the room. Taking the case of a single source, e.g.
  • the distances between the source and the left and right microphones give rise to different delays, and there will also be different degrees of attenuation.
  • the signal reaching each microphone may have travelled via many reflected paths (e.g. from walls or ceilings) as well as directly, producing time spreading, frequency dependent colouration due to resonances and antiresonances, and perhaps discrete echos.
  • H interchannel transferfunction
  • US patent no. 4815132 describes a stereophonic coding system which receives right- and left-hand channels. It transmits the right-hand channel but for the left-hand channel it uses a plural-order adaptive filter to generate filter coefficients (or a filter residual) which are transmitted instead. The receiver uses this information to control a filter which filters the right-hand channel to generate a reconstructed left-hand channel.
  • a polyphonic signal coding apparatus comprising:
  • the reconstructing data are filter coefficients.
  • the residual signal representing the difference between (for example) a difference signal and a sum signal when thus filtered is formed at the transmitter, and this is transmitted as the reconstruction data.
  • the prediction residual signal may be efficiently encoded to allow a backward adaptation technique to be used at the decoder for deriving the prediction filter coefficients.
  • the residual is also used as an error signal which is added to the prediction filter's output at the decoder to correct for inaccuracies in the prediction of the second channel from the first.
  • the means for generating the filter coefficients is an adaptive filter, advantageously a lattice filter.
  • This type of filter also gives advantages in non-sum and difference polyphonic systems.
  • variable delay means are disposed in at least one of the input signal paths, and controlled to time align the two signals prior to forming the sum and difference signals so that causal prediction filters of reasonable order can be used.
  • This aspect of the invention has several important advantages:
  • a method of calculating approximations to H(s) when the source signals are not white (which, of course, includes all speech or music signals) is proposed in a second aspect of the invention, using the idea of a 'prewhitening filter'.
  • a method of coding polyphonic input signals comprising:
  • prediction and predictor in this specification include not only prediction of future data from past data, but also estimation of present data of a channel from past and present data of another channel.
  • One possible way of removing the redundancy between two input signals (or predicting one from the other) would be to connect between the two channels an adaptive predictor filter whose slowly changing parameters are calculated by standard techniques (such as, for example, block cross-correlation analysis or sequential lattice adaptation) .
  • the two signals will originate from sound sources within a room, and the acoustic transfer function between each source and each microphone will be characterised typically by weak poles (from room resonances) and strong zeros (due to absorption and destructive interference).
  • An all-zero filter could therefore produce a reasonable approximation to the acoustic transfer function between a source and a microphone and such a filter could also be used to predict say the left microphone signal x L (t) from x R (t) when the source is close to the right microphone.
  • the filter must now model a transfer function with weak zeros and strong poles - a difficult task for an all-zero filter.
  • Other types of filter are not, in general, inherently stable. The net effect of this is to cause unequal degradation in the reconstructed channel when the source shifts from one microphone to the other. This further makes the simplistic prediction of one channel (say, the left) from the other (say, the right) hard to realise.
  • x R (t) and x L (t) will be processed in sampled data form as the digital signals x R [n] and x L [n] ( or x S [n] and x D [n] ) and it will be more convenient to use the 'z-transform' transfer fuction H(z) rather than H(s)
  • the invention in its essential form the invention comprises a pair of inputs 1a, 1b for receiving a pair of speech signals, e.g. from left and right microphones.
  • the signals at the inputs, x R (t) and x L (t) may be in digital form. It may be convenient at this point to pre-process the signals, e.g. by band limiting.
  • X D (t) H(s) X S (s).
  • the sum and difference signals are then supplied to filter derivation stage 4, which derives the coefficients of a multi-stage prediction filter which, when driven with the sum signal, will approximate the difference signal.
  • the difference between the approximated difference signal and the actual difference signal, the prediction residual signal, will usually also be produced (although this is not invariably necessary).
  • the sum signal is then encoded (preferably using LPC or sub-band coding), for transmission or storage, along with further data enabling reconstruction of the difference signal.
  • the filter coefficients may be sent, or alternatively (as discussed further below), the residual signal may be transmitted, the difference channel being reconstituted by deriving the filter parameters at the receiver using a backwards adaptive process known in the art; or both may be transmitted.
  • one simple and effective way of providing the derivation stage 4 is to use an adaptive filter (for example, an adaptive transversal filter) receiving as input the sum channel and modelling the difference channel so as to reduce the prediction residual.
  • an adaptive filter for example, an adaptive transversal filter
  • Such general techniques of filter adaptation are well-known in the art.
  • the sum signal x S (t) is received together with either the filter parameters or the residual signal, or both, for the difference channel, and an adaptive filter 5 corresponding to that for which the parameters were derived at the coder receives as input the sum signal and produces as output the reconstructed difference signal when configured either with the received parameters or with parameters derived by backwards adaptation from the received residual signal.
  • Sum and difference signals are then both fed to an adder 6 and a subtracter 7, which produce as outputs respectively the reconstructed left and right channels at output nodes 8a and 8b.
  • the encoder Since a high-quality sum signal is sent, the encoder is fully mono-compatible. In the event of loss of stereo information, monophonic back-up is thus available.
  • one component of the transfer functions H L and H R is a delay component relating to the direct distance between the signal source and each of the microphones, and there is a corresponding delay difference d. There is thus a strong cross-correlation between one channel and the other when delayed by d.
  • An alternative method of delay estimation found in papers on sonar research is to use an adaptive filter.
  • the left channel input is delayed by half the filter length and the coefficients are updated using the LMS algorithm to minimise the mean-square error or the output.
  • the transversal filter coefficients will, in theory, become the required cross-correlation coefficients. This may seem like unnecessary repetition of filter coefficient derivation were it not for the property of this delay estimator that the maximum value of the cross-correlation coefficient (at the position of the maximum filter coefficient) is obtained some time before the filter has converged.
  • This method may be improved further because spatial information is also available from the relative amplitudes of the input channels; this could be used to apply a weighting function to the filter coefficients to speed convergence.
  • the complexity and length of the filter to be calculated is therefore reduced by calculating the required value of d in a delay calculator stage 9 (preferably employing one of the above methods), and then bringing the channels into time alignment by delaying one or other by d using, for example, a pair of variable delays 10a, 10b (although one fixed and one variable delay could be used) controlled by the delay calculator 9. With the major part of the speech information in the channels time aligned, the sum and difference signals are then formed.
  • the delay length d is preferably transmitted to the decoder, so that after reconstructing the difference channel and subsequently the left and right channels, corresponding variable length delay stages 11a, 11b in one or other of the channels can restore the interchannel delay.
  • the "sum" signal is thus no longer quite the true sum of x L (t) + x R (t); because of the delay d it is x L (t) + x R (t-d). It may therefore be preferred to locate the delays 10a, 10b (and, possibly, the delay calculator) downstream of the adder and subtractor 2 and 3; this gives, for practical purposes, the same benefits of reducing the necessary filter length.
  • the delay is generally imperceptible; typically, up to 1.6 ms.
  • a fixed delay sufficiently long to guarantee causality, may be used, thus removing the need to encode the delay parameter.
  • the filter parameters are transmitted as difference signal data. With 16 bits per coefficient, this meant that a transmission capacity of 5120 bits/sec is needed for the difference channel (plus 8 bits for the delay parameter). This is well within the capacity of a standard 64 kbit/sec transmission system used which allocates 48 kbits/sec to the sum channel (efficiently transmitted by an existing monophonic encoding technique) and offers 16 kbits/sec for other "overhead" data.
  • This mode of the embodiment gives a good signal to noise ratio and the stereo image is present, although it is highly dependent on the accuracy of the algorithm used to adapt the predictive filter. Inaccuracies tend to cause the stereo image to wander during the course of a conference particularly when the conversation is passed from one speaking person to another at some distance from the first.
  • the residual signal is transmitted as difference signal data.
  • the sum signal is encoded (12a) using, for example, sub-band coding. It is also locally decoded (13a) to provide a signal equivalent to that at the decoder, for input to adaptive filter 4.
  • the residual difference channel is also encoded (possibly including bandlimiting) by residual coder 12b, and a corresponding local decoder 13b provides the signal minimised to adapt filter 4.
  • the analysis filter parameters are recovered from the transmitted residual by using a backwards-adapting replica filter 5 of the adaptive filter 4 at the coder.
  • Decoders 13c, 13d are identical to local decoders 13a, 13b and so the filter 5 receives the same inputs, and thus produces the same parameters, as that of encoder filter 4.
  • both filter parameters and residual signal are transmitted as side-information, overcoming many of the problems with the residual-only embodiment because the important stereo information in the first 2 kHz is preserved intact and the relative amplitude information at higher frequencies is largely retained by the filter parameters.
  • the parameter-only embodiment described above preferably uses a single adaptive filter 4 to remove redundancy between the sum and difference channels.
  • An effect discovered during testing was a curious 'whispering' effect if the coefficients were not sent at a certain rate, which was far above what should have been necessary to describe changes in the acoustic environment. This was because the adaptive filter, in addition to modelling the room acoustic transfer function, was also trying to perform an LPC analysis of the speech.
  • the adaptive filter 4 which models the acoustic transfer functions may be the same as before (for example, a lattice filter of order 10).
  • the sum channel is passed through a whitening filter 14a (which may be lattice or a simple transversal structure).
  • the master whitening filter 14a receives the sum channel and adapts to derive an approximate spectral inverse filter to the sum signal (or, at least, the speech components thereof) by minimising its own output.
  • the output of the. filter 14a is therefore substantially white.
  • the parameters derived by the master filter 14a are supplied to the slave whitening filter 14b, which is connected to receive and filter the difference signal.
  • the output of the slave whitening filter 14b is therefore the difference signal filtered by the inverse of the sum signal, which substantially removes common signal components, reducing the correlation between the two and leaving the output of 14b as consisting primarily of the acoustic response of the room. It thus reduces the dynamic range of the residual considerably.
  • the effect is to whiten the sum channel and to partially whiten the difference channel without affecting the spectral differences between them as a result of room acoustics, so that the derived coefficients of adaptive filter 4 are model parameters of the room acoustics.
  • the coefficients only are transmitted and the decoder is simply that of Figure 2 (needing no further filters).
  • residual encoder 12b and decoder 13b are omitted.
  • An adaptive filter will generally not be long enough to filter out long-term information, such as pitch information in speech, so the sum channel will not be completely "white”.
  • a long-term predictor known in LPC coding
  • filter 4 could, in principle, be connected to filter the difference channel alone, and thus to model the inverse of the room acoustic.
  • this second aspect of the invention reduces the dynamic range of the residual, it is particularly advantageous to employ this whitening scheme with the residual-only transmission described above.
  • an adaptive whitening filter 24a (identical to 14a at the encoder) receives the (decoded) sum channel and adapts to whiten its output.
  • a slave filter 24b (identical to 14b at the encoder) receives the coefficients of 24a.
  • adaptive filter 5 regenerates a filtered signal which is added to the (decoded) residual and the sum is filtered by slave filter 24b to yield the difference channel.
  • the sum and difference channels are then processed (6, 7 not shown) to yield the original left and right channels.
  • both residual and coefficients are transmitted.
  • the residual will have a bandwidth of 8 kHz and must be quantised and transmitted using spare channel capacity of about 16 kbit/s.
  • the whitened residual will be, in principle, small in mean square value, but will not be optimally whitened since the copy pre-whitening filter 14b through which the residual passes has coefficients derived to whiten the sum channel and not necessarily the difference channel.
  • the dynamic range of the filtered signal is reduced by 12dB over the unfiltered difference channel.
  • One approach to this residual quantisation problem is to reduce the bandwidth of the residual signal. This allows downsampling to a lower rate, with a consequential increase in bits per sample.
  • the structure uses a lattice filter 14a to pre-whiten the spectrum of the primary input
  • the decorrelated backwards residual outputs are then used as inputs to a simple linear combiner which attempts to model the input spectrum of the secondary input.
  • the modelling process is the same as with the simple transversal FIR filter, the effect of the lattice filter is to point the error vector in the direction of the optimum LMS residual solution. This speeds convergence considerably.
  • a lattice filter of order 20 is found effective in practice.
  • the lattice filter structure is particularly useful as described above, but could also be used in a system in which, instead of forming sum and difference signals, a (suitably delayed) left channel is predicted from the right channel.
  • the invention is implemented by forming a sum signal and 3 difference signals, and predicting each from the sum signal as above.

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Multimedia (AREA)
  • Computational Linguistics (AREA)
  • Mathematical Physics (AREA)
  • Macromolecular Compounds Obtained By Forming Nitrogen-Containing Linkages In General (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Treatments For Attaching Organic Compounds To Fibrous Goods (AREA)
  • Polymers With Sulfur, Phosphorus Or Metals In The Main Chain (AREA)
  • Stereophonic System (AREA)
  • Stereo-Broadcasting Methods (AREA)
  • Preparation Of Compounds By Using Micro-Organisms (AREA)
  • Separation Using Semi-Permeable Membranes (AREA)

Claims (12)

  1. Polyphonsignalkodierungsvorrichtung, die umfaßt:
    Empfangsvorrichtung für einen ersten (xS) und wenigstens einen zweiten Kanal (x0);
    Vorrichtung (4) für die periodische Generierung von Rekonstruktionsdaten, die aufgrund des ersten Kanals die Abschätzung des zweiten Kanals ermöglichen, wobei die Generatorvorrichtung (4) dazu dient, eine Vielzahl von Filterkoeffizienten zu erzeugen, welche bei Anwendung in einem Prädiktorfilter multipler Ordnung die Vorhersage des zweiten Kanals aufgrund des so gefilterten ersten Kanals ermöglichen,
    Ausgabevorrichtung für Daten, die den besagten ersten Kanal repräsentieren, und der Rekonstruktionsdaten,
    dadurch gekennzeichnet, daß
    die Vorrichtung außerdem umfaßt:
    Filtervorrichtung für den ersten und zweiten Kanal in Übereinstimmung mit einem Filter (14a, 14b), das sich dem spektralen Inversen des ersten Kanals nähert, um jeweils gefilterte Kanäle zu erzeugen, wobei der erste besagte gefilterte Kanal im wesentlichen spektral weißgefärbt wird und die Generatorvorrichtung (4) so geschaltet ist, daß sie die gefilterten Kanäle empfängt.
  2. Vorrichtung nach Anspruch 1, bei welcher die Generatorvorrichtung ein adaptives Filter (4) umfaßt, das so geschaltet ist, daß es den ersten Kanal empfängt und daraus einen vorhergesagten zweiten Kanal erstellt; und Vorrichtung für die Erzeugung eines Restsignals, das die Differenz zwischen dem besagten vorhergesagten zweiten Kanal und dem tatsächlichen zweiten Kanal darstellt, und bei welchem die besagten Rekonstruktionsdaten Daten umfassen, die das besagte Restsignal darstellen.
  3. Vorrichtung nach Anspruch 1 oder 2, wobei die Rekonstruktionsdaten die besagten Filterkoeffizienten (hi) umfassen.
  4. Vorrichtung nach Anspruch 2, bei welcher das adaptive Filter (4) nur durch das besagte Restsignal gesteuert wird und die besagten Rekonstruktionsdaten aus dem besagten Restsignal bestehen.
  5. Vorrichtung nach einem der Ansprüche 1 bis 4, wobei die besagte Filtervorrichtung einen adaptiven Hauptfilter (14a), der dazu dient, den ersten Kanal zu filtern, um einen weißgefärgten Ausgang zu erzeugen, und einen Nebenfilter (14b) umfaßt, der dazu dient, den zweiten Kanal zu filtern, wobei der Nebenfilter so konfiguriert ist, daß er äquivalent auf den adaptiven Filter der Filtervorrichtung reagiert.
  6. Vorrichtung nach einem der Ansprüche 1 bis 5, die außerdem umfaßt:
    Eingangsvorrichtung für den Empfang von Eingangssignalen; und
    Vorrichtung (2, 3) für die Erzeugung von deren Kanälen, wobei der erste Kanal ein Summenkanal ist, der die Summe der Eingangssignale darstellt, und der zweite oder weitere Kanäle die Differenzen dazwischen darstellen.
  7. Vorrichtung nach einem der Ansprüche 1 bis 6, die variable Verzögerungsvorrichtungen für die Verzögerung wenigstens eines der Kanäle und eine Vorrichtung für die Steuerung der differentiellen Verzögerung umfaßt, die auf die Kanäle angewendet werden, um die Korrelation oberhalb der Generatorvorrichtung zu erhöhen, wobei die Ausgangsvorrichtung dazu dient, um auch Daten auszugeben, die die besagte differentielle Verzögerung darstellt.
  8. Vorrichtung nach Anspruch 6, bei welcher die Eingangsvorrichtung variable Verzögerungsvorrichtungen (10a, 10b) für die Verzögerung wenigstens eines der Eingangssignale und eine Vorrichtung (9) für die Steuerung der differentiellen Verzögerung umfaßt, die auf die Signale angewendet wird, um die Korrelation oberhalb der Erzeugungsvorrichtung zu erhöhen, wobei die Ausgangsvorrichtung dazu dient, auch Daten auszugeben, die die besagte differentielle Verzögerung darstellen.
  9. Polyphonsignaldekodiervorrichtung, die umfaßt:
    Vorrichtung für den Empfang von Daten, die ein Summensignal darstellen, und Signalrekonstruktionsdaten; und Vorrichtung, die dazu dient, auf die Rekonstruktionsdaten hin das Summensignal zu modifizieren, um wenigstens zwei Ausgangssignale zu erzeugen, wobei die Modifizierungsvorrichtung umfaßt:
    ein konfigurierbares Prädiktorfilter multipler Ordnung (5) für den Empfang der besagten Signalrekonstruktionsdaten und Modifizierung der Koeffizienten in Übereinstimmung damit, wobei das Filter so geschaltet ist, daß es das besagte Summensignal empfängt und daraus das Ausgangsdifferenzsignal rekonstruiert; und
    Vorrichtung (6) für das Addieren des rekonstruierten Differenzsignals auf das Summensignal, und (7) für die Subtraktion des rekonstruierten Differenzsignals von dem Summensignal, um wenigstens zwei Ausgangssignale zu erzeugen,
    gekennzeichnet durch
    einen adaptiven Hauptfilter (24a) zum Filtern des Summensignals in Übereinstimmung mit in etwa dem spektral Inversen des Summensignals, um einen weißgefärbten Ausgang zu erzeugen, und einen Nebenfilter (24b) zum Filtern des Differenzsignals, wobei der Nebenfilter so konfiguriert ist, daß er eine äquivalente Antwortfunktion zum adaptiven Hauptfilter hat.
  10. Vorrichtung nach Anspruch 9, bei welcher die Differenzsignalrekonstruktionsdaten Restsignaldaten umfassen und die Vorrichtung eine Vorrichtung zur Addition der Restsignaldaten zu dem Ausgang des Filters umfaßt, um das rekonstruierte Differenzsignal zu bilden.
  11. Vorrichtung nach Anspruch 10, bei welcher das Prädiktorfilter (5) so geschaltet ist, daß es die Restsignaldaten empfängt und seine Koeffizienten in Übereinstimmung damit modifiziert.
  12. Verfahren zur Kodierung polyphoner Eingangssignale, das umfaßt:
    Erzeugung eines Summensignals daraus, das die Summe solcher Signale darstellt; und Rekonstruktionsdaten, so daß die Bildung eines weiteren der Eingangssignale aus dem Summensignal ermöglicht wird,
    Erzeugung mindestens eines Differenzsignals aus den Eingangssignalen, das die Differenz dazwischen darstellt;
    Analysieren des besagten Summen- und Differenzsignals und Erzeugung einer Vielzahl von Koeffizienten daraus, welche bei Anwendung in einem mehrstufigen Prädiktorfilter die Vorhersage des Differenzsignals aufgrund des so gefilterten Summensignals ermöglichte;
    wobei der kodierte Ausgang das besagte Summensignal und die Daten umfaßt, die die Rekonstruktion des besagten Differenzsignals daraus ermöglichen,
    gekennzeichnet durch
    Filtern des Summensignals und Differenzsignals in Übereinstimmung mit einem Filter, das sich dem spektral Inversen des Summensignals annähert, vor dem Analysieren, wobei das Summensignal dadurch spektral im wesentlichen weißgefärt wird.
EP90909155A 1989-06-15 1990-06-15 Polyphonische kodierung Expired - Lifetime EP0478615B2 (de)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
GB8913758 1989-06-15
GB898913758A GB8913758D0 (en) 1989-06-15 1989-06-15 Polyphonic coding
PCT/GB1990/000928 WO1990016136A1 (en) 1989-06-15 1990-06-15 Polyphonic coding

Publications (3)

Publication Number Publication Date
EP0478615A1 EP0478615A1 (de) 1992-04-08
EP0478615B1 EP0478615B1 (de) 1995-04-26
EP0478615B2 true EP0478615B2 (de) 1998-04-15

Family

ID=10658483

Family Applications (1)

Application Number Title Priority Date Filing Date
EP90909155A Expired - Lifetime EP0478615B2 (de) 1989-06-15 1990-06-15 Polyphonische kodierung

Country Status (13)

Country Link
EP (1) EP0478615B2 (de)
JP (1) JP2703405B2 (de)
AT (1) ATE121900T1 (de)
AU (1) AU640667B2 (de)
CA (1) CA2058984C (de)
DE (1) DE69018989T3 (de)
DK (1) DK0478615T3 (de)
ES (1) ES2071823T3 (de)
FI (1) FI915873A0 (de)
GB (1) GB8913758D0 (de)
HK (1) HK137196A (de)
NO (1) NO180030C (de)
WO (1) WO1990016136A1 (de)

Families Citing this family (25)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5274740A (en) * 1991-01-08 1993-12-28 Dolby Laboratories Licensing Corporation Decoder for variable number of channel presentation of multidimensional sound fields
KR100228688B1 (ko) * 1991-01-08 1999-11-01 쥬더 에드 에이. 다차원 음장용 인코우더/디코우더
DE4136825C1 (de) * 1991-11-08 1993-03-18 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung Ev, 8000 Muenchen, De
US5278909A (en) * 1992-06-08 1994-01-11 International Business Machines Corporation System and method for stereo digital audio compression with co-channel steering
EP0608937B1 (de) * 1993-01-27 2000-04-12 Koninklijke Philips Electronics N.V. Tonsignalverarbeitungsanordnung zur Ableitung eines Mittelkanalsignals und audiovisuelles Wiedergabesystem mit solcher Verarbeitungsanordnung
DE4320990B4 (de) * 1993-06-05 2004-04-29 Robert Bosch Gmbh Verfahren zur Redundanzreduktion
US5488665A (en) * 1993-11-23 1996-01-30 At&T Corp. Multi-channel perceptual audio compression system with encoding mode switching among matrixed channels
GB2303516A (en) * 1995-07-20 1997-02-19 Plessey Telecomm Teleconferencing
DE19526366A1 (de) 1995-07-20 1997-01-23 Bosch Gmbh Robert Verfahren zur Redundanzreduktion bei der Codierung von mehrkanaligen Signalen und Vorrichtung zur Dekodierung von redundanzreduzierten, mehrkanaligen Signalen
US6016473A (en) * 1998-04-07 2000-01-18 Dolby; Ray M. Low bit-rate spatial coding method and system
DE19829284C2 (de) * 1998-05-15 2000-03-16 Fraunhofer Ges Forschung Verfahren und Vorrichtung zum Verarbeiten eines zeitlichen Stereosignals und Verfahren und Vorrichtung zum Decodieren eines unter Verwendung einer Prädiktion über der Frequenz codierten Audiobitstroms
SE519552C2 (sv) * 1998-09-30 2003-03-11 Ericsson Telefon Ab L M Flerkanalig signalkodning och -avkodning
SE519981C2 (sv) 2000-09-15 2003-05-06 Ericsson Telefon Ab L M Kodning och avkodning av signaler från flera kanaler
SE519976C2 (sv) * 2000-09-15 2003-05-06 Ericsson Telefon Ab L M Kodning och avkodning av signaler från flera kanaler
SE519985C2 (sv) * 2000-09-15 2003-05-06 Ericsson Telefon Ab L M Kodning och avkodning av signaler från flera kanaler
FR2821475B1 (fr) * 2001-02-23 2003-05-09 France Telecom Procede et dispositif de reconstruction spectrale de signaux a plusieurs voies, notamment de signaux stereophoniques
MX2007005261A (es) * 2004-11-04 2007-07-09 Koninkl Philips Electronics Nv Codificacion y descodificacion de un conjunto de senales.
US9197977B2 (en) * 2007-03-01 2015-11-24 Genaudio, Inc. Audio spatialization and environment simulation
JPWO2009122757A1 (ja) * 2008-04-04 2011-07-28 パナソニック株式会社 ステレオ信号変換装置、ステレオ信号逆変換装置およびこれらの方法
WO2009141775A1 (en) 2008-05-23 2009-11-26 Koninklijke Philips Electronics N.V. A parametric stereo upmix apparatus, a parametric stereo decoder, a parametric stereo downmix apparatus, a parametric stereo encoder
EP2375409A1 (de) * 2010-04-09 2011-10-12 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Audiocodierer, Audiodecodierer und zugehörige Verfahren zur Verarbeitung von Mehrkanal-Audiosignalen mithilfe einer komplexen Vorhersage
KR101698439B1 (ko) 2010-04-09 2017-01-20 돌비 인터네셔널 에이비 Mdct-기반의 복소수 예측 스테레오 코딩
ES2911893T3 (es) 2010-04-13 2022-05-23 Fraunhofer Ges Forschung Codificador de audio, decodificador de audio y métodos relacionados para procesar señales de audio estéreo usando una dirección de predicción variable
UA107771C2 (en) * 2011-09-29 2015-02-10 Dolby Int Ab Prediction-based fm stereo radio noise reduction
US9380387B2 (en) 2014-08-01 2016-06-28 Klipsch Group, Inc. Phase independent surround speaker

Family Cites Families (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
AU546069B2 (en) * 1981-09-07 1985-08-15 Kahn, Leonard Richard Am stereo distortion correction
JPH0761043B2 (ja) * 1986-04-10 1995-06-28 株式会社東芝 ステレオ音声伝送蓄積方式
GB8628046D0 (en) * 1986-11-24 1986-12-31 British Telecomm Transmission system

Also Published As

Publication number Publication date
FI915873A7 (fi) 1991-12-13
AU640667B2 (en) 1993-09-02
EP0478615A1 (de) 1992-04-08
JPH04506141A (ja) 1992-10-22
CA2058984A1 (en) 1990-12-16
DE69018989T2 (de) 1995-09-07
AU5837990A (en) 1991-01-08
FI915873A0 (fi) 1991-12-13
ATE121900T1 (de) 1995-05-15
NO180030B (no) 1996-10-21
GB8913758D0 (en) 1989-08-02
CA2058984C (en) 1998-12-01
DK0478615T3 (da) 1995-07-17
EP0478615B1 (de) 1995-04-26
DE69018989T3 (de) 1998-11-19
ES2071823T3 (es) 1995-07-01
JP2703405B2 (ja) 1998-01-26
NO914947L (no) 1992-02-13
HK137196A (en) 1996-08-02
DE69018989D1 (de) 1995-06-01
NO914947D0 (no) 1991-12-13
WO1990016136A1 (en) 1990-12-27
NO180030C (no) 1997-01-29

Similar Documents

Publication Publication Date Title
US5434948A (en) Polyphonic coding
EP0478615B2 (de) Polyphonische kodierung
US5701346A (en) Method of coding a plurality of audio signals
US8873764B2 (en) Acoustic echo suppression unit and conferencing front-end
CA2645910C (en) Methods and apparatuses for encoding and decoding object-based audio signals
CA2903900C (en) Apparatus and method for multichannel direct-ambient decomposition for audio signal processing
KR100931309B1 (ko) Ｈｅ－ａａｃ 디코더와 ｍｐｅｇ 서라운드 디코더를이용하여 멀티채널 오디오 신호를 복원하는 장치 및 방법
RU2420026C2 (ru) Способы и устройства для кодирования и декодирования аудиосигналов на основе объектов
US7725324B2 (en) Constrained filter encoding of polyphonic signals
US7024008B2 (en) Acoustic quality enhancement via feedback and equalization for mobile multimedia systems
US6012025A (en) Audio coding method and apparatus using backward adaptive prediction
Crochiere et al. Frequency domain techniques for speech coding
JP4323520B2 (ja) ポリフォニック信号の制約付きフィルタ符号化
Minami et al. Stereophonic adpcm voice coding method

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

17P Request for examination filed

Effective date: 19911206

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AT BE CH DE DK ES FR GB IT LI LU NL SE

17Q First examination report despatched

Effective date: 19940203

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): AT BE CH DE DK ES FR GB IT LI LU NL SE

REF Corresponds to:

Ref document number: 121900

Country of ref document: AT

Date of ref document: 19950515

Kind code of ref document: T

ITF It: translation for a ep patent filed
PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: AT

Payment date: 19950515

Year of fee payment: 6

REF Corresponds to:

Ref document number: 69018989

Country of ref document: DE

Date of ref document: 19950601

REG Reference to a national code

Ref country code: ES

Ref legal event code: FG2A

Ref document number: 2071823

Country of ref document: ES

Kind code of ref document: T3

REG Reference to a national code

Ref country code: DK

Ref legal event code: T3

ET Fr: translation filed
PLBQ Unpublished change to opponent data

Free format text: ORIGINAL CODE: EPIDOS OPPO

PLBI Opposition filed

Free format text: ORIGINAL CODE: 0009260

26 Opposition filed

Opponent name: PHILIPS ELECTRONICS N.V.

Effective date: 19951218

PLBF Reply of patent proprietor to notice(s) of opposition

Free format text: ORIGINAL CODE: EPIDOS OBSO

NLR1 Nl: opposition has been filed with the epo

Opponent name: PHILIPS ELECTRONICS N.V.

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: LU

Payment date: 19960601

Year of fee payment: 7

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: AT

Effective date: 19960615

PLBF Reply of patent proprietor to notice(s) of opposition

Free format text: ORIGINAL CODE: EPIDOS OBSO

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DK

Payment date: 19970513

Year of fee payment: 8

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: SE

Payment date: 19970520

Year of fee payment: 8

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: CH

Payment date: 19970603

Year of fee payment: 8

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: ES

Payment date: 19970612

Year of fee payment: 8

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LU

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 19970615

PLAW Interlocutory decision in opposition

Free format text: ORIGINAL CODE: EPIDOS IDOP

PLAW Interlocutory decision in opposition

Free format text: ORIGINAL CODE: EPIDOS IDOP

PUAH Patent maintained in amended form

Free format text: ORIGINAL CODE: 0009272

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: PATENT MAINTAINED AS AMENDED

27A Patent maintained in amended form

Effective date: 19980415

AK Designated contracting states

Kind code of ref document: B2

Designated state(s): AT BE CH DE DK ES FR GB IT LI LU NL SE

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: SE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 19980616

Ref country code: ES

Free format text: LAPSE BECAUSE OF THE APPLICANT RENOUNCES

Effective date: 19980616

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 19980630

Ref country code: CH

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 19980630

NLR2 Nl: decision of opposition
ITF It: translation for a ep patent filed
PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 19980715

NLR3 Nl: receipt of modified translations in the netherlands language after an opposition procedure
ET3 Fr: translation filed ** decision concerning opposition
REG Reference to a national code

Ref country code: CH

Ref legal event code: PL

REG Reference to a national code

Ref country code: ES

Ref legal event code: FD2A

Effective date: 20001009

REG Reference to a national code

Ref country code: GB

Ref legal event code: IF02

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: NL

Payment date: 20020522

Year of fee payment: 13

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: BE

Payment date: 20020603

Year of fee payment: 13

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: BE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20030630

BERE Be: lapsed

Owner name: BRITISH *TELECOMMUNICATIONS P.L.C.

Effective date: 20030630

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: NL

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20040101

NLV4 Nl: lapsed or anulled due to non-payment of the annual fee

Effective date: 20040101

PLAB Opposition data, opponent's data or that of the opponent's representative modified

Free format text: ORIGINAL CODE: 0009299OPPO

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 20090615

Year of fee payment: 20

Ref country code: IT

Payment date: 20090626

Year of fee payment: 20

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: GB

Payment date: 20090618

Year of fee payment: 20

Ref country code: DE

Payment date: 20090622

Year of fee payment: 20

REG Reference to a national code

Ref country code: GB

Ref legal event code: PE20

Expiry date: 20100614

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: GB

Free format text: LAPSE BECAUSE OF EXPIRATION OF PROTECTION

Effective date: 20100614

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DE

Free format text: LAPSE BECAUSE OF EXPIRATION OF PROTECTION

Effective date: 20100615