BRPI0304231B1 - METHOD FOR CODIFYING A MULTIPLE CHANNEL SIGNAL, METHOD AND ARRANGEMENT FOR DECODING MULTIPLE CHANNEL SIGNAL, PROVISION FOR CODING A MULTIPLE CHANNEL SIGNAL, DATA SIGN INCLUDING MULTIPLE CHANNEL SIGNAL INFORMATION, AND A COMMUNICATION DEVICE SIGNAL - Google Patents

Description

“METHOD FOR CODING A MULTIPLE CHANNEL SIGNAL, METHOD AND PROVISION FOR DECODING MULTIPLE CHANNEL SIGNAL, ARRANGEMENT FOR CODING A MULTIPLE CHANNEL, DATA SIGN INCLUDING MULTIPLE CHANNEL INFORMATION, A MULTIPLE CHANNEL SIGNAL This invention relates to the coding of multi-channel signals including at least a first and a second signal component. More generally, the invention relates to the coding of multiphonic audio signals, such as stereo signals.

Stereo audio signals comprise a left (L) and a right (R) signal component, which may originate from a stereo signal source, for example, from separate microphones. The coding of audio signals is intended to reduce the bit rate of a stereo signal, for example, to enable efficient transmission of sound signals over a communication network, such as the Internet, over a modem and lines. analog telephone lines, mobile communication channels or other wireless networks, etc., and for storing a stereo sound signal on an ehip card or other storage medium with limited storage capacity. U.S. Patent 4,589,127 discloses a transmitter for stereo signals which generate sum and difference signals from stereo L and R signals. The sum and difference signals are subsequently used in a modulation scheme, resulting in a modulated signal comprising both the sum and difference information.

However, the above prior art method does not address the problem of encoding a stereo signal with efficient bit rate utilization, that is, with a low bit rate for a given sound quality.

The above and other problems are solved by a method for encoding a multi-channel signal including at least one first signal component and a second signal component, the method comprising the steps of transforming at least the first and second signal components by by means of a predetermined transformation into a main signal including most of the signal energy and at least one residual signal including less energy than the main signal, the predetermined transformation being parameterized by at least one transformation parameter; and representing the multi-channel signal at least by means of the main signal and the transformation parameter.

Accordingly, by transforming the multi-channel signal into a main signal, which includes most of the energy, and into a residual signal, which only includes low signal energy, the multi-channel signal can be represented by the main signal. , by the transformation parameter, and optionally by a small residual signal, thereby enhancing the coding efficiency for the multi-channel signal. Indeed, the multi-channel signal may be encoded with a bit rate that is only slightly higher than that of a single channel, for example a single channel. The resulting encoded signal may be stored and / or communicated to / from a receiver.

When the method also comprises the step of adaptively determining the transformation parameter based on at least the first and second signal components, the optimal transformation parameter can be continuously tracked thereby ensuring that the transformation remains optimal even if the characteristics of the input signal change, that is, in the example of an audio signal, due to a moving sound source or changes in the acoustic properties of the environment.

When the predetermined transformation is a rotation and the transformation parameter corresponds to a rotation angle, a simple transformation is provided, based on only one parameter, the rotation angle. By adapting the angle such that the signal components, for example the signal components L and R of a stereo signal, are rotated to a main component signal and a residual signal, efficient coding is provided while a high quality signal is maintained.

In a preferred embodiment of the invention, the step of representing the multi-channel signal by at least the main signal and the transformation parameter also comprises the step of representing the multi-channel signal by the main signal, the transformation parameter and the signal. thus further enhancing the quality of the encoded signal as if signal information were discarded. Since the residual signal is small compared to the main signal, bit allocation can be negotiated between those signals. In addition, as bit rate allocation can be varied, a mechanism for elegant degradation is provided, for example by adaptively raising or lowering the permitted bit rate for the residual signal.

Accordingly, in another preferred embodiment, the step of representing the multi-channel signal by the main signal, the transformation parameter and the residual signal also comprises the steps of: encoding the main signal with a first bit rate; and encoding the residual signal with a second bit rate lower than the first bit rate.

In another preferred embodiment the method also comprises the step of estimating the residual signal from the main signal using a prediction filter corresponding to a set of filter parameters; and the step of representing the multi-channel signal by at least the main signal and the transformation parameter comprises the step of representing the multi-channel signal by the main signal, the transformation parameter, and the filter parameter set.

This embodiment of the invention is based on the recognition that for many multi-channel signals, for example in the case of audio signals for music and voice signals, the residual signal may be estimated as a filtered version of the main signal. Consequently, when determining a filter parameter set of an adaptive filter, which models the residual signal, the filter parameters can be encoded together with the main signal and the transformation parameter. Consequently, communication of the residual signal is avoided, without losing the information included in this signal, thereby providing efficient coding that preserves a high level of quality. It is an advantage of the invention that it provides efficient bit rate utilization, that is, an encoding scheme which uses a low bit rate for a given sound quality. The encoding scheme according to the invention may be used to reduce bit rate without significantly reducing sound quality, to maintain bit rate while improving sound quality, or a combination of the above. The invention also relates to a method for decoding multiple channel signal information, the method comprising the steps of: receiving a main signal and a transform parameter, the main signal corresponding to a result of a predetermined transformation of at least one first and second signal components of a multi-channel source signal, the predetermined transformation being parameterized by at least the transformation parameter; and generating first and second decoded signal components by inverse transformation of the received main signal and at least one residual signal. The present invention may be implemented in different ways, including the methods described above and in the following arrangements for encoding and decoding multi-channel signals, respectively, a data signal, and other product media, each producing one or more of the following. benefits and advantages described in connection with the first mentioned method, and each having one or more preferred embodiments corresponding to the embodiments described in connection with the first mentioned method and disclosed in the dependent claims. It is noted that the characteristics of the methods described above and below may be implemented in software and executed in a data processing system or other processing means caused by the execution of computer executable instructions. The instructions may be program code means loaded into a memory such as RAM from a storage medium or from another computer through a computer network. Alternatively, the described features may be implemented by means of circuits in the form of physical conductors, rather than software, or in combination with software. The invention also relates to an arrangement for encoding a multi-channel signal including at least a first signal component and a second signal component, the arrangement comprising: first processing means adapted to transform at least the first and second signal components by means of a predetermined transformation into a main signal including most of the signal energy and at least one residual signal including less energy than the main signal, the predetermined transformation being parameterized by at least one transformation parameter; and second processing means adapted to represent the multi-channel signal by at least the main signal and the transformation parameter. The invention also relates to an arrangement for decoding multiple channel signal information, the arrangement comprising: receiving means for receiving a main signal and a transform parameter, the main signal corresponding to a result of a predetermined transformation of a first and second multi-channel source signals, the predetermined transformation being parameterized by at least the transformation parameter; and processing means for generating first and second multi-channel signals by reverse transformation of the received main signal and a residual signal.

The above provisions may be part of any electronic equipment including computers, such as stationary and portable PCs, stationary and portable radio communication equipment, and other palm-sized and portable devices such as mobile phones, radiolocals, reader units audio players, multimedia player units, communicators, that is, electronic organizers, smart phones, personal digital assistants (PDAs), laptops, or the like. The term processing means comprises general purpose or special programmable microprocessors, Digital Signal Processors (DSP), Application Specific Integrated Circuits (ASICS), Programmable Logic Arrays (PLA), Field Programmable Gate Arrays (FPGA), electronic circuits special purpose, etc., or a combination thereof. The above first and second processing means may be separate processing means or they may be comprised in a processing medium. The term receiving means includes suitable circuits and / or devices for enabling data communication, for example over a wired or wireless data connection. Examples of such receiving means include a network interface, a network card, a radio receiver, a receiver for other suitable electromagnetic signals, such as infrared light, for example through an IrDa port, radio based communication, for example. , Bluetooth transceivers, or the like. Other examples of such receiving means include a cable modem, a telephone modem, an Integrated Services Digital Network (ISDN) adapter, a Digital Subscriber Line (DSL) adapter, a satellite transceiver, an Ethernet adapter, or the like. The term receiving means also includes other input circuits / devices for receiving data signals, for example data signals stored in a computer readable medium. Examples of such receiving means include a floppy disk reader drive, a CD-ROM reader drive, a DVD reader drive, or any other suitable disk reader drive, a memory card adapter, a smart card adapter, etc. The invention also relates to a data signal including multiple channel signal information, the data signal being generated by a method described above and hereinafter. The signal may be incorporated as a data signal in a carrier wave, for example as a data signal transmitted by means of communication, as described above and below. The invention also relates to a computer readable medium comprising a data record indicative of the multi channel signal information generated by a method described above and hereinafter. The term computer readable media includes magnetic tape, optical disc, digital video disc (DVD), compact disc (CD or CD-ROM), mini disk, hard disk, floppy disk, ferroelectric memory, read-only memory. programmable, electrically erasable (EEPROM), flash memory, EPROM, read-only memory (ROM), static random access memory (SRAM), dynamic random access memory (DRAM), synchronous dynamic random access memory (SDRAM) , ferromagnetic memory, optical storage, charge coupled devices, smart cards, PCMCIA card, etc. The invention also relates to a device for communicating a multi-channel signal including at least a first signal component and a second signal component, the device comprising an arrangement for encoding the multi channel signal as described above and hereinafter. .

These and other aspects of the invention will be apparent from, and elucidated with reference to, the embodiments, and with reference to the drawing, in which: Figure 1 shows a schematic view of a system for communicating stereo signals according to an embodiment of the invention; Figure 2 shows a schematic view of an arrangement for encoding a stereo signal according to a first embodiment of the invention; Figure 3 illustrates the determination of signal transformation according to one embodiment of the invention; Fig. 4 shows a schematic view of an arrangement for decoding a stereo signal according to a first embodiment of the invention; Fig. 5 shows a schematic view of an arrangement for encoding a stereo signal according to a second embodiment of the invention; Fig. 6 shows a schematic view of an arrangement for decoding a stereo signal according to a second embodiment of the invention; Figures 7a-c show schematic views of examples of a filter circuit for use in an embodiment of the invention; Figure 8 shows a schematic view of an arrangement for encoding a stereo signal according to a third embodiment of the invention; Figure 9 shows a schematic view of an arrangement for encoding a stereo signal according to a fourth embodiment of the invention; Fig. 10 shows a schematic view of an arrangement for decoding a stereo signal according to a fourth embodiment of the invention; Fig. 11 shows a schematic view of an arrangement for encoding a multi-channel signal according to a fifth embodiment of the invention; Fig. 12 shows a schematic view of an arrangement for encoding a multi-channel signal according to a sixth embodiment of the invention; and Figure 13 shows a schematic view of a subtraction circuit for use with an embodiment of the invention. Fig. 1 shows a schematic view of a system for communicating stereo signals in accordance with an embodiment of the invention. The system comprises a coding device 101 for generating a coded stereo signal and a decoding device 105 for decoding a received coded signal into a stereo signal component L and a stereo signal component R. Coding device 101 e The decoding device 105 each may be any electronic equipment or part of such equipment. Here, the term electronic equipment includes computers, such as stationary and portable PCs, stationary and portable radio communication equipment, and other palm-sized or portable devices, such as mobile phones, radiolocalizers, audio player units, multimedia players, communicators, that is, electronic organizers, smart phones, personal digital assistants (PDAs), laptops, or the like. It is noted that the encoding device 101 and the decoding device 105 may be combined in electronic equipment, where stereo signals are stored in a computer readable medium for later reproduction. The coding device 101 comprises an encoder 102 for encoding a stereo signal according to the invention, the stereo signal including an L signal component and an R signal component. The encoder receives the L and R signal components and generates a T-coded signal Stereo L and R signals may originate from a set of microphones, for example through other electronic equipment such as mixing equipment, etc. Signals may also be received as output from another stereo reader unit, through the air as a radio signal, or by any other suitable means. Preferred embodiments of such an encoder according to the invention will be described below. According to one embodiment, encoder 102 is connected to a transmitter 103 for transmitting T-coded signal through a communication channel 109 to coding device 105. Transmitter 103 may comprise suitable circuits for enabling communication. for example over a wired or wired data connection 109. Examples of such a transmitter include a network, interface, a network card, a radio transmitter, a transmitter for other suitable electromagnetic signals such as a LED, for transmitting infrared light, for example through an IrDa port, radio-based communication, for example via a Bluetooth transceiver, or the like. Other examples of suitable transmitters include a cable modem, a telephone modem, an Integrated Services Digital Network (ISDN) adapter, an Integrated Subscriber Line (DSL) adapter, a satellite transceiver, an Ethernet adapter, or similar. Correspondingly, the communication channel 109 may be any suitable wired or wireless data connection, for example from a packet-based communication network, such as the Internet or another TCP / IP network, a communication communication link. short range, such as an infrared connection, a Bluetooth connection, or another radio-based connection. Other examples of communication channel include computer networks and wireless telecommunications networks, such as a Cellular Digital Packet Data (CDPD) network, a Global System for Mobile Communications (GSM), a Code Division Multiple Access (CDMA) network, a Time Division Multiple Access NetWork (TDMA) network, a service network General Packet Radio service (GPRS), a third generation network, such as a UMTS network, or the like. Alternatively or additionally, the encoding device may comprise one or more other interfaces 104 for communicating the T encoded stereo signal to the encoding device 105. Examples of such interfaces include a disk reader unit for storing data on computer readable medium 110, for example a floppy disk drive, a read / write CD-ROM drive, a DVD drive. Other examples include a memory card space, a magnetic card read / write device, an interface for accessing a smart card, etc. Correspondingly, the encoding device 105 comprises a corresponding receiver 108 for receiving the signal transmitted by the transmitter and / or another interface 106 for receiving the encoded stereo signal communicated via interface 104 and computer readable medium 110. The decoding device also comprises a decoder 107 which receives the received signal T and decodes it into the corresponding stereo components L 'and R. Preferred embodiments of such a decoder according to the invention will be described below. The decoded L 'and R' signals may subsequently be fed into a stereo playback unit for playback through a set of speakers, headphones, or the like. Fig. 2 shows a schematic view of an arrangement 102 for encoding a stereo signal according to a first embodiment of the invention. The arrangement comprises circuits 201 for performing a rotation of the stereo signal in LR space by an angle a, resulting in yer rotated signal components according to the transformation y = L cos α + R sin α = wL L + wR R r = -Lsena + Rcosa = -WrL + wlR, (1) where wL = cos α and wR = sen α will be referred to as weighting factors.

According to the invention, the angle α is determined so that it corresponds to a direction of high signal variance, that is, the principal component can be estimated by means of a principal component analysis, so that the rotated y component corresponds to the principal component signal, which includes most of the signal energy, and r is a residual signal. Correspondingly, the arrangement of figure 2 comprises circuits 200 which determine the angle α or, alternatively, the weight factors wL and wR.

Referring to Figure 3, according to a preferred embodiment, the above weight factors wL and wR are determined according to the following algorithm: Initially, the stereo arrival signals L and R are rectified and low pass filtered. , resulting in wrap signals p (k) and q (k), respectively, where p (k) and q (k) are adequately sampled and the sample index is denoted by k. Thus, the vector x (k) = (p (k), q (k)) denotes the arrival signal vector. Alternatively, the L and R signals may be used directly, ie without filtering, or other versions of L and R may be used, for example high pass filtered L and R signals. In Figure 3, numerous signal points are illustrated as circles. As an example, the signal point x (k) and its corresponding components p (k) and q (k) are indicated. According to the invention, the signals are rotated towards the main component of the signal vectors. In the example in Figure 3, this corresponds to the y direction, where α is an angle between the y direction and the p direction. The weight vector w = (wL, wR) indicates the direction of the main component, and the rotated components of x (k) are denoted by y (k) and r (k), respectively. The major component can be determined by any suitable method known in the art. In a particularly advantageous embodiment, an iterative method using the Oja's rule is used (see, for example, S. Haykin: "Neural Networks", Prentice Hall, N.J., 1999). According to this embodiment, the weight vector w is iteratively estimated according to the following equation: w (k) + w (kl) + μ [x (k-1) - w (k-1) y ( k-1)], (2) where w (k) = (wL (k), wR (k)) corresponds to the estimate at time k. The above iteration may, for example, be initiated with a group of small random weights w (0), or in any other suitable manner. The above estimated weight vector can be used to calculate the rotated signal according to y (k) = w (k) x (k). Alternatively, the iteration of equation (2) can be performed on a block basis, for example for a block of N samples, where N depends on the particular implementation, eg N = 512, 1024, 2048, etc. In this embodiment, the estimated weight vector w (N) for a block can be used to transform all samples of that block according to y (k) = wT (N) x (k). The factor μ in equation (2) corresponds to a time scale of the tracking algorithm. If μ = 0, the weighting factors and thus angle a remain constant while they rapidly change to large μ. As an example, for a block size of 2048 samples, μ can be selected in the order of 10'3 for a sampling rate of 44.1 kHz. It is an advantage of the iterative algorithm above that it is linear, that is, it does not require the calculation of any trigonometric functions, square roots or the like. It is another advantage that the above iteration produces a normalized weight vector w, since the term - pw (kl) y (kl) in equation (2) corresponds to a weight decay term penalizing large weights, while the term + μ x (kl) triggers the weight vector in the direction of the main component. It is also noted that in the present embodiment, since x (k) is the wrapper signal, wL, wR € [0,1], that is, the weight vector w is in the first quadrant in figure 3. , thereby ensuring that μ is positive. It is another advantage of this embodiment that it is sufficient to transmit one of wL and wR because the other factor can be determined accordingly. Alternatively, the angle α can be transmitted.

Again with reference to FIG. 2, circuit 200 provides the determined angle α or, alternatively, one or both of the weight factors wL and wR. Angle information is fed into the rotation circuit 201 which generates the rotated signal components y and r. It is understood that circuits 200 and 201 can be combined into a single circuit that performs the iterative calculation of equation (2) and the calculation of y and r according to equation (1). The arrangement also comprises encoders 202 which perform proper encoding of the y and r signals respectively. For example, the signals may be encoded according to MPEG, for example layer and MPEG Wind Power Installation (MP3), according to sine coding (SSC), or subband based, parametric audio coding schemes, or transformation schemes, or any other suitable schemes or combinations thereof. It is understood that encoders 202 may be of the same or different types, for example an MP3 encoder and an SSC encoder, etc. The resulting encoded signals y and re respectively are fed into a combiner circuit 204 together with the angle information α. Combiner circuitry 204 performs framing, bit rate allocation, and lossless coding, resulting in a combined signal T to be communicated. In one embodiment, the angle α or, alternatively, wL and / or wR may be communicated as part of a header previously transmitted to a signal frame, signal block or the like. According to the invention, when the transformation angle α is tracked so that the main component signal includes most of the signal energy, the bit rates allocated to the yer signals may be selected to be different, thus optimizing coding efficiency.

For example, the stereo signals L and R may be expressed as L = M + SeR = M-S, where M corresponds to a medium or central signal and S corresponds to a stereo or lateral signal. In the case of acoustic recording from a stationary sound source, for example a speaker recorded by two microphones, the L and R signals are substantially the same if the speaker is positioned exactly between the microphones and assuming no distortion. such as reflections, etc. Accordingly, in this case, S is substantially zero or at least small, and the coding scheme according to the invention substantially yields y corresponding to L + R and r corresponding to L - R being zero or small; this corresponds to α = 45 degrees. If the speaker is not positioned exactly between the microphones, that is, there is an asymmetry, but still assuming that there are no reflections or other distortions, the rotated signal y according to the invention also corresponds to the speaker and the speaker. residual signal r is substantially zero. However, in this case, the angle α differs from 45 degrees. If the sound source moves, for example, from left to right, the method according to the invention still produces a main component signal y corresponding to the source, and a small residual signal r, ideally being r = 0. In this case, α changes from 0 (fully left) to 90 ° (fully right). The above example illustrates the advantage of tracking angle a. In the above situations, it may be sufficient to transmit the main component signal y and the angle a to allow the decoder to reconstruct the original R and L signals without a significant loss in quality. In general, because the residual signal r is small compared to the main signal, the bit allocation or coding efficiency is negotiable between y and r. Accordingly, it is an advantage of the invention that it allows efficient coding of stereo signals. Fig. 4 shows a schematic view of an arrangement 107 for decoding a stereo signal according to the first embodiment of the invention. The arrangement receives a T-encoded stereo signal, for example originating from an encoder according to the embodiment described in connection with FIG. 2. The arrangement comprises a circuit 404 for extracting the encoded signals y and re and the angle information. c from the combined signal T, that is, circuit 404 performs an inverse operation of combiner 204 of FIG. 2. The extracted signals y and re are fed into the corresponding decoders 402 performing audio decoding, corresponding to the encoding performed by encoders 202 of Figure 2, resulting in the decoded main component signal y 'and the decoded residual signal r \ Signals y', r 'and angle information α are fed into a rotation circuit 401 which rotates signals y', r 'back towards the original L and R components, thus resulting in the received signals L' and R '. Figure 5 shows a schematic view of an arrangement for coding a stereo signal according to a second embodiment of the invention. The arrangement comprises circuits 201 performing a rotation of stereo signals L and R by an angle α, resulting in rotated signal components, as described in connection with Figure 2. The arrangement also comprises circuits 200 for determining the rotation angle, as described in connection with figures 2 and 3. According to this embodiment of the invention, it is recognized that the residual signal r may be estimated as a filtered version of the main signal y. As described in connection with Figure 2, in an acoustic recording of an audio source recorded by two microphones in the absence of acoustic distortion, for example due to reflections, etc., the main signal y corresponds to the audio source and the signal. residual is substantially zero. However, in a more realistic situation, distortions are present, for example due to signal reflections on the walls of an enclosure and on the speaker's head and torso, etc. These effects influence the residual signal r. Consequently, when estimating the residual signal by means of a filter, the filter in effect models the acoustic characteristics of the enclosure, etc. For a classical orchestra, the situation is similar, whereas in the case of modern pop music, the situation may be slightly different. In this case, a sound engineer typically mixes multiple channels on two channels, often using artificial reverberation, soundboxes, and so on. In this case, the filter models the acoustic effects introduced by the mixing process.

Still referring to Fig. 5, the arrangement comprises an adaptive filter 501 which receives the main signal y as an input and generates an Ar filtered signal. The filter parameters Fp of the adaptive filter are selected so that the filtered signal Ar approaches the residual signal r, for example, by controlling the adaptive filter 501 by the error signal and indicating the difference between re Ar as generated by a subtraction circuit 502. The resulting filter parameters Fp are fed into a circuit. of combiner 204, preferably after suitable encoding by means of an encoder 503, for example an encoder providing a Huffman encoding or any other suitable encoding scheme. The filter 501 may be any suitable filter known in the art. Examples of such filters include an adaptive or fixed response filter, with shear frequencies and magnitudes being recursively fixed or tracked, or the like. The filter may be of any order, preferably smaller than 10. The filter type may be Butterworth, Chebychev, or any other suitable type of filter. The arrangement also comprises an encoder 202 for encoding the main signal as described in connection with Figure 2, resulting in the encoded main signal y1 which is fed into the combiner circuit 204 together with the filter parameters Fp and angle information. The. As described in connection with FIG. 2, the combiner circuit 204 performs framing, bit rate allocation, and lossless coding, resulting in a combined signal T to be communicated, which includes the encoded main signal y, the parameters of Fp filter and angle information a. According to this embodiment of the invention, the bit rate to be allocated to filter parameters Fp may be considerably lower than the bit rate required for the main signal y, for example, in one embodiment, the rate bit to Fp may, on average, be less than 10% of the bit to y rate. Accordingly, it is an advantage of the invention that it reduces the bit rate required for the transmission of a stereo signal. The total bit rate according to the invention is only slightly higher than for a single mono channel. It is noted, however, that this ratio may vary during recording. For example, the ratio may become smaller, for example in a situation with few distortions and a stationary source, but may also become larger, for example, if the L and R signals are momentarily independent. Fig. 6 shows a schematic view of an arrangement 107 for decoding a stereo signal according to the second embodiment of the invention. The arrangement receives a T-coded stereo signal, for example from a coding according to the embodiment described in connection with FIG. 5. The arrangement comprises a circuit 404 for extracting the coded signal y, the filter parameters Fp. , and the angle information a from the combined signal T, that is, circuit 404 performs an operation opposite to that of combiner 404 of FIG. 5. The extracted signal y is fed into a decoder 402 to perform audio decoding, corresponding to encoding performed by encoder 202 of Fig. 5, resulting in the decoded main component signal y \ Preferably, the filter parameters are decoded by means of a decoder 602 corresponding to the encoding of filter parameters by encoder 503 of Fig. 5. Signal y 'is fed into a filter 601, along with the received filter parameters Fp. Filter 601 generates a corresponding estimated residual signal r. The received main component signal y ', the estimated residual signal r' and the received angle information α are fed into a rotation circuit 401 which rotates or rotates the signals y ', r' back towards the original components. L and R, thus resulting in the received signals L 'and R \ In the embodiment described in connection with Figures 5 and 6, filters 501 and 601 may be a standard adaptive filter in the time or time domain (see, for example). , “Adaptive Filter Theory” by S. Haykin, Prentice Hall, 2001), for example, a known adaptive filter from the echo cancellation field. Other examples of filters include a fixed FIR or IIR filter with a fixed or adaptive cutoff frequency and magnitude. Alternatively, in one embodiment, the filter may be based on a psycho-acoustic model of the human auditory system, for example as known as MPEG coding, thereby reducing the number of filter parameters. According to yet another embodiment, the filter is further simplified, for example by the use of a 10th order filter, which uses 5 BiQuadratic filters and an artificial reverb unit. In this embodiment, on the coding side, the filter is fitted and the reverberation time is determined. These parameters are slowly variable, thus reducing the bit rate required for their transmission.

Figures 7a-c show schematic views of examples of a filter circuit for use in an embodiment of the invention.

In the example of Figure 7a, filter 501 comprises a combination of a filter 701 and a reverb filter 702. For example, filter 701 may be a standard time or time domain adaptive filter, a fixed FIR or IIR filter with a fixed or adaptive cutoff frequency and magnitude, etc., for example a high pass filter. According to this embodiment, both filter filter parameters 701 and reverberation filter parameters 702, such as the reverberation time denoted by T6o, are transmitted to the decoder as filter parameters Fp.

In the example of figure 7b, in addition to filters 701 and 702, two control circuits 703 - 704 are added. A control circuit 703 is added to ensure that the average residual signal power r and the average reverberator output power 702 are approximately the same, for example by multiplying the reverberator output 702 by a parameter βι. A second control circuit 704 multiplies the scaled reverberator output by β2. The factor β2 can be selected from - 3dBe + 6dB, and is determined so that the cross correlation p between r and r is as high as possible, ie that the signals r r are as similar as possible. Accordingly, the filter arrangement of Fig. 7b also comprises a circuitry 705 for determining cross correlation p. The filter arrangement also comprises a multiplier 706 for generating the product β = βι · β2, which is effective power as a part of the filter parameters Fp. Consequently, βι is a gain that is automatically controlled, for example by comparing the absolute mean of r and r, and β2 is another gain that is automatically controlled, for example by using the cross-relation coefficient p. The first gain is intended to make sure that the energy of r is preserved, that is, that the predicted signal energy r 'at the receiver corresponds to the energy of r. The second gain is to make sure that r and r 'are well correlated.

In one embodiment, reverb 702 and filter 701 may be fixed, i.e. not adapted according to filter parameters Fp. In addition, β2 can be fixed, thus leaving the slowly variable parameter βι as the only adaptive parameter that needs to be adjusted and transmitted. Accordingly, a particularly simple arrangement and filter is provided. It is an advantage of this embodiment that it only requires about half of the original stereo bit rate r to transmit a stereo signal. It is noted that other variations of the above embodiment may be used. For example, in one embodiment, filter 701 may be left out.

In addition, alternatively or in addition to the p-correlation, other correlation measures can be used to ensure a high degree of similarity between the original signal and the signal after coding-decoding. For example, in one embodiment, two correlators may be used in place of the correlator 705. One correlator may compute the cross correlation Plr of the input signals L and R. In addition, a second correlator may compute the cross correlation p ' 1r of the resulting outputs L 'and R' of the encoder-decoder, that is, according to this embodiment, the encoder also comprises a decoder circuit for determining the signals L 'and R'. This embodiment uses the difference ερ = pLR - p'lr to control β2, so that ερ is minimal. This is illustrated in Fig. 7c, where the correlator of Fig. 7b is replaced by circuit 707, which receives the signals L and R as well as L 'and R' as inputs, and generates, as an output, a signal indicating the difference. ερ. output ερ of circuit 707 controls circuit 704 to encrust the estimated residual r, so that ερ is minimized. In one embodiment, the inputs in circuit 707 are high pass filtered, for example at 250 Hz, so that low frequencies have a decreasing contribution to ερ. As in the embodiment of figure 7b, it is an advantage of this embodiment that the correlation between the resulting stereo image and the original stereo image prior to encoding-decoding is very high. Figure 8 shows a schematic view of an arrangement for encoding a stereo signal according to a third embodiment of the invention. The arrangement is a variation of the embodiment described in conjunction with Figure 5, and it comprises circuitry 201 for performing rotation of stereo signals L and R, circuitry 200 for determining rotation angle, an adaptive filter 501, a subtraction circuit 502, an encoder 202, an encoder 503, and a combiner circuit 204 as described in connection with FIG. 5. According to this embodiment, the main component signal y is not directly fed to filter 501. Instead, the arrangement also comprises a decoder 402, as described in connection with Figure 6. Decoder 402 receives the encoded main component signal y, generated by encoder 202, and generates the decoded main signal y ', which filter 501. It is an advantage of this embodiment that it reduces the effect of coding errors introduced by the coding and decoding of the y signal. These coding errors cause the decoded signal y 'to be slightly different from the original signal y, due to the fact that decoder 402 in practice is not a perfect inverse of encoder 202, that is, E E'111. Accordingly, by applying a y signal encoding and decoding to the decoder, the input y 'for filter 501 corresponds to the input y' fed to filter 601 on the receiver, thereby improving the result of signal r 'prediction. receiver. Accordingly, the encoder according to this embodiment may be used in connection with a decoder according to the embodiment of Figure 6. Figure 9 shows a schematic view of an arrangement for encoding a stereo signal according to a fourth embodiment of the invention. The arrangement is a variation of the embodiment described in connection with Figure 5, and comprises circuits 201 for performing a rotation of stereo signals L and R, circuits 200 for determining the angle of rotation, an adaptive filter 501, a subtraction circuit 502, an encoder 202, an encoder 503, and a combiner circuit 204 as described in connection with Figure 5. According to this embodiment, the main component signal y is not directly fed to filter 501. Instead furthermore, the arrangement also comprises a multiplication circuit 901 which multiplies the residual signal r received from circuit 201 by a constant γ, and an addition circuit 902 for adding the scaled residual signal to the main component signal y, resulting in a signal y + γτ, which is fed into filter 501. Here, γ is a small positive value, for example on the order of 10 '. In one embodiment, the constant γ is adaptively traced. It is an advantage of this embodiment that frequencies, which are not substantially present in the y signal spectrum, but present in the r spectrum, can be used in modeling the residual signal r by filter 501, thereby improving the quality. of the coded signal. According to this embodiment, the signal y + γ r is fed into the encoder 202, which generates the decoded main signal y to be transmitted to the receiver. Furthermore, according to this embodiment, the constant γ is fed into the combiner 204 and transmitted to the receiver. Figure 10 shows a schematic view of an arrangement for decoding a stereo signal according to the fourth embodiment of the invention, that is, suitable for decoding a signal received from an encoder according to Figure 9. The arrangement comprises a circuit 404 for extracting the information received from the combined signal T, a decoder 402, a decoder 602, a filter 601, and a rotation circuit 401, as described in connection with FIG. 6. In the embodiment, circuit 404 also extracts the constant γ from the combined signal T, and the arrangement also comprises a multiplication circuit 1001 for multiplication of the predicted residual signal r \ generated by filter 601 by the received constant γ. The arrangement also comprises a circuit 1002 for subtracting the resulting scaled predicted residual signal γ r 'from the decoded main signal y. Figure 11 shows a schematic view of an arrangement for encoding a multi-channel signal of according to a fifth embodiment of the invention. The arrangement receives a multi channel signal x = (xj, ..., xn) comprising n channels. The arrangement comprises a principal component analyzer 1100 for performing a principal component analysis of signal x, resulting in a weight vector w = (wi, ..., wn) to transform input signal x into a principal component signal. ye n-1 residual signs rb r2, ..., rn_i. The arrangement also comprises a transformation circuit 1101 receiving the input signal x and the given weight vector w, and generating the signals y and r1} ..., rn.i, according to the above transformation. Transformed signals are encoded by suitable encoders 202 and 1102, and combined by means of a combiner circuit 204 together with the weight vector w as described in connection with Figure 2. According to this embodiment, the encoder 1102 is adapted to encode the residual signals rb ..., rn „i. For example, encoder 1102 may comprise n-1 parallel encoders, each encoding one of the residual signals, as described in connection with encoder 202. Figure 12 shows a schematic view of an arrangement for encoding a multi-channel signal of according to a sixth embodiment of the invention. In addition to transforming a multi-channel signal, as described in connection with Figure 11, according to this embodiment, the main component signal is fed into a set of adaptive filters 501, each predicting one of the residual signals. η, ..., rn.i, as described in connection with figure 5, resulting in corresponding filter parameters Fpi, ..., FP (n.]), which are fed into corresponding encoders 503 and subsequently at combiner 204. In a matching decoder (not shown), matching filters are used to generate estimates of r 'j, ... r'n_i of residual signals, based on filter parameters, as described in connection with Figure 6. It is understood that according to one embodiment, only a subset of residual signals, for example rh ..., rk, k <n-1, can be transmitted to the receiver or fed into corresponding filters, thereby reducing the required rate of b while maintaining the majority of signal quality. Figure 13 shows a schematic view of a subtraction circuit for use with an embodiment of the invention. In the above embodiments of Figures 5, 8, 9 and 12, the filter parameters are determined by comparing a target signal with an estimated signal ie by means of the error signal and indicating the difference between rer t when generated by a subtraction circuit 502. It is understood that the subtraction circuit may generate different measures of difference between rer, for example a difference may be determined in the time domain or the frequency domain. Referring to Fig. 13, circuit 502 may comprise circuitry 1301 for transforming signals r and r respectively in the frequency domain, for example by performing a fast Fourier transform (FFT). The resulting frequency components may also be processed by respective circuits 1304. For example, different frequencies may be differently weighted, preferably according to the properties of the human auditory system, thereby weighting differences in the audible frequency range more strongly. Other examples of further processing by circuits 1304 include averaging predetermined frequency components, calculating the magnitude of complex frequency components, grouping of filter components, or the like. For example, in a preferred embodiment, a grouping is performed before subtraction in the frequency domain. This grouping can be performed using a filter bank, for example with linear or logarithmic bandwidths. Alternatively, grouping can be performed using so-called rectangular equivalent bandwidth (ERB) (see, for example, Brian Moore's An Introduction to the Psychology of Hearing, Academic Press, London, 1997). The equivalent rectangular bandwidth technique groups frequency bands that correspond to human auditory filters, for example the so-called critical bands. According to this embodiment, the corresponding ERB value, as a center frequency function, f (in kHz) can be calculated according to ERB = 24.7 (4.37 f + 1). Still referring to Fig. 13, circuit 502 also comprises a subtraction circuit 1303 for subtracting the processed frequency components. Alternatively, the transformed signals generated by circuits 1301 are directly fed into subtraction circuit 1304 without further processing. The difference signal generated by subtraction circuit 1304 is fed into a transformation circuit 1302 to transform the error signal back to the time domain, for example by executing an inverse fast Fourier transform (IFFT). Alternatively, the frequency domain difference signal can be used directly. It is understood that one skilled in the art can adapt the above embodiments, for example by adding or removing features, or by combining features of the above embodiments. For example, it is understood that the features introduced in the embodiments of figures 8 and 9 may also be incorporated in the embodiment of figure 12. As a further example, the error signal e, which describes the estimated residual signal quality in the 5, can be compared to a limit error indicating a maximum acceptable error. If the error is not acceptable, the error signal may, after proper coding, be transmitted together with the T signal, similar to methods used within the Linear Predictive Coding (LPC) field. It is also noted that the invention is not limited to stereo signals, but may also be applied to other multi-channel input signals having two or more input channels. Examples of such multi channel signals include signals received from a Digital Versatile Disc (DVD) or a Super Audio Compact Disc, etc. In this more general case, a main component signal y and one or more residual signals r may also be generated according to the invention. The number of residual signals transmitted depends on the number of channels and the desired bit rate, as higher order residues can be omitted without significantly degrading the signal quality.

In general, it is an advantage of the invention that bit rate allocation can be adaptably varied, thereby allowing for elegant degradation. For example, if the communication channel momentarily only allows a low bit rate to be transmitted, for example due to high network traffic, noise, or the like, the bit rate of the transmitted signal may be reduced without significantly degrading the transmission rate. noticeable signal quality. For example, in the case of a stationary sound source discussed above, the bit rate may be reduced by a factor of approximately two, without significantly degrading signal quality, corresponding to the transmission of a single channel rather than two. It is noted that the above provisions may be implemented as general purpose or special programmable microprocessors, Digital Signal Processors (DSP), Application Specific Integrated Circuits (ASIC), Programmable Logic Arrays (PLA), Field Programmable Gate Arrangements ( FPGA), special purpose electronic circuits, etc., or a combination thereof.

It should be noted that the aforementioned embodiments more properly illustrate the limit of the invention, and that those skilled in the art will be able to design many alternative embodiments without departing from the scope of the appended claims. In the claims, any reference signs enclosed in parentheses should not be construed as limiting the claim. The word “understanding” does not exclude the presence of elements or steps other than those listed in a claim. The invention may be implemented by hardware comprising several distinct elements, and by means of a suitably programmed computer. In a device claim that enumerates multiple media, several of these media may be incorporated by one and the same hardware item. The mere fact that certain measures are mentioned in mutually different dependent claims does not indicate that a combination of these measures cannot be used to advantage.

Claims (16)

Method for encoding a multi-channel signal including at least one first signal component and a second signal component, the first signal component and the second signal component being time domain signal components characterized in that comprises the steps of: transforming in the time domain at least the first and second signal components by means of a predetermined transformation into a main signal including most of the signal energy and at least one residual signal including less energy than the signal principal, the predetermined transformation being parameterized by at least one transformation parameter; and representing the multi-channel signal by at least the main signal and the transformation parameter. Method according to claim 1, characterized in that it further comprises the step of adaptively determining the transformation parameter based on at least the first and second signal components. Method according to claim 1 or 2, characterized in that the main signal corresponds to a main component of the first and second signal components. Method according to any one of claims 1 to 3, characterized in that the predetermined transformation is a rotation and the transformation parameter corresponds to an angle of rotation. Method according to any one of Claims 1 to 3, characterized in that the step of representing the multi-channel signal by at least the main signal and the transformation parameter comprises the step of representing the multi-channel signal by the main signal. , by the transformation parameter and the residual signal. Method according to claim 5, characterized in that the step of representing the multi-channel signal by the main signal, the transformation parameter and the residual signal also comprises the steps of: encoding the main signal with a first rate bit; and encoding the residual signal with a second bit rate lower than the first bit rate. Method according to any one of claims 1 to 6, characterized in that the main signal corresponds to a first signal energy and the residual signal corresponds to a second signal energy smaller than the first signal energy. Method according to any one of claims 1 to 7, characterized in that: the method also comprises the step of estimating the residual signal from the main signal using a prediction filter corresponding to a set of filter parameters; and the step of representing the multi-channel signal by at least the main signal and the transformation parameter comprises the step of representing the multi-channel signal by the main signal, the transformation parameter and the filter parameter set. Method according to any one of claims 1 to 8, characterized in that the multi-channel signal comprises a stereo signal including a left and a right signal component. Method for decoding multiple channel signal information, characterized in that it comprises the steps of: receiving a main signal and a transformation parameter, the main signal corresponding to a result of a predetermined transformation of at least one first and a second time domain signal component of a multi-channel source signal, the predetermined transformation being parameterized by at least the transformation parameter; and generating first and second decoded time domain signal components by inverse transformation of the received main signal and at least one residual signal. A method according to claim 10, characterized in that the step of receiving a main signal and a transformation parameter also comprises the step of receiving the residual signal. The method according to claim 10 or 11, characterized in that the step of receiving the main signal and the transform parameter also comprises the step of receiving a set of filter parameters, and the method also comprises the step of residual signal prediction from the main signal using a prediction filter corresponding to the received set of filter parameters. Arrangement for encoding a multi-channel signal including at least one first signal component and a second signal component, the first signal component and the second signal component being time domain signal components, characterized in that comprising: first processing means adapted to transform in the time domain at least the first and second signal components by a predetermined transformation into a main signal including most of the signal energy and at least one residual signal including less energy than the main signal, the predetermined transformation being parameterized by at least one transformation parameter; and second processing means adapted to represent the multi-channel signal by at least the main signal and the transformation parameter. Arrangement for decoding multiple channel signal information, characterized in that the arrangement comprises: receiving means for receiving a main signal and a transformation parameter, the main signal corresponding to a result of a predetermined transformation of a first and second signals in the multiple channel source time domain, the predetermined transformation being parameterized by at least the transformation parameter; and processing means for generating first and second multi-channel time domain signals by reverse transformation of the received main signal and a time domain residual signal. Data signal including multi-channel signal information, characterized in that a multi-channel signal comprising at least one first signal component and a second signal component, which comprises the steps of coding, is encoded by a method for encoding. : transforming at least the first and second signal components by a predetermined transformation into a main signal including most of the signal energy and at least one residual signal including less energy than the main signal, the predetermined transformation being parameterized by means of at least one transformation parameter; and representing the multi-channel signal at least the main signal and the transformation parameter, resulting in said data signal. Device for communicating a multi-channel signal including at least one first signal component and a second signal component, the first signal component and the second signal component being time domain signal components, characterized in that comprise an arrangement for encoding the multi-channel signal, the arrangement including: first processing means adapted to transform in the time domain at least the first and second signal components by a predetermined transformation into a main signal including most of signal energy and at least one residual signal including less energy than the main signal, the predetermined transformation being parameterized by at least one transformation parameter; and second processing means adapted to represent the multi-channel signal by at least the main signal and the transformation parameter.