PatentDe  


Dokumentenidentifikation EP1473710 19.04.2007
EP-Veröffentlichungsnummer 0001473710
Titel Verfahren und Vorrichtung zur Audiokodierung mittels einer mehrstufigen Mehrimpulsanregung
Anmelder NEC Corp., Tokyo, JP
Erfinder Toshiyuki, Nomura, Tokyo, JP
Vertreter derzeit kein Vertreter bestellt
DE-Aktenzeichen 69837296
Vertragsstaaten DE, FR, GB, IT, NL, SE
Sprache des Dokument EN
EP-Anmeldetag 02.04.1998
EP-Aktenzeichen 040902223
EP-Offenlegungsdatum 03.11.2004
EP date of grant 07.03.2007
Veröffentlichungstag im Patentblatt 19.04.2007
IPC-Hauptklasse G10L 19/10(2006.01)A, F, I, 20051017, B, H, EP

Beschreibung[en]
BACKGROUND OF THE INVENTION Field of the Invention:

The present invention relates to an audio encoding apparatus according to the preamble of claim 1 and a hierarchical encoding method according to the preamble of claim 4.

Description of the Prior Art:

Heretofore, the aim of introducing an audio encoding apparatus and decoding apparatus which adapt the hierarchical multistage encoding method which enables decoding audio signals from a part of bitstream of encoded signals as well as all of it, is to cope with the case that a part of packets of encoded audio signals is lost in a packet transmission network. An example of such apparatus based on CELP (Code Excited Linear Prediction) encoding method comprises excitation signal encoding blocks in a multistage connection. This is disclosed in "Embedded CELP coding for variable bit-rate between 6.4 and 9.6 kbit/s" by R. Drog in proceedings of ICASSP, pp. 681-684, 1991 and "Embedded algebraic CELP coders for wideband speech coding" by A. Le Guyader, et. al. in proceedings of EUSIPCO, signal processing VI, pp. 527-530, 1992.

In this document, the CELP excitation signal is progressively modelled using an increasing number of waveforms depending on the available bit rate. The waveforms are generated by a plurality of algebraic codebooks corresponding to each coding stage.

Another example of multistage coding using a cascaded arrangement of regular-pulse setting excitation determination units is disclosed in the document S.Zhang and G.Lockhart "An embedded scheme for regular pulse excited (RPE) linear prediction coding", Proceedings of the ICASSP 1995, pp. 37-40. This document shows how the coding error is progressively modelled by a set of pulses, and how an updated coding error signal is generated as a result of said coding which is to be the subject of a further coding in a subsequent stage.

With reference to Figs. 2A and 2B, the operation of an example of the prior art will be explained. Although only two excitation signal encoding blocks are connected in the example for simplicity, the following explanation can be extended to the structure of three or more stages.

Frame dividing circuit 101 divides an input signal into frames and supplies the frames to sub-frame dividing circuit 102.

Sub-frame dividing circuit 102 divides the input signal in a frame into sub-frames and supplies the sub-frames to linear-predictive analysis circuit 103 and psychoacoustic weighting signal generating circuit 105.

Linear predictive analyzing circuit 103 applies linear predictive analysis to each sub-frame of the input from sub-frame dividing circuit 102 and supplies linear predictor coefficients a(i) (i = 1, 2, 3, ···, Np) to linear predictor coefficient quantizing circuit 104, psychoacoustic weighting signal generating circuit 105, psychoacoustic weighting signal reproducing circuit 106, adaptive codebook searching circuit 109, multi-pulse searching circuit 110, and auxiliary multi-pulse searching circuit 112. Number Np in the former sentence represents the degree of linear predictive analysis and, for example takes value 10. There are the correlation method and the covariance method among linear predictive analysis and they are explained in detail in chapter five of "Digital Audio Processing" published by Tohkai University Press in Japan.

Linear predictor coefficient quantizing circuit 104 quantizes the linear predictor coefficients for each frame instead of sub-frame. In order to decrease bitrate, it is common to adapt the method in which only the last sub-frame in the present frame is quantized and the rest subframes of the sub-frames in the frame are interpolated using the quantized linear predictor coefficients of the present frame and the preceding frame. The quantization and interpolation are executed after converting linear predictor coefficients to line spectrum pairs (LSP). The conversion from linear predictor coefficients to LSP is explained in "Speech data Compression by LSP Speech Analysis-Synthesis Technique" in Journal of the Institute of Electronics, Information and Communication Engineers, J64-A, pp. 599 - 606, 1981. Well-known methods can be used for quantizing LSP. One example of such methods is explained in Japanese Patent Laid-open 4-171500.

After converting quantized LSPs to quantized linear predictor coefficients a' (i) (i = 1,2,3, ···, Np), linear predictive coefficient quantizing circuit 104 supplies the quantized linear predictor coefficients to a psychoacoustic weighting signal reproducing circuit 106, adaptive codebook searching circuit 109, multi-pulse searching circuit 110, and auxiliary multi-pulse searching circuit 112 and supplies indices representing quantized LSPs to multiplexer 114.

Psychoacoustic weighting signal generating circuit 105 drives the psychoacoustically weighting filter Hw(z) represented by equation (1) by an input signal in a sub-frame to generate a psychoacoustically weighted signal which is supplied to target signal generating circuit 108: Hw z = 1 - i = 1 N p &agr; i R 2 i z - i 1 - i = 1 N p &agr; i R 1 i z - i

where R1 and R2 are weighting coefficients which control the amount of psychoacoustic weighting. For example, R1 = 0.6 and R2 = 0.9.

Psychoacoustic weighting signal reproducing circuit 106 drives a psychoacoustically weighting synthesis filter by an excitation signal of the preceding sub-frame which is supplied via sub-frame buffer 107. The psychoacoustic weighting synthesis filter consists of a linear predictive synthesis filter represented by equation (2) and a psychoacoustically weighting filter Hw (z) in cascade connection whose coefficients are of the preceding sub-frame and have been held therein: H s z = 1 1 - i = 1 N p &agr;ʹ i z - i

After the driving, the psychoacoustic weighting signal reproducing circuit 106 drives the psychoacoustically weighting synthesis filter by a series of zero signals to calculate the response to zero inputs. The response is supplied to the target signal generating circuit 108.

Target signal generating circuit 108 subtracts the response to zero inputs from the psychoacoustic weighting signal to get target signals X(n) (n = 0, 1, 2, ···, N-1). Number N in the former sentence represents the length of a sub-frame. Target signal generating circuit 108 supplies the target signals to an adaptive codebook searching circuit 109, a multi-pulse searching circuit 110, a gain searching circuit 111, an auxiliary multi-pulse searching circuit 112, and an auxiliary gain searching circuit 113.

Using the excitation signal of the preceding sub-frame supplied through sub-frame buffer 107, the adaptive codebook searching circuit 109 renews an adaptive codebook which has held past excitation signals. Adaptive vector signal Ad(n) (n = 0,1,2,···,N-1) corresponding to pitch d is a signal delayed by pitche d which has been stored in the adaptive codebook. Here, if pitch d is longer than the length of a sub-frame N, adaptive codebook searching circuit 109 detaches d samples just before the present sub-frame and repeatedly connects the detached samples until the number of the samples reaches the length of a sub-frame N. Adaptive codebook searching circuit 109 drives the psychoacoustic weighting synthesis filter which is initialized for each sub-frame (hereinafter referred to as a psychoacoustic weighting synthesis filter in zero-state) by the generated adaptive code vector Ad(n) (n = 0,1,2,···,N-1) to generate reproduced signals SAd(n) (n = 0,1,2,···,N-1) and selects pitch d' which minimizes error E(d), which is the difference between target signals X (n) and SAd(n), from a group of d within predetermined searching range, for example d = 17, ···,144. Hereinafter the selected pitch d' will be referred to as d for simplicity. E d = n = 1 N X n 2 - ( n = 1 N X n SAd n ) 2 n = 1 N SAd ( n ) 2

Adaptive codebook searching circuit 109 supplies the selected pitch d to multiplexer 114, the selected adaptive code vector Ad (n) to gain searching circuit 111, and the regenerated signals SAd(n) to gain searching circuit 111 and multi-pulse searching circuit 110.

Multi-pulse searching circuit 110 searches for P pieces of non-zero pulse which constitute a multi-pulse signal. Here, the position of each pulse is limited to the pulse position candidates which were determined in advance. The pulse position candidates for a different non-zero pulse are different from one another. The non-zero pulses are expressed only by polarity. Therefore, the coding the multi-pulse signal is equivalent to selecting index j which minimizes error E(j) in equation (4): E j = n = 1 N n 2 - ( n = 1 N n SC j n ) 2 n = 1 N SC j ( n ) 2

where SCj (n) (n = 0, 1, 2,···,N-1) is a reproduced signal obtained by driving the psychoacoustic weighting synthesis filter in zero-state by multi-pulse signals Cj (n = 0, 1, 2,···, N-1) which is constituted for index j (j = 0, 1, 2,···, J-1) which represents one of J pieces of combination of the pulse position candidate and the polarity, and X' (n) (n = 0,1,2,···,N-1) is a signal obtained by orthogonalizing the target signal X(n) by the reproduced signal SAd(n) of the adaptive code vector signal and given by equation (5): n = X n - n = 1 N X n SAd n n = 1 N SAd ( n ) 2 SAd n

This method is explained in detail in "Fast CELP coding based on algebraic codes" in proceedings of ICASSP, pp. 1957-1960, 1987.

Index j representing the multi-pulse signal can be transmitted with p = 0 p - 1 1 + log 2 M p bits where M (p) (p 0,1,2,···,P-1) is the number of the pulse position candidates for p-th pulse. For example, the number of bits necessary to transmit index j is 20 provided that sampling rate is 8 kHz, the length of a sub-frame is 5 msec (N = 40 samples), the number of pulses P is five, the number of the pulse position candidates M (p) = 8, p = 0,1,2,···, P-1, and the number of the pulse position candidates is, for simplicity, constant.

Multi-pulse searching circuit 110 supplies selected multi-pulse signal Cj (n) and the reproduced signal SCj(n) for the multi-pulse signal to gain searching circuit 111 and corresponding index j to multiplexer 114.

Gain searching circuit 111 searches for the optimum gain consisting of GA(k) and GE(K) (k = 0, 1, 2,···,K-1) for a pair of the adaptive code vector signal and the multi pulse signal from a gain codebook of size K. Index k of the optimum gain is selected so as to minimize error E(k) in equation (6): E k = n = 1 N [ X n - GA k SAd n - GE k SCj n ] 2

where X(n) is the target signal, SAd(n) is the reproduced adaptive code vector, and SCj(n) is the reproduced multi-pulse signal.

Gain searching circuit 111 also generates excitation signal D(n) (n=0,1,2,···,N-1) using the selected gain, the adaptive code vector, and the multi-pulse pulse signal. Excitation signal D(n) is supplied to sub-frame buffer 107 and auxiliary multi-pulse searching circuit 112. Moreover, gain searching circuit 111 drives the psychoacoustic weighting filter in zero-state by excitation signal D(n) to generate reproduced excitation signal SD(n) (n = 0, 1, 2, ···, N-1) which is supplied to auxiliary multi-pulse searching circuit 112, auxiliary gain searching circuit 113, and multiplexer 114.

Similarly to multi-pulse searching circuit 110, auxiliary multi-pulse searching circuit 112 generates auxiliary multi-pulse signal Cm(n) (n = 0, 1, 2, ···, N-1) and regenerated auxiliary multi-pulse signal SCm(n) (n=0, 1,2,···,N-1) and selects m which minimizes error E(m) in equation (7): E m = n = 1 N n 2 - ( n = 1 N n SCm n ) 2 n = 1 N SCm ( n ) 2

where X"(n) (n = 0, 1, 2, ···, N-1) is a signal obtained by orthogonalizing target signal X(n) by reproduced signal SD(n) of the excitation signal and given by equation (8): n = X n - n = 1 N X n SD n n = 1 N SD ( n ) 2 SD n

Index m representing multi-pulse signal C(m) can be transmitted with p = 0 - 1 1 + log 2 p bits where P' is the number of auxiliary multi-pulse signals and M' (p) (p = 0,1, 2, ···, P'-1) is the number of the pulse position candidates for p-th pulse. For example, the number of bits necessary to transmit index m is 20 provided that the number of pulses P' is five, the number of the pulse position candidates for each pulse M'(p) is 8, p= 0,1,2,···,P'-1, and the number of the pulse position candidates is, for simplicity, constant.

Auxiliary multi-pulse searching circuit 112 also supplies regenerated signal SCm(n) to auxiliary gain searching circuit 113 and corresponding index m to multiplexer 114.

Auxiliary gain searching circuit 113 searches for the optimum gain consisting of GEA(1) and GEC(1) (1 = 0, 1, 2, ···, K'-1) for a pair of the excitation signal and the auxiliary multi-pulse signal from a gain codebook of size K'. Index l of the optimum gain is selected so as to minimize error E (1) in equation (9) : E l = n = 1 N [ X n - GEA l Sd n - GEC l SCm n ] 2

where X(n) is the target signal, SD(n) is the reproduced excitation signal, and SCm(n) is the reproduced auxiliary multi-pulse signal.

Selected index l is supplied to multiplexer 114.

Multiplexer 114 converts indices, which correspond to the quantized LSP, the adaptive code vector, the multi-pulse signal, the gains, the auxiliary multi-pulse signal and the auxiliary gains, into a bitstream which is supplied to first output terminal 115.

Bitstream from second input terminal 116 is supplied to demultiplexer 117. Demultiplexer 117 converts the bitstream into the indices which correspond to the quantized LSP, the adaptive code vector, the multi-pulse signal, the gains, the auxiliary multi-pulse signal and the auxiliary gains. Demultiplexer 117 also supplies the index of the quantized LSP to linear predictor coefficient decoding circuit 118, the index of the pitch to adaptive codebook decoding circuit 119, the index of the multi-pulse signal to multi-pulse decoding circuit 120, the index of the gains to gain decoding circuit 121, the index of the auxiliary multi-pulse signal to auxiliary multi-pulse decoding circuit 124, and the index of the auxiliary gains to auxiliary gain decoding circuit 125.

Linear predictor coefficient decoding circuit 118 decodes the index of the quantized LSP to quantized linear predictor coefficients a' (i) (i = 1, 2, 3, ···, Np) which is supplied to first signal reproducing circuit 122 and second signal reproducing circuit 126.

Adaptive codebook decoding circuit 119 decodes the index of the pitch to adaptive code vector Ad(n) which is supplied to gain decoding circuit 121. Multi-pulse decoding circuit 120 decodes the index of the multi-pulse signal to multi-pulse signal Cj(n) which is supplied to gain decoding circuit 121. Gain decoding circuit 121 decodes the index of the gains to gains GA(k) and GC(k) and generates a first excitation signal using gains GA(k) and GC (k), adaptive code vector Ad (n), multi-pulse signal Cj(n) and gains GA(k) and GC(k). The first excitation signal is supplied to first signal reproducing circuit 122 and auxiliary gain decoding circuit 125.

First signal reproducing circuit 122 generates a first reproduced signal by driving linear predictive synthesis filter Hs(z) with the first excitation signal. The first reproduced signal is supplied to second output terminal 123.

Auxiliary multi-pulse decoding circuit 124 decodes the index of the auxiliary multi-pulse signal to auxiliary multi-pulse signal Cm(n) which is supplied to auxiliary gain decoding circuit 125. Auxiliary gain decoding circuit 125 decodes the index of the auxiliary gains to auxiliary gains GEA(1) and GEC(1) and generates a second excitation signal using the first excitation signal, auxiliary multi-pulse signal Cm(n) and auxiliary gains GEA(1) and GEC(1).

Second signal reproducing circuit 126 generates a second reproduced signal by driving linear predictive synthesis filter Hs (z) with the second excitation signal. The second reproduced signal is supplied to third output terminal 127.

The conventional method explained above has a disadvantage that coding efficiency of a multi-pulse signal in the second stage and following stages is not sufficient because .. there is a possibility that each stage locates pulses in the same positions with those of pulses encoded in former stages. Because a multi-pulse signal is represented by positions and polarities of pulses, the same multi-pulse is formed when plural pulses are located in the same position and when one pulse is located therein. Therefore, coding efficiency is not improved when plural pulses are located in the same position.

US 5 193 140 discloses a multipulse determination method suitable for implementing the pulse searching stages according to the preamble of claim 1.

SUMMARY OF THE INVENTION

The object of the present invention is to provide an audio encoding apparatus and method which efficiently encode a multi-pulse in multiple stages.

The invention solves this object with the features of claims 1 and 4.

According to the present invention, the multi-pulse setting circuit (an auxiliary multi-pulse setting circuit) sets candidates for pulse positions so that pulse positions to which no pulse has been assigned are selected with priority over pulse positions at which pulses have been already encoded, and a multi-pulse searching circuit following the multi-pulse setting circuit selects pulse positions from the candidates and encodes the selected pulse positions. Thus, the multi-pulse searching circuit encodes the information concerning the selected pulse positions among candidates of pulse positions from which positions of already encoded pulses are excluded, whereby the required number of bits for the encoding can be reduced.

These and other objects, features and advantages of the present invention will become more apparent in light of the following detailed description of the best mode embodiments thereof, as illustrated in the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

  • Fig. 1A shows an audio encoding apparatus according to one embodiment of the present invention;
  • Fig. 1B shows an audio decoding apparatus;
  • Fig. 2A shows an audio encoding apparatus in the prior art; and
  • Fig. 2B shows an audio decoding apparatus in the prior art.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

Although only two excitation signal encoding blocks are connected in the apparatuses for simplicity, the following explanation can be extended to the structure of three or more stages.

Differences between the apparatuses according to Figs. 1A, 1B and the prior art shown in Figs. 2A, 2B are addition of multi-pulse setting circuits 130 and 132, replacement of auxiliary multi-pulse searching circuit 112 by auxiliary multi-pulse searching circuit 131, and replacement of auxiliary multi-pulse decoding circuit 124 by auxiliary multi-pulse decoding circuit 133. Therefore, only differences are explained as follows.

Auxiliary multi-pulse setting circuit 130 set candidates for pulse positions so that pulse positions to which no pulse has been assigned are selected in auxiliary multi-pulse searching circuit 131 prior to those of pulses already encoded in multi-pulse searching circuit 110. For example, auxiliary multi-pulse setting circuit 130 operates as follows: Auxiliary multi-pulse setting circuit 130 divides each sub-frame into Q pieces of sub-areas. One pulse is assigned to each sub-area. Candidates for the position of each pulse is the sub-area. Auxiliary multi-pulse setting circuit 130 selects a limited number of sub-areas from the top of the ascending order of the number of pulses already encoded therein, and outputs the indices of the selected sub-areas. The indices may be called the indices of pulses because the pulses and the sub-areas are connected biuniguely. Auxiliary multi-pulse setting circuit 130 has candidates for pulse positions X (q, r) (q= 0, 1,2,···,Q-1; r = 0,1, 2,···,M"(q)-1) for Q pieces of pulse in advance, where Q represents the number of pulses, q represents pulse number, M"(q) represents the total number of candidates for pulse positions corresponding to pulse q, and r represents serial number of a candidate of a pulse position. Here, the number of pulses Q, for example, 10, is different from the number of pulses of the multi-pulse signal, for example, five which is the same as the prior art. In this embodiment, M" (q) is constant and four, which is quotient of division of the length of sub-frame 40 by the number of pulses 10, for all the values of q. A candidate for a pulse position X(q, r) for a certain pair of q and r is different from that for another pair of q and r. Auxiliary multi-pulse setting circuit 130 comprises counters Ctr(q) (q = 0,1,2,···,Q-1) corresponding to Q pieces of pulses. The initial values of counters Ctr(q) are zero. Pulse number q is extracted by searching for one candidate of which position is the same as that of a pulse of the multi-pulse signal supplied from multi-pulse searching circuit 110 from candidates for pulse positions X(q,r). The counter Ctr(q) corresponding to the extracted pulse number q is incremented. The same operation is repeated for all the pulses supplied from multi-pulse searching circuit 110. Subsequently, Q', for example, five, pieces of counters are selected from the top in ascending order of count values. Serial numbers of selected counters are represented by s (t) (t = 0,1,2,···,Q'-1). Therefore, s (t) indicates one of pulse numbers ranging from zero to Q-1. In this meaning, s(t) may be called pulse number. In the selection, if plural counters take the same count value, for example the counter with minimum q is selected. Moreover, auxiliary multi-pulse setting circuit 130 supplies Q' pieces of selected pulse number s(t) (t = 0,1, 2,··· Q'-1) to auxiliary multi-pulse searching circuit 131.

Similarly to auxiliary multi-pulse setting circuit 130, auxiliary multi-pulse searching circuit 131 comprises candidates for pulse positions X(q,r) (q = 0,1,2,···,Q-1; r = 0, 1, 2,···, M"(q)-1) for Q pieces of pulse in advance. Auxiliary multi-pulse searching circuit 131 searches for Q' pieces of non-zero pulse constituting an auxiliary multi-pulse signal. Here, the position of each pulse is limited to within candidates for pulse position X(s(t),r) (r = 0, 1, 2, ···, M"(s(t))-1) in accordance with Q' pieces of pulse number s(t) (t = 0,1,2,···,Q'-1). Moreover, the amplitudes of the pulses are represented only by polarity. Therefore, encoding of the auxiliary multi-pulse is performed by constituting auxiliary multi-pulse signals Cm (n) (n = 0, 1, 2, ···, N-1) for index m which represents one of all the combinations of candidates for pulse position and polarities, driving the psychoacoustic weighting synthesis filter in zero-state with auxiliary multi-pulse signals Cm(n) so as to generate reproduced signals SCm(n) (n = 0, 1, 2, ···, N-1), and selecting index m which minimizes error E (m) represented by equation (7). Selected index m can be encoded and transmitted with t = 0 - 1 1 + log 2 s t bits. For example, substituting Q' = 5 and M" (s(t)) = 4 for the equation, the number of bit is 15. That is, the number of bits required to encode an auxiliary multi-pulse signal is 15. The corresponding number in the prior art is 20. Therefore, the number of bits is reduced by five. Auxiliary multi-pulse searching circuit 131 supplies reproduced auxiliary multi-pulse signal SCm(n) to auxiliary gain searching circuit 113 and corresponding index m to multiplexer 114.

Auxiliary multi-pulse setting circuit 132 in the audio decoding apparatus operates in the same way as auxiliary multi-pulse setting circuit 130 in the audio encoding apparatus. That is, auxiliary multi-pulse setting circuit 132 selects pulse numbers s(t) (t = 0,1,2,···,Q'-1) for Q' pieces of pulse in a multi-pulse supplied from multi-pulse decoding circuit 120, and supplies selected pulse numbers s(t) to auxiliary multi-pulse decoding circuit 133.

Auxiliary multi-pulse decoding circuit 133 reproduces the auxiliary multi-pulse signal using the index of the auxiliary multi-pulse signal supplied from demultiplexer 117 and pulse number s (t) (t = 0, 1, 2,···, Q'-1) selected in auxiliary multi-pulse setting circuit 132 and referring to candidates for each pulse position X(s(t),r) (r = 0, 1, 2,···,M"), and supplies the auxiliary multi-pulse signal to auxiliary gain decoding circuit 125.

As explained above, according to the audio encoding apparatus of the present invention, the efficiency of encoding a multi-pulse signal in a second stage and following stages in multistage connection can be improved because plural pulses constituting the multi-pulse signal are rarely located in the same position and the number of bits required for encoding can be reduced without deteriorating coding quality.


Anspruch[de]
Audiokodiervorrichtung zum Kodieren eines Mehrpulssignals, das ein Anregungssignal eines reproduzierten Audiosignals wiedergibt, in mehreren Stufen, indem mehrere Pulse Pulspositionen zugewiesen werden, die eine Differenz zwischen dem reproduzierten Audiosignal und einem eingegebenen Audiosignal minimieren, wobei das reproduzierte Audiosignal erhalten wird, indem ein linearer Vorhersage-Synthesefilter mittels des Anregungssignals getrieben wird,

wobei die Audiokodiervorrichtung umfasst: eine erste Mehrpulssuchschaltung (110), die in einer ersten Stufe ein Mehrpulssignal bestimmt, in der das Mehrpulssignal kodiert wird und ein erstes minimiertes Differenzsignal erzeugt wird, mindestens eine Hilfs-Mehrpulssuchschaltung (131), in der Pulspositionen von Pulspositionskandidaten in jeder folgenden der mehreren Stufen ausgewählt werden und die ausgewählten Pulspositionen kodiert werden, um ein Differenzsignal einer vorigen Stufe zu modellieren und ein weiteres minimiertes Differenzsignal zu erzeugen, dadurch gekennzeichnet, dass die Audiokodiervorrichtung zwischen den Stufen eine Mehrpulseinstellschaltung (130) umfasst, die die Pulspositionskandidaten an Positionen setzt, denen keine Pulse zugewiesen worden sind, mit Priorität über Positionen, an denen Pulse bereits in vorhergehenden Stufen kodiert worden sind.
Audiokodiervorrichtung nach Anspruch 1, wobei die Mehrpulseinstellschaltung (130) jeden Subrahmen in mehrere Unterbereiche teilt, eine begrenzte Zahl der Unterbereiche entsprechend der Zahl von bereits darin kodierten Pulsen auswählt, wobei die Unterbereiche mit der kleineren Zahl von bereits kodierten Pulsen zuerst ausgewählt werden, und die Indizes der ausgewählten Unterbereiche an die nächste Stufe ausgibt. Audiokodiervorrichtung nach Anspruch 2, wobei jede der mehreren Stufen Pulse des Mehrpulssignals nur in den Unterbereichen kodiert, die den Indizes von der Mehrpulseinstellschaltung (130) entsprechen. Audiokodierverfahren zum Kodieren eines Mehrpulssignals, das ein Anregungssignal eines reproduzierten Audiosignals wiedergibt, in mehreren Stufen, indem mehrere Pulse Pulspositionen zugewiesen werden, die eine Differenz zwischen dem reproduzierten Audiosignal und einem eingegebenen Audiosignal minimieren, wobei das reproduzierte Audiosignal erhalten wird, indem ein linearer Vorhersage-Synthesefilter mittels des Anregungssignals getrieben wird,

wobei das Audiokodierverfahren umfasst: einen ersten Mehrpulssuchschritt, der in einer ersten Stufe ein Mehrpulssignal bestimmt, in der das Mehrpulssignal kodiert wird und ein erstes minimiertes Differenzsignal erzeugt wird, mindestens einen Hilfs-Mehrpulssuchschritt, in dem Pulspositionen von Pulspositionskandidaten in jeder folgenden der mehreren Stufen ausgewählt werden und die ausgewählten Pulspositionen kodiert werden, um ein Differenzsignal einer vorigen Stufe zu modellieren und ein weiteres minimiertes Differenzsignal zu erzeugen, dadurch gekennzeichnet, dass das Audiokodierverfahren zwischen den Stufen einen Mehrpulseinstellschritt umfasst, der die Pulspositionskandidaten an Positionen setzt, denen keine Pulse zugewiesen worden sind, mit Priorität über Positionen, an denen Pulse bereits in vorhergehenden Stufen kodiert worden sind.
Anspruch[en]
An audio encoding apparatus for encoding in multiple stages a multi-pulse signal representing an excitation signal of a reproduced audio signal by assigning plural pulses to pulse positions which minimize a difference between said reproduced audio signal and an input audio signal, said reproduced audio signal being obtained by driving a linear predictive synthesis filter by means of said excitation signal,

said audio encoding apparatus comprising: a first multi-pulse searching circuit (110) determining in a first stage a multi-pulse signal, in which the multi-pulse signal is encoded, and a first minimized difference signal is generated, at least one auxiliary multi-pulse searching circuit (131) in which, in each subsequent of said multiple stages, pulse positions from pulse position candidates are selected and the selected pulse positions are encoded for modelling a difference signal of a previous stage and generating a further minimized difference signal, characterized in that said audio encoding apparatus comprises, between said stages, a multi-pulse setting circuit (130) which sets the pulse position candidates at positions to which no pulses have been assigned with priority over positions at which pulses have been already encoded in preceding stages.
The audio encoding apparatus as set forth in claim 1, wherein said multi-pulse setting circuit (130) divides each sub-frame into plural sub-areas, selects a limited number of said sub-areas according to the number of pulses already encoded therein, wherein the sub-areas having the smaller number of pulses already encoded are selected first, and outputs the indices of the selected sub-areas to next stage. The audio encoding apparatus as set forth in claim 2, wherein each of said multiple stages encodes pulses of said multi-pulse signal only in said sub-areas corresponding to said indices from said multi-pulse setting circuit (130). An audio encoding method for encoding in multiple stages a multi-pulse signal representing an excitation signal of a reproduced audio signal by assigning plural pulses to pulse positions which minimize a difference between said reproduced audio signal and an input audio signal, said reproduced audio signal being obtained by driving a linear predictive synthesis filter by means of said excitation signal,

said audio encoding method comprising

a first multi-pulse searching step determining in a first stage a multi-pulse signal, in which the multi-pulse signal is encoded, and a first minimized difference signal is generated,

at least one auxiliary multi-pulse searching step in which, in each subsequent of said multiple stages, pulse positions from pulse position candidates are selected and the selected pulse positions are encoded for modelling a difference signal of a preceding stage and generating a further minimized difference signal, characterized in that

said audio encoding method comprises, between said stages, a multi-pulse setting step which sets the pulse position candidates at positions to which no pulses have been assigned with priority over positions at which pulses have been already encoded in preceding stages.
Anspruch[fr]
Dispositif de codage audio pour coder dans des étages multiples un signal multi-impulsionnel représentant un signal d'excitation d'un signal audio reproduit en affectant plusieurs impulsions à des positions d'impulsion qui minimisent une différence entre ledit signal audio reproduit et un signal audio d'entrée, ledit signal audio reproduit étant obtenu en actionnant un filtre de synthèse par prédiction linéaire au moyen dudit signal d'excitation,

ledit dispositif de codage audio comprenant : un premier circuit de recherche multi-impulsionnel (110) déterminant dans un premier étage un signal multi-impulsionnel, dans lequel le signal multi-impulsionnel est codé, et un premier signal de différence minimisée est généré, au moins un circuit de recherche multi-impulsionnel auxiliaire (131) dans lequel, dans chaque étage ultérieur desdits étages multiples, des positions d'impulsion des candidats de position d'impulsion sont sélectionnées et les positions d'impulsion sélectionnées sont codées pour modéliser un signal de différence d'un étage antérieur et générer un autre signal de différence minimisée, caractérisé en ce que ledit dispositif de codage audio comprend, entre lesdits étages, un circuit de réglage multi-impulsionnel (130) qui règle les candidats de position d'impulsion au niveau de positions auxquelles aucune impulsion n'a été affectée avec une priorité sur les positions au niveau desquelles des impulsions ont déjà été codées dans des étages précédents.
Dispositif de codage audio selon la revendication 1, dans lequel ledit circuit de réglage multi-impulsionnel (130) divise chaque secteur de trame en plusieurs sous-zones, sélectionne un nombre limité desdites sous-zones selon le nombre d'impulsions déjà codées à l'intérieur, dans lequel les sous-zones ayant le plus petit nombre d'impulsions déjà codées sont sélectionnées en premier, et affiche les indices des sous-zones sélectionnées à l'étage suivant. Dispositif de codage audio selon la revendication 2, dans lequel chacun desdits étages multiples code les impulsions dudit signal multi-impulsionnel seulement dans lesdites sous-zones correspondant auxdits indices dudit circuit de réglage multi-impulsionnel. Procédé de codage audio pour coder dans des étages multiples un signal multi-impulsionnel représentant un signal d'excitation d'un signal audio reproduit en affectant plusieurs impulsions à des positions d'impulsion qui minimisent une différence entre ledit signal audio reproduit et un signal audio d'entrée, ledit signal audio reproduit étant obtenu en actionnant un filtre de synthèse par prédiction linéaire au moyen dudit signal d'excitation,

ledit procédé de codage audio comprenant

une première étape de recherche multi-impulsionnelle déterminant dans un premier étage un signal multi-impulsionnel, dans lequel le signal multi-impulsionnel est codé, et un premier signal de différence minimisée est généré,

au moins une étape de recherche multi-impulsionnelle auxiliaire dans laquelle, dans chaque étage ultérieur desdits étages multiples, des positions d'impulsions des candidats de position d'impulsion sont sélectionnées et les positions d'impulsion sélectionnées sont codées pour modéliser un signal de différence d'un étage antérieur et générer un autre signal de différence minimisée, caractérisé en ce que

ledit procédé de codage audio comprend, entre lesdits étages, une étape de réglage multi-impulsionnel qui règle les candidats de position d'impulsion au niveau de positions auxquelles aucune impulsion n'a été affectée avec une priorité sur les positions au niveau desquelles des impulsions ont déjà été codées dans des étages précédents.






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