EP1016319B1 - Verfahren und vorrichtung zum codieren eines zeitdiskreten stereosignals - Google Patents
Verfahren und vorrichtung zum codieren eines zeitdiskreten stereosignals Download PDFInfo
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- EP1016319B1 EP1016319B1 EP98932156A EP98932156A EP1016319B1 EP 1016319 B1 EP1016319 B1 EP 1016319B1 EP 98932156 A EP98932156 A EP 98932156A EP 98932156 A EP98932156 A EP 98932156A EP 1016319 B1 EP1016319 B1 EP 1016319B1
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—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
- G10L19/16—Vocoder architecture
- G10L19/18—Vocoders using multiple modes
- G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S1/00—Two-channel systems
- H04S1/007—Two-channel systems in which the audio signals are in digital form
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
- G10L19/0212—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
Definitions
- the present invention relates to scalable audio encoders and particularly on methods and devices for coding a discrete-time stereo signal.
- Scalable audio encoders are encoders that have a modular structure are. So there is an endeavor to already exist Speech encoder to use the signals, e.g. B. with 8 kHz are sampled, process and data rates of, for example Output 4.8 to 8 kilobits per second.
- This known encoders such as. B. those known to experts Encoders G. 729, G.723, FS1016, CELP or parametric Models of the MPEG-4 audio VM are mainly used for Encode speech signals and are generally for Coding of higher quality music signals not suitable, as they are usually used for 8 kHz sampled signals are designed, which is why they are only one audio bandwidth of a maximum of 4 kHz. However, they show in generally fast operation and low Computing effort.
- a speech encoder with a Audio encoder which signals with a higher sampling rate, such as B. 48 kHz, can encode.
- a speech encoder mentioned above it is also possible to use the speech encoder mentioned above to replace another encoder, for example by a music / audio encoder according to the standards MPEG1, MPEG2 or MPEG4.
- Such a chain connection of a speech encoder a higher quality audio encoder is commonly used the method of differential coding in the time domain.
- On Input signal for example, a sampling rate of 48 kHz, is based on a downsampling filter downsampled the sampling frequency suitable for the speech encoder. Now the down sampled signal coded.
- the encoded signal can be sent directly to a bit stream formatter be fed to be transmitted. However, it only contains signals with a bandwidth of e.g. B. maximum 4 kHz.
- the encoded signal is also restored decoded and up-sampled using an upsampling filter. However, the signal now received has due to of the downsampling filter only with useful information a bandwidth of 4 kHz, for example.
- the spectral content of the sampled up coded / decoded signal in the lower band up to 4 kHz not exactly the first 4 kHz band of the one sampled at 48 kHz Corresponds to input signal since encoder in general Introduce coding errors.
- a scalable encoder has a well known speech coder as well an audio encoder that receives signals with higher sampling rates can process.
- This difference can then be determined using a known Audio encoders can be quantized and encoded as it is for Is known to experts.
- the difference signal that is in the audio encoder the signals can code with higher sampling rates, is fed in lower frequency range apart from coding errors of the Speech encoder is much smaller than the original.
- the difference signal corresponds to essentially the true input signal, which with z. B. 48 kHz was scanned.
- the level of the speech coder is usually an encoder with a low sampling frequency used because generally a very low bit rate of coded signal is sought.
- the maximum possible audio bandwidth is 4 kHz and is limited in practice to about 3.5 kHz. Should now in the further stage, d. H. in the stage with the audio encoder, bandwidth improvement must be achieved another stage with a higher sampling frequency.
- For Adjustment of the sampling frequencies are decimation and Interpolation filter used for downsampling or upsampling.
- the object of the present invention is a Method and device for coding a time-discrete To create stereo signal which use of joint stereo techniques.
- This task is accomplished by a method of encoding a discrete-time stereo signal according to claim 1 and by a device for coding a discrete-time stereo signal solved according to claim 14.
- the present invention is based on the finding that that a combination of joint stereo techniques with the Principle of scalability can be achieved when out the left and right channels of a stereo signal first a mono signal is formed, which is preferably by summation can happen.
- the mono signal is generated by means of a first Encoder coded, whereupon the resulting Signal is fed to a bitstream multiplexer.
- the encoded Mono signal is also decoded again to an encoded / decoded Obtain mono signal that differs from the original Mono signal differs in that it has coding errors having introduced by the first encoder have been.
- the coded / decoded can also Mono signal itself or the difference of the original Mono signal from the coded / decoded mono signal as Stereo information used to be used along with the Difference from left and right channel, which also as S signal is called, directly a middle / side coding to surrender.
- the stereo information can now by means of a second encoder, which is identical to the first encoder or can also be constructed differently from the first encoder, coded and also fed to a bitstream multiplexer, which is a bit stream of the encoded mono signal and the encoded stereo information as well as from later Decoding necessary page information generated.
- Forming and encoding the mono signal can take place in the time domain if as the first encoder or Core encoder z.
- a frequency domain encoder can be used which can be done using the psychoacoustic model can encode as distortion-free as possible.
- an encoder is used, which is a lower one Sampling rate has as the time-discrete stereo signal to be encoded, so it has to be from the summation of left and right Channel formed mono signal first to the lower sampling frequency be implemented, which is also known as downsampling becomes.
- the converted to the lower sampling frequency Mono signal is now encoded and decoded again, whereby the encoded / decoded mono signal is also the lower Sampling frequency.
- the coded / decoded mono signal back to the sampling frequency of the time-discrete stereo signal which is also implemented as Upsampling is called.
- Fig. 1 shows a basic block diagram of a scalable Stereo encoder 100 according to a first embodiment of the present invention.
- the scalable stereo encoder receives a discrete-time stereo signal that a first or left channel L and a second or right channel R includes.
- the stereo signal is preferred by summation by sample by means of a summator 102 a sum signal is formed, which is then by means of multiplier 104 multiplied by a factor of 0.5 is to a mono signal in this embodiment generate that to the middle signal known from the M / S coding is identical.
- the mono signal at the output of the multiplier 104 is fed into a downsampling filter 106, around the sampling rate to a preferably implement lower sampling rate, which encodes the Mono signal by means of a time domain encoder, which Part of the core codec 108 is to enable.
- the encoded Mono signal is sent together with corresponding page information written in a bitstream multiplexer 110 which generates a bit stream at its output 112, which encodes a Representation of the discrete-time stereo signal is.
- the encoded mono signal is within the core codec 108 decoded again by means of an upsampling filter 114 to be converted back to the first sampling rate so the encoded / decoded mono signal with the left and the right channel for later formation of stereo information can be related.
- the discrete-time stereo signal could, for example, by means of a first sampling rate, e.g. B. 48 kHz has been sampled his.
- the downsampling filter 106 could use this signal the first sampling rate to a second sampling rate of e.g. B. 8 implement kHz. Preferably form the first and the second Sampling rate an integer ratio.
- the downsampling filter 106 can be implemented as a decimation filter, for example his.
- the core codec 108 could, for example a speech coder such as e.g. B. G.729, G.723, FS1016, MPEG-4 CELP, MPEG-4 PAR, or a similar encoder.
- Such encoders operate at data rates of 4.8 kilobits per Second (FS1016) up to data rates of 8 kilobits per second (G.729).
- FS1016 4.8 kilobits per Second
- G.729 8 kilobits per second
- the coded mono signal has a maximum of one Bandwidth of 4 kHz, since the downsampling filter 106 does Mono signal z. B. by decimation to a sampling frequency of 8 kHz. Within the range of 0 - 4 kHz are now the encoded / decoded mono signal and the original one Mono signal at the input of the downsampling filter 106 apart from those introduced by the core codec 108 Coding errors equal.
- the coding error introduced by the core codec 108 is not are always small mistakes, but that they are easily can come in orders of magnitude of the useful signal, if for example a strongly transient signal in the first encoder is encoded. For this reason, as will be discussed later is checked whether differential coding at all makes sense.
- the output signal of the upsampling filter 114 now becomes the same like the left and right channels using MDCT filter banks 116 implemented in the frequency domain.
- the output signals of MDCT filter banks 116 as shown in FIG. 1 is shown, a first frequency-selective switching device (FSS) 118a or a second frequency selective Switching device 118b directly or via a first one Totalizer 120a or a second totalizer 120b indirectly fed.
- FSS frequency-selective switching device
- the output signal of the MDCT filter bank for the left channel of the first frequency-selective switching device (FSS) 118a which is also the sum of the transformed left channel and the one with negative Signed transformed coded / decoded Mono signal received.
- the second frequency-selective switching device 118b receives the next to the transformed R channel Sum of the transformed R channel and the one with negative Signed coded / decoded mono signal.
- the output signal of the first frequency-selective switching device 118a is both a third summer 122a and also a fourth summer 122b with a positive sign fed while the output signal of the second frequency selective Switching device 118b the third summer 122a with a positive sign and the fourth summer 122b with negative sign is supplied.
- the third Summer 122a is now either the sum of the transformed left and right channels or the difference from the Sum of the uncoded left and right channels and the encoded / decoded sum of the left and right channels in front.
- This signal which is now in contrast to the encoded Mono signal of the core codec has 108 stereo information, is, for example, by means of an M encoder 124 Consideration of the psychoacoustic model coded and fed to the bitstream multiplexer 110.
- this signal also in technology as a side signal is referred to, which is fed into an S-encoder 126 with the S encoder 126 as well as the M encoder 124 code taking into account the psychoacoustic model can.
- the output signal of the S encoder 126 also becomes fed and included in the bitstream multiplexer also stereo information regarding the time discrete Stereo signal at the input of the scalable stereo encoder 100 according to the first embodiment of the present Invention. It is obvious to experts that a complete bitstream page information needed.
- Side information is especially information of the frequency-selective switching devices 118a and 118b regarding the fact in which frequency band difference signals or transformed L or R signals to the third summer 122a or to the fourth summer 122b have been issued.
- the output signal of the core codec 108 points, as it already does was mentioned, e.g. B. a sampling frequency of 8 kHz.
- This signal i.e. H. the mono signal, with a lower sampling rate than the original discrete-time stereo signal but now related to the left or right channel brought to form stereo information.
- the signal must therefore be included lower sampling rate in a signal with the same sampling rate how the sampling rate of the discrete-time stereo signal is implemented become.
- the number of zero values is calculated from the ratio of the first and the second Sampling frequency.
- the ratio of the first (high) to the second (low) sampling frequency is called the upsampling factor designated.
- the Inserting zeros with very little computation is possible to generate an aliasing disorder that is such affects that the low frequency or zero spectrum of the encoded / decoded mono signal at the output of the core codec 108 is repeated, in total as many times as many Zeros have been inserted.
- the aliasing signal is now in the frequency range using the MDCT filter bank 116 transformed.
- the coded / decoded converted up to the first sampling frequency Mono signal is only in the lower frequency band a correct representation of the original mono signal on Output of the multiplier 104, which is why at the output of the MDCT filter bank 116 only a maximum of one / upsampling factor of the entire spectral lines is used.
- the insert the zeros in the encoded / decoded mono signal on Output of core codec 108 causes the spectral representation of the encoded / decoded mono signal now the same time and frequency resolution as the transformed one has left and right channels.
- any encoder can be used may happen that the encoder by certain the M-encoder 124 or by the S-encoder 126 difficult coding signal components produced.
- the core codec 108 should preferably phase information of the coded by him Preserve signals, which in the professional world as “waveform coding” or “waveform encoding”.
- the decision which is the frequency selective switching module 118a or 118b is carried out, preferably frequency-dependent.
- “Differential coding” means that only the difference the transformed left or right channel and the transformed encoded / decoded mono signal becomes. If this differential coding is not cheap is because the energy content of the difference signal is greater than the energy content of the transformed left or right Signal is apart from a difference coding and switched to simulcast mode.
- Forming stereo information based on the encoded / decoded Mono signal and the first and second Channel therefore includes a determination of where it's cheaper the transformed left or right channel or one Difference between the same and the encoded / decoded mono signal to process.
- a frequency-selective comparison of respective energies carried out If the energy in a certain frequency band the difference signal is the energy of the other Signal multiplied by a predetermined factor k exceeds, it is determined that the output of the frequency-selective switching device 118a the original transformed left signal is. Otherwise it is determined that the difference spectral values are output.
- the Factor k can range, for example, from about 0.1 to 10.
- simulcast coding is already used used when the difference signal is lower Has energy than the other signal.
- differential coding is still used, even if the energy content of the difference signal already larger than that of the original left or right channel is.
- stereo information can also form stereo information be carried out such that, for. B. a ratio or another link of the encoded / decoded mono signal and the transformed left and right channels is implemented.
- FIG 2A shows a scalable stereo encoder 200 according to FIG a second embodiment of the present invention.
- the same elements have the same reference numerals and if they behave the same way, not again described.
- the scalable stereo encoder 200 differs different from the scalable stereo encoder 100 according to the first embodiment of the present invention in essential in the fact that either a middle / side coding or L / R coding can be performed.
- the scalable stereo encoder 200 includes further summing devices 202a, 202b in order to derive from the transformed left and right channel a center signal M or to generate a side signal S. That transformed encoded / decoded mono signal is referred to here as M '.
- the signal M and the signal M ' is also additional frequency-selective switching device 204 fed in, which generates a signal M ′′, the frequency-selective Switching device 204 also a summer 206 is connected upstream, as is the case with all other frequency-selective Switching devices is the case.
- the scalable Stereo encoder 200 also includes a block joint stereo decision 208, which 4 input signals L ', M ", S and R 'receives.
- the block joint stereo decision 208 decides in a known way, whether from a stereo encoder 210 an L / R, an M / S or an intensity coding is to be carried out.
- the index T is intended to indicate that this is a middle signal in the time domain.
- the core encoder 108 now operates as was shown in connection with FIG. 1.
- an MDCT is also carried out on the L and R signals.
- the frequency-selective switching device now serves as it has already been mentioned for calculating M ''. Damn either equal to M - M 'or M itself, as already shown has been.
- the frequency selective switching device 118 computes the signal L ', which is either equal to 0.5 (L - M') or equal to 0.5 ⁇ L.
- the switching devices 118a, 118b and 204 operate frequency selective.
- a decision is now made in the usual way whether a coding of the signals L 'and R' or M "or S has to take place. This function is known in the art and is therefore not explained in more detail.
- FIG. 2B shows a scalable stereo encoder which differs in some points from the scalable stereo encoder 200 according to the second exemplary embodiment of the invention.
- the same comprises the two multipliers 214a and 214b, which are arranged after the frequency-selective switching device 204 and after the frequency-selective switching device 118b.
- 2B also includes a somewhat more detailed illustration of the frequency selective switching devices.
- the switch state of the frequency selective switch 118a which is referred to as S 1LR
- S ' 1LR will always be complementary to the switch state of the frequency selective switch 118b, which is referred to as S ' 1LR .
- S 2 assumes a different state, ie state b, as shown in the drawing, it is sufficient to transmit the state S 1M of the frequency-selective switching device 204, which indicates whether differential or simulcast coding of the signal M is carried out. If the switch S 2 is in a position c, side information is transmitted that there is an intensity stereo coding, in which case the position of the switch S 1M is also transmitted, while here the positions of S 1LR and S ' 1LR are irrelevant.
- FIG. 3 includes another embodiment 300 of a scalable stereo encoder according to the present invention.
- the embodiment shown in Fig. 3 differs differs from the embodiment shown in FIG. 2 essentially in that the mono signal in two Levels is encoded.
- the first stage is through the core codec 108 formed during the second stage by one Encoder / decoder 302 is formed, which in the preferred embodiment works in the frequency domain and implemented as a psychoacoustic frequency domain encoder can be. It receives the input signal M " Output signal of the frequency-selective switching device 204, it is also checked here whether differential coding or simulcast coding makes sense or not.
- the output signal of encoder / decoder 302 becomes a summer 304 supplied, the output signal M '' 'the difference of the Signal M and the output signal of the encoder / decoder 302 corresponds.
- This signal M '' ' is just like that Signals L ', S and R' of a joint stereo decision (not shown) and then a stereo encoder (also not shown) supplied.
- the core codec 108 also includes that encoder / decoder 302 an output to the bit stream multiplexer, to transmit encoded data to the same.
- the Outputs of the frequency-selective switching devices to the Bitstream multiplexers are intended to illustrate that page information of frequency-selective switching devices regarding the use of differential and simulcast coding in a frequency band also the bit stream multiplexer need to be fed to a trouble-free To enable decoding.
- the bit stream includes in 3 additionally shows scalable stereo encoder 300 to the first layer or the first layer that is created by the encoded mono signal of the core codec 108 is formed, a second layer by the coded signal M '' at the bitstream multiplexer output of encoder / decoder 302 is formed, the encoder 300 shown in FIG. 3 an encoding of the mono signal at full sampling rate can enable.
- a scalable audio encoder 400 represents a mono signal formation only in the frequency domain performs.
- the signals L and R by means of MDCT filter banks 116 transformed into the frequency domain, after which an M / S matrix by means of summers 202a and 202b and the subsequent multiplier by a factor of 0.5 is carried out.
- the multiplier is thus at the output on the one hand a center signal M and on the other hand a side signal S on.
- the middle signal used as a mono signal can be is by means of a first encoder / decoder 402 encoded and decoded again, the encoded mono signal M is written in the bitstream, as it already is has been mentioned several times.
- a summing device 404 Downstream of encoder / decoder 402 is a summing device 404 which the Difference between the encoded / decoded mono signal and forms the original mono signal M, this difference is designated as M '.
- the signals L ', M', S and R ' can again use a joint stereo decision facility are supplied, which, however, not shown in Fig. 4 is.
- the encoder 400 presented in FIG. 4 thus operates completely in the frequency domain, the encoder / decoder 402 preferably as a frequency domain encoder full sampling rate is executed.
- the stereo encoder (not shown) after the IS decision level (also in FIG. 4 (not shown) is preferably also used as a frequency domain encoder executed at full sampling rate.
- the in Fig. 4 scalable stereo encoder thus represents a Generalization of the term "scalability" because the Bitstream here no layers or "layers" with different Audio bandwidths but (like the other exemplary embodiments) comprises a monolayer and a stereo layer, which encodes separately from one another by an encoder can be.
- An older monodecoder that doesn't is equipped for stereo operation for example the bit stream of the encoders according to the invention decode to generate at least one mono audio signal.
- the scalable stereo encoders according to the invention are thus backwards compatible with existing monodecoders.
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Description
- Fig. 1
- einen skalierbaren Stereocodierer mit Monosignalbildung und -codierung im Zeitbereich und Mitte/Seite-Codierung im Frequenzbereich gemäß einem ersten Ausführungsbeispiel der vorliegenden Erfindung;
- Fig. 2A
- einen skalierbaren Stereocodierer mit Monosignal-bildung und -codierung im Zeitbereich und einer L/R- oder M/S-Codierung im Frequenzbereich gemäß einem zweiten Ausführungsbeispiel;
- Fig. 2B
- eine detailliertere Darstellung des skalierbaren Stereocodierers von Fig. 2A;
- Fig. 3
- eine erweiterte Darstellung des skalierbaren Stereocodierers, der in Fig. 2A gezeigt ist, gemäß einem dritten Ausführungsbeispiel der vorliegenden Erfindung; und
- Fig. 4
- einen skalierbaren Stereocodierer mit Monosignalbildung im Zeitbereich und wahlweiser L/R- oder M/S-Codierung im Frequenzbereich.
Claims (14)
- Verfahren zum Codieren eines zeitdiskreten Stereosignals, wobei das Stereosignal einen ersten und einen zweiten Kanal (L, R) aufweist, mit folgenden Schritten:(a) Bilden eines Monosignals (M) aus dem Stereosignal;(b) Codieren des Monosignals und Übertragen des codierten Monosignals in einen Bitstrom;(c) Decodieren des codierten Monosignals;(d) Bilden von Stereoinformationen aufgrund des codierten/decodierten Monosignals (M') und des ersten und zweiten Kanals (L, R); und(e) Codieren der Stereoinformationen und Übertragen derselben in den Bitstrom.
- Verfahren nach Anspruch 1, bei dem das zeitdiskrete Stereosignal eine erste Abtastrate aufweist, wobei der Schritt (a) folgende Teilschritte aufweist:(a21) abtastwertweises Summieren des linken und des rechten Kanals (L, R) um ein Summensignal zu erhalten; und(a22) Umsetzen des Summensignals auf eine zweite Abtastrate, die kleiner als die erste Abtastrate ist, um das Monosignal zu erhalten; und(c21) Decodieren des codierten Monosignals, das die zweite Abtastrate aufweist; und(c22) Umsetzen des codierten/decodierten Monosignals auf die erste Abtastrate.
- Verfahren nach einem der vorhergehenden Ansprüche, das ferner folgenden Schritt aufweist:
Transformieren des linken und des rechten Kanals und des codierten/decodierten Monosignals in den Frequenzbereich, wobei die transformierten Signale alle eine im wesentlichen gleiche Zeit- und Frequenzauflösung aufweisen. - Verfahren nach Anspruch 3, bei dem der Schritt (d) folgende Teilschritte aufweist:(d41) frequenzselektives Vergleichen des transformierten linken Kanals mit der Differenz aus dem transformierten linken Kanal und dem transformierten codierten/decodierten Monosignal und Auswählen des Signals, das die kleinere gehörmäßige Entropie oder die kleinere Energie hat oder mit einer geringeren Bitzahl codierbar ist;(d42) frequenzselektives Vergleichen des transformierten rechten Kanals mit der Differenz aus dem transformierten rechten Kanal und dem transformierten codierten/decodierten Monosignal und Auswählen des Signals, das die kleinere gehörmäßige Entropie oder die kleinere Energie hat oder mit einer geringeren Bitzahl codierbar ist;(d43) Summieren der in den Schritten (d41) und (d42) ausgewählten Signale, um als erste Stereoinformationen ein Mitte-Signal (M) zu erhalten; und(d44) Subtrahieren des in dem Schritt (d42) ausgewählten Signals von dem in dem Schritt (d41) ausgewählten Signal, um als zweite Stereoinformationen ein Seite-Signal (S) zu erhalten.
- Verfahren nach einem der Ansprüche 1 - 3, bei dem der Schritt (d) folgende Teilschritte aufweist:(d51) Summieren des transformierten linken Kanals (L) und des transformierten rechten Kanals (R), um ein Mitte-Signal (M) zu erhalten; und(d52) Subtrahieren des transformierten rechten Kanals (R) von dem transformierten linken Kanal (L), um ein Seite-Signal (S) zu erhalten.
- Verfahren nach Anspruch 5, bei dem der Schritt (d) ferner folgende Teilschritte aufweist:(d61) frequenzselektives Vergleichen des transformierten codierten/decodierten Monosignals (M') mit der Differenz aus dem Mitte-Signal (M) und dem codierten/decodierten Monosignal (M') und Auswählen des Signals mit der kleineren Energie;(d62) frequenzselektives Vergleichen des linken Kanals mit der Differenz aus dem linken Kanal (L) und dem transformierten codierten/decodierten Monosignal (M'); und(d63) frequenzselektives Vergleichen des rechten Kanals mit der Differenz aus dem rechten Kanal (R) und dem transformierten codierten/decodierten Monosignal (M').
- Verfahren nach Anspruch 6, bei dem der Schritt (d) ferner folgenden Teilschritt aufweist:
(d71) Entscheiden, ob als erste und zweite Stereoinformationen die Ergebnisse der Schritte (d61) und (d52) bzw. die Ergebnisse der Schritte (d62) und (d63) verwendet werden. - Verfahren nach Anspruch 7, bei dem der Schritt (d) vor dem Schritt (d71) ferner folgenden Teilschritt aufweist:
(d81) Halbieren der Ergebnisse der Schritte (d61) und (d52). - Verfahren nach Anspruch 7 oder 8, bei dem der Schritt (d) ferner folgenden Teilschritt aufweist:
(d91) falls in den Schritten (d71) die Ergebnisse der Schritte (d62) und (d63) als erste und zweite Stereoinformationen verwendet werden, Übertragen von Seiteninformationen, die entweder auf das Ergebnis des Schritts (d62) oder des Schritts (d63) hinweisen, sonst, Übertragen von Seiteninformationen, die auf das Ergebnis des Schritts (d61) hinweisen. - Verfahren nach einem der Ansprüche 1 - 5, bei dem der Schritt (d) ferner folgende Teilschritte aufweist:
(d101) frequenzselektives Vergleichen des Mitte-Signals (M) mit der Differenz aus dem Mitte-Signal (M) und dem transformierten codierten/decodierten Monosignal (M') und Auswählen des Signals mit der kleineren Energie als weiteres Monosignal;
bei dem der Schritt (b) ferner folgenden Schritt aufweist:(b101) Codieren des weiteren Monosignals (M") und Übertragen des codierten weiteren Monosignals in den Bitstrom; und(b102) Decodieren des codierten weiteren Monosignals. - Verfahren nach Anspruch 10, bei dem der Schritt (d) folgende Teilschritte aufweist:(d111) Subtrahieren des codierten/decodierten weiteren Monosignals (M") von dem Mitte-Signal (M);(d112) frequenzselektives Vergleichen des transformierten linken Kanals (L) mit der Differenz des linken Kanals und dem Ergebnis des Schritts (d111) und Auswählen des Signals mit der kleineren Energie;(d113) frequenzselektives Vergleichen des transformierten linken Kanals (L) mit der Differenz des rechten Kanals und dem Ergebnis des Schritts (d111) und Auswählen des Signals mit der kleineren Energie; und(d114) Entscheiden, ob als erste und zweite Stereoinformationen die Ergebnisse der Schritte (d111) (M''') und (d52) (S) bzw. die Ergebnisse der Schritte (d112) (L') und (d113) (R') verwendet werden.
- Verfahren nach Anspruch 1, bei dem vor dem Schritt (a) der linke und der rechte Kanal in den Frequenzbereich transformiert werden, wobei der Schritt (a) folgenden Teilschritt aufweist:
(a121) spektralwertweises Summieren des transformierten linken und rechten Kanals, um das Monosignal (M) zu erhalten. - Verfahren nach Anspruch 12, bei dem der Schritt (d) folgende Teilschritte aufweist:(d131) Subtrahieren des codierten/decodierten Monosignals von dem Monosignal (M);(d132) Subtrahieren des transformierten rechten Kanals (R) von dem transformierten linken Kanal (L), um ein transformiertes Seite-Signal (S) zu erhalten;(d133) spektralwertweises Vergleichen des transformierten linken Signals (L) mit der Differenz aus dem transformierten linken Signal (L) und dem Ergebnis des Schrittes (d131) und Auswählen des Signals mit kleinerer Energie;(d134) spektralwertweises Vergleichen des transformierten rechten Signals (R) mit der Differenz aus dem transformierten rechten Signal und dem Ergebnis des Schritts (d131) und Auswählen des Signals mit kleinerer Energie; und(d135) Entscheiden, ob als erste und zweite Stereoinformationen die Ergebnisse der Schritte (d133) (L') und (d134) (R') oder die Ergebnisse der Schritte (d131) (M') und (d132) (S) verwendet werden.
- Vorrichtung (100; 200; 300; 400) zum Codieren eines zeitdiskreten Stereosignals, wobei das Stereosignal einen ersten und einen zweiten Kanal (R, L) aufweist, mit folgenden Merkmalen:(a) einer Einrichtung (102,104; 202a) zum Bilden eines Monosignals aus dem Stereosignal;(b) einer Einrichtung (108; 402) zum Codieren des Monosignals und zum Übertragen des codierten Monosignals in einen Bitstrom;(c) einer Einrichtung (108; 402) zum Decodieren des codierten Monosignals;(d) einer Einrichtung (116, 118a, 118b, 120a, 120b, 122a, 122b; 202a, 202b, 204, 208; 214a, 214b; 302, 304; 402, 404) zum Bilden von Stereoinformationen aufgrund des codierten/decodierten Monosignals und des ersten und des zweiten Kanals; und(e) einer Einrichtung (124, 126; 210) zum Codieren der Stereoinformationen und zum Übertragen derselben in den Bitstrom.
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DE19742655A DE19742655C2 (de) | 1997-09-26 | 1997-09-26 | Verfahren und Vorrichtung zum Codieren eines zeitdiskreten Stereosignals |
DE19742655 | 1997-09-26 | ||
PCT/EP1998/003605 WO1999017587A1 (de) | 1997-09-26 | 1998-06-15 | Verfahren und vorrichtung zum codieren eines zeitdiskreten stereosignals |
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EP1016319A1 EP1016319A1 (de) | 2000-07-05 |
EP1016319B1 true EP1016319B1 (de) | 2001-08-29 |
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EP98932156A Expired - Lifetime EP1016319B1 (de) | 1997-09-26 | 1998-06-15 | Verfahren und vorrichtung zum codieren eines zeitdiskreten stereosignals |
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US (1) | US6629078B1 (de) |
EP (1) | EP1016319B1 (de) |
AT (1) | ATE205041T1 (de) |
DE (2) | DE19742655C2 (de) |
DK (1) | DK1016319T3 (de) |
ES (1) | ES2161059T3 (de) |
WO (1) | WO1999017587A1 (de) |
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CN101091206B (zh) * | 2004-12-28 | 2011-06-01 | 松下电器产业株式会社 | 语音编码装置和语音编码方法 |
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EP1233556A1 (de) * | 2001-02-16 | 2002-08-21 | Sony International (Europe) GmbH | Empfänger für den Empfang von Rundfunksignalen mit Verwendung von zwei Empfängern, für den Empfang eines Rundfunksignals das auf zwei unterschiedlichen Rundfunkfrequenzen oder mit zwei unterschiedlichen Übertragungssystemen übertragen wird |
US7644001B2 (en) * | 2002-11-28 | 2010-01-05 | Koninklijke Philips Electronics N.V. | Differentially coding an audio signal |
CN1765153A (zh) * | 2003-03-24 | 2006-04-26 | 皇家飞利浦电子股份有限公司 | 表示多信道信号的主和副信号的编码 |
JP4555299B2 (ja) * | 2004-09-28 | 2010-09-29 | パナソニック株式会社 | スケーラブル符号化装置およびスケーラブル符号化方法 |
BRPI0516739A (pt) * | 2004-09-30 | 2008-09-23 | Matsushita Electric Ind Co Ltd | dispositivo de codificação escalável, dispositivo de decodificação escalável, e método dos mesmos |
JP5046652B2 (ja) * | 2004-12-27 | 2012-10-10 | パナソニック株式会社 | 音声符号化装置および音声符号化方法 |
WO2006070760A1 (ja) * | 2004-12-28 | 2006-07-06 | Matsushita Electric Industrial Co., Ltd. | スケーラブル符号化装置およびスケーラブル符号化方法 |
EP1876586B1 (de) * | 2005-04-28 | 2010-01-06 | Panasonic Corporation | Audiocodierungseinrichtung und audiocodierungsverfahren |
EP1876585B1 (de) * | 2005-04-28 | 2010-06-16 | Panasonic Corporation | Audiocodierungseinrichtung und audiocodierungsverfahren |
US20090276210A1 (en) * | 2006-03-31 | 2009-11-05 | Panasonic Corporation | Stereo audio encoding apparatus, stereo audio decoding apparatus, and method thereof |
US8150702B2 (en) * | 2006-08-04 | 2012-04-03 | Panasonic Corporation | Stereo audio encoding device, stereo audio decoding device, and method thereof |
US9009032B2 (en) * | 2006-11-09 | 2015-04-14 | Broadcom Corporation | Method and system for performing sample rate conversion |
EP2201566B1 (de) * | 2007-09-19 | 2015-11-11 | Telefonaktiebolaget LM Ericsson (publ) | Gemeisame mehrkanal-audio kodierung/dekodierung |
WO2009057329A1 (ja) * | 2007-11-01 | 2009-05-07 | Panasonic Corporation | 符号化装置、復号装置およびこれらの方法 |
US9330671B2 (en) * | 2008-10-10 | 2016-05-03 | Telefonaktiebolaget L M Ericsson (Publ) | Energy conservative multi-channel audio coding |
BR122019023877B1 (pt) * | 2009-03-17 | 2021-08-17 | Dolby International Ab | Sistema codificador, sistema decodificador, método para codificar um sinal estéreo para um sinal de fluxo de bits e método para decodificar um sinal de fluxo de bits para um sinal estéreo |
JP5333257B2 (ja) * | 2010-01-20 | 2013-11-06 | 富士通株式会社 | 符号化装置、符号化システムおよび符号化方法 |
EP2625688B1 (de) * | 2010-10-06 | 2014-12-03 | Fraunhofer Gesellschaft zur Förderung der angewandten Forschung e.V. | Vorrichtung und verfahren zur verarbeitung eines audiosignals und zur bereitstellung einer höheren zeitlichen auflösung für einen kombinierten einheitlichen sprach- und audio-codec (usac) |
EP2544466A1 (de) * | 2011-07-05 | 2013-01-09 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Verfahren und Vorrichtung zur Zerlegung einer Stereoaufzeichnung mittels Frequenzdomänenverarbeitung unter Verwendung eines spektralen Subtrahieres |
TWI557727B (zh) * | 2013-04-05 | 2016-11-11 | 杜比國際公司 | 音訊處理系統、多媒體處理系統、處理音訊位元流的方法以及電腦程式產品 |
EP4297026A3 (de) * | 2013-09-12 | 2024-03-06 | Dolby International AB | Verfahren zur decodierung und decodierer. |
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DE4217276C1 (de) * | 1992-05-25 | 1993-04-08 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung Ev, 8000 Muenchen, De | |
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DE4345171C2 (de) * | 1993-09-15 | 1996-02-01 | Fraunhofer Ges Forschung | Verfahren zum Bestimmen der zu wählenden Codierungsart für die Codierung von wenigstens zwei Signalen |
KR960012475B1 (ko) * | 1994-01-18 | 1996-09-20 | 대우전자 주식회사 | 디지탈 오디오 부호화장치의 채널별 비트 할당 장치 |
DE4409368A1 (de) * | 1994-03-18 | 1995-09-21 | Fraunhofer Ges Forschung | Verfahren zum Codieren mehrerer Audiosignale |
DE19549621B4 (de) * | 1995-10-06 | 2004-07-01 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Vorrichtung zum Codieren von Audiosignalen |
US5852806A (en) * | 1996-03-19 | 1998-12-22 | Lucent Technologies Inc. | Switched filterbank for use in audio signal coding |
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-
1997
- 1997-09-26 DE DE19742655A patent/DE19742655C2/de not_active Expired - Lifetime
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1998
- 1998-06-15 US US09/445,894 patent/US6629078B1/en not_active Expired - Lifetime
- 1998-06-15 DK DK98932156T patent/DK1016319T3/da active
- 1998-06-15 DE DE59801343T patent/DE59801343D1/de not_active Expired - Lifetime
- 1998-06-15 AT AT98932156T patent/ATE205041T1/de active
- 1998-06-15 WO PCT/EP1998/003605 patent/WO1999017587A1/de active IP Right Grant
- 1998-06-15 ES ES98932156T patent/ES2161059T3/es not_active Expired - Lifetime
- 1998-06-15 EP EP98932156A patent/EP1016319B1/de not_active Expired - Lifetime
Cited By (1)
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CN101091206B (zh) * | 2004-12-28 | 2011-06-01 | 松下电器产业株式会社 | 语音编码装置和语音编码方法 |
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DE59801343D1 (de) | 2001-10-04 |
ES2161059T3 (es) | 2001-11-16 |
ATE205041T1 (de) | 2001-09-15 |
WO1999017587A1 (de) | 1999-04-08 |
US6629078B1 (en) | 2003-09-30 |
DE19742655A1 (de) | 1999-04-22 |
EP1016319A1 (de) | 2000-07-05 |
DE19742655C2 (de) | 1999-08-05 |
DK1016319T3 (da) | 2001-10-08 |
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