KR20010023723A - 5-2-5 matrix encoder and decoder system - Google Patents

Abstract

At least one signal component is directionally encoded and correlated, and at least one signal component is not directionally encoded and stereo signals of two input channels that are uncorrelated in two input channels to signals for a few output channels. Sound playback system to convert

Increasing the correlation component of the input signal in the desired direction and reducing the strength of the signals of the channel not associated with the encoded direction, while the tools recorded in the right input channel remain on the right side of the output channel and record on the left side. And the separation between each left and right output channel so that the distinct loudness of all the tools of all the output channels is maintained independent of the direction of the directionally encoded components of the input signal. A decoding device for maintaining the overall strength of the uncorrelated component of the input channel at the output channel;

Encoding means for encoding five input channels to be decoded in a correction direction and level in a decoder according to the invention and a decoder according to the current film standard,

The decoding device optimally adjusts the level of the center channel to maintain the disappearance of the center component of the input signal at the intensity ratio of the input signal for film reproduction or the left and right front output for the music signal. It also includes a frequency dependent element at the surround output to improve reproduction of the lateral and rear signal components in the speaker bourbon.

Description

5-2-5 matrix encoder and decoder system {5-2-5 MATRIX ENCODER AND DECODER SYSTEM}

See related applications

This application is based on US Provisional Patent Application No. 60,058,169, filed Sep. 5, 1997 entitled “5-2-5 Matrix Encoder and Decoder System”.

The present invention maintains high separation between the various outputs of the signal in the defined direction and directly encodes the input audio signal, even when there is pure forward or reverse bias as the input signal, or when there are strong sound components in a particular direction. Improves the balance between the front and rear signals, including maintaining high separation between the left and right components of the stereo signal, while maintaining the omni-directional encoded components at a constant sound level regardless of the direction of the components. Provides a smooth sound condition around the 7-channel version and produces standard 5-channel sound that is closer to the 7-channel sound, as well as the standard 2-channel component, including maintaining high separation between the left and right components of the stereo signal under all circumstances. As well as optimal mental sound to reproduce encoded multichannel components It relates to design criteria and an improved set of solutions for generating encoding matrices with directional performance.

The invention also relates to an improved set of design criteria and solutions for generating encoding circuitry for encoding multichannel sound into two channels for reproduction in a standard two channel receiver by means of a decoder according to the invention.

The present invention is part of a continuing effort for refinement means which is a means of encoding a multichannel audio signal into two separate channels and again separating these two channels into an extracted multichannel signal. One of the purposes of this encode / decode process is to recreate the original signal so that it is recognized as identically as possible by the original signal. Another important purpose of this decoder is to extract five or more separate channels from a two-channel source that has not yet been encoded from the original five channels. The extracted five channel expression should be at least musically felt and enjoyed like the original two channel expression.

The present invention relates to improving the derivation of appropriately usable matrix coefficients. In order to assist understanding of such an improvement, the following literature is referred to in the present specification. Grisinger, U.S. Patent No. 4,862,502 (1989) (referenced as '89), U.S. Patent 5,136,650 (1992) (referenced as '92), July 1996 Grishinger U.S. Patent Application 08 / 684,948 Commercial versions of decoders based on this latest application, such as the November 1996 U.S. Patent Application 08 / 742,460 (referred to as November, 1996) and Grisinger's November 1996 application V1.11). Another enhancement is disclosed in Provisional Patent Application 60 / 058,169, filed in September 1997, referred to as version 2.01 (or V2.01). These versions, together with the V1.11 and V2.01, will be referred to as "Logic 7" decoders.

Other technical references include: [1] “Multichannel Matrix Surround Decorders for Two-Eared Linsteners”, October 1996 AES preprint # 4402 by David Grisinger, and [2] September 1997 AES preprint # 4625 by David Grisinger. "Progress in 5-2-5 Matrix Systems".

Summary of the Invention

This means fulfills the two objectives of reproducing the original signal encoded in five to two channels and creating a satisfactory reproduction of the two-channel component in the five-channel format, which allows the human to continue to understand physically and psychoacoustically. It is to. The aforementioned patents and patent applications represent the design philosophy of making useful decoder devices.

The present invention is concerned with the realization of an active matrix having the property of maximizing psychoacoustic performance. In another aspect, disclosed herein is a frequency independent modulation of a portion of the output from the active matrix. In another aspect, the present invention encodes five input channels into two output channels with both standard two-channel equipment and a decoder according to the invention, and with the industry standard "Dolby Pro-Logic" decoder module. An active circuit group is provided.

The present invention is part of an active matrix decoder with matrix elements that change in accordance with the directional component of the input signal. This matrix element improves the loudness of the signal in the direction involved to compensate for the intended direction, while preserving the left / right separation of other signals that may be present at the same time at the same time, while maintaining the desired direction in the intended direction. It changes to produce loudness of a directionally encoded signal in an output that is not included. Moreover, the matrix element according to the invention stores the left / right separation of directionally encoded uncorrelated two-channel components by increasing or decreasing the mixing between the two inputs, for example two inputs such as stereo width control. do. The matrix element according to the invention is also designed to preserve the energy balance between the various components of the input signal as much as possible so that the balance between voice and accompaniment appears at the decoder output. The matrix element according to the invention thus preserves both the loudness of the omni-encoded element of the input sound and the left / right separation of these elements.

In addition, the decoder according to the present invention improves the compatibility of the decoder output when standard two-channel components are played, changes the surround output from two channels for five-channel decoder to four channels for seven-channel decoder, and A frequency independent circuit is included to modify the spectrum of the back channel so that the sound direction is the same as the sound direction from the seven channel decoder.

The encoder according to the invention mixes five input channels (or one low frequency in five full ranges) into two output channels such that the input level of a particular input is strong such that the energy present in that input is within the output, The direction of the strong input is encoded in the phase / amplitude ratio of the output signal, and the strong signal can be panned between any two inputs of the encoder and the output to be directionally encoded correctly. In addition, the uncorrelated component applied to the two rear inputs of the encoder is encoded in two channels so that the encoder output is left / right separated when the encoded output is decoded by the decoder according to the invention. The in-phase input to the rear channel of the output makes a two-channel output to be decoded into the rear channel of the decoder according to the invention and the decoder according to the Dolby standard, and the anti-phase input to the two rear channels of the encoder is In addition, a low level echo signal applied to the two rear inputs of the encoder is 3 dB at a level within the two channel input to produce an output corresponding to an omnidirectional signal of the encoder yaw according to and corresponding to a decoder according to the Dolby standard. Preserved to be encoded as a reduction.

The present invention is directed to a sound reproduction system comprising not only decoding an audible input audio signal for reproduction after being amplified through a plurality of speakers arranged around a listener into a plurality of output signals, but also encoding multiple channel components into two channels. It is about.

The novel features of the invention are set forth in the appended claims. The invention and other features of the invention will be well understood from the detailed description of the embodiments described with reference to the accompanying drawings,

1 is a block schematic diagram of a direction detection portion and a 2 to 5 channel matrix portion of a decoder according to the present invention without the aspects further shown in FIGS. 2 and 3;

2 is a block schematic diagram of a five channel frequency dependent active signal processor that is also coupled between the output of the matrix section of FIG. 1 and the decoder output;

3 is a block schematic diagram of a 5-7 channel frequency dependent active signal processor that may alternatively be connected between the output of the matrix section of FIG. 1 and the decoder output;

4 is a block schematic diagram of a two-channel encoder in an active five-channel according to the present invention;

5 is a three-dimensional schematic representation of a prior art Left Front Left (LFL) matrix device from the '89 patent and Dolby Pro Logic, referenced to a maximum value of 1;

6 is a three-dimensional schematic representation of a prior art Left Front Right (LFR) matrix device from the '89 patent and Dolby Pro Logic, .07 standardized so that the minimum value is -0.5 and the maximum value is +0.5,

7 is a three-dimensional schematic representation of the square root of the sum of the conventional LFL and LFR squares from the '89 patent referenced to a maximum value of 1;

8 is a three-dimensional graphical representation of the square root of the sum of the LFL and LFR from Application No. 08 / 742,460, standardized to have a maximum value of 1;

9 is a three-dimensional schematic representation of the LFL matrix device of V1.11,

10 is a three-dimensional schematic representation of a partially completed LFL matrix device of the present invention,

11 is a graph showing the operation of the LFL and LFR of the present invention along the posterior vicinity between the left and the entire rear;

12 is a three-dimensional schematic representation of a fully completed LFL matrix element of the present invention as viewed from the left rear;

13 is a three-dimensional schematic representation of a fully completed LFR matrix device of the present invention,

14 is a three-dimensional schematic representation of the mean square sum of LFL and LFR in accordance with the present invention;

15 is a three-dimensional graphical representation of the square root of the sum of LFL and LFR squares according to the present invention, including correction to the rear level seen from the left rear;

16 is a graph showing a solid line which is a central matrix element to be used in a conventional Dolby Pro Logic decoder as a dB function of CS and a dotted line which is an actual value of the center matrix element in the Dolby Pro Logic decoder;

17 is a graph showing a solid line, which is an ideal value of the center matrix element, and a dotted line, which is an actual value of the center matrix element in the conventional Dolby Pro Logic;

18 is a three-dimensional graphical representation of the square root of the sum of the LRL and LRR product products using the conventional device of V.11,

19 is a graphical representation of several solutions of GS (lr) and GR (lr) for zero output along a constant intensity level along cs = 0 and along the boundary between left and center,

20 is a three-dimensional graphical representation of the square root of the sum of the LRL and LRR product products using the values of GR and GS according to the present invention,

FIG. 21 is a three-dimensional schematic representation of a conventional Center Left (CL) matrix element of the '89 patent 4-channel decoder (and the Dolby Pro Logic decoder), which may also represent a left and right CR (Center Right) matrix element;

22 is a three-dimensional schematic representation of the CL matrix element in a Logic 7 V1.11 decoder;

23 is a graph showing the solid line which is the center output channel attenuation required for the new LFL and LFR and the dotted line which is the center attenuation for the conventional standard Dolby Pro Logic decoder,

24 is a graph showing a solid line which is an ideal center attenuation for the "film" strategy according to the present invention, a dashed line which is a better working value, and a dotted line which is the center attenuation for the standard Dolby decoder for comparison;

25 shows the median attenuation used in the “music” strategy of the present invention;

FIG. 26 is a graph showing a dashed-dotted line as the LFR element sin (cs), a dotted line as the sin (cs) value, and a solid line as the GF value required for a constant energy ratio with the “music” attenuation according to the present invention;

27 is a three-dimensional schematic representation of the LFR in the present invention, including modifications for the center level along the lr = 0 axis,

28 is a three-dimensional schematic representation of the CL matrix device with new central amplifier capability and

Fig. 29 is a graph showing the output level from the left front output (dotted line) and the center output (solid line) as if a strong signal is panned from center to left.

The design presented here reflects most of the design philosophy of the conventional decoder, but the actual design varies in many ways. It's a tremendous amount to make a complete document that fully describes the evolution of this design. To clarify such a document, Applicant intends to show a mathematical solution to the problems known through this specification and to represent the most important elements of the design philosophy claiming on the original solution of the present application. It may be useful to look at the previous applicant's application on this issue, but it is not necessary.

Experience with decoders and encoders as described in the July '96 and November '96 applications and the September '97 patent application resulted in further improvements that have not yet been introduced. This application will present the most essential features of the still unproven encoders and decoders of the present invention and claim new features that have been added since US patent application 08 / 742,460.

1. Introduction to Decoder

The decoder of the present application will be described as being composed of two separate parts. The first part is a matrix, usually identified as medium injuries, which divides two input channels into five output channels. The second part consists of a series of delays and filters that modify the spectrum and level of the two rear outputs. One of the functions of the second part is to drive additional pairs of outputs left and right when a seven channel version of the decoder is required. In the 08 / 742,460 application, the second part is not clear, where the two additional channels were originally derived from an additional pair of matrix elements in the matrix.

In the equations describing the decoder and the encoder, it is assumed herein that the most usable and simple variables are italicized as the most usable standard prints, and the vector quantities are shown in bold lowercase and the matrix in bold uppercase. Matrix elements that are coefficients from the named output channel resulting from the named input channel will be represented in standard lowercase letters. Some simple variables, such as lr and cs, are represented by two characters rather than the meaning of two separate simple variables. In some cases the other variables l / r and c / s represent left-right values and center-surround ratios, but they are also in the form of control signal voltages derived from these ratios. This rule is also used in the earlier US patents and patent applications mentioned herein. Program segments in the Matlab language will also be distinguished by using various types and sizes and marking these lines. Equations will be numbered to distinguish them from matlab designation sentences and to provide references for specific features described herein.

1, which is the same as FIG. 4 of US patent application 08 / 742,460, shows a two-channel block diagram of a five-channel matrix 90, which is the first part of the decoder. The left half of FIG. 4, divided by a vertical dotted line, shows the means for inducing two steering voltages l / r and c / s. This voltage indicates that the input signal has uniqueness or represents encoded directional components in the left / right or forward / rear directions, respectively. The description of this part in the drawings is sufficiently explained in the above-mentioned patent application referred to in this specification and will not be described separately.

In Fig. 1, the direction detecting means of the decoder 90 is composed of elements 98-138 that follow a 5x2 matrix to the right of the vertical dotted line. These matrix elements 140-158 are linearly coupled with other input channels to determine the amount of each input channel forming each output channel. These matrix elements are assumed to be real. (The case of complex matrix devices is described in US patent application 08 / 742,460 and not described herein.) The matrix devices function as two steering voltages l / r and c / s. US patent application 08 / 742,460 shows mathematical formulas for this function. The novelty part of the present application lies in graphically expressing these formulas and explaining how they are shaped.

2. Brief description of steering voltage

As shown in FIG. 1, the steering voltages c / s and l / r are the logarithm of the ratio from the left input at terminal 92 to the right input amplitude at terminal 94 and the sum of the amplitudes. Derived from the logarithm of the ratio of In describing the matrix element, it is effective to describe l / r and c / s at angles varying from + 45 ° to -45 °. In the decoders of V1.11 and V2.01, these voltages have an explanation unit. You can convert this steering parameter to the following angles:

lr = 90-arctan (10 ^ ((lr) / 20))

cs = 90-arctan (10 ^ ((c / s) / 20))

These angles lr and cs determine the extent to which the input signal has a directional component. For example, if the input to the decoder is uncorrelated, both lr and cs are zero. For the signal coming from the center, only lr is zero and cs has a value of 45 °. For a signal coming in from the back, lr is 0 and cs is -45 °. Similarly, the signal coming from the left has an lr value of 45 ° and a cs value of 0, and the signal coming from the right has an lr value of -45 ° and a cs value of 0. We assume that the design has the characteristics of encoding the left rear signal to lr = 22.5 ° and cs = 22.5 ° to the encoder that produces the encoded signal. Similarly, it is assumed that the signal to which the right rear input is applied to the encoder has values of lr = 22.5 ° and cs = 22.5 °.

From the definition of l / r and c / s and the derivation of lr and cs, it can be seen that the sum of the absolute values of lr and cs cannot be greater than 45 °. The allowed values of lr and cs are planes delimited by the formula abs (lr) -abs (cs) = 45 °. Any input signal that produces lr and cs values along the boundary of this plane is completely local, ie composed of a single sound that is encoded to come from a particular direction.

In this application we will use the matrix element as a graph as a function on these two dimensional planes. It is generally difficult to derive the matrix element in the four quadrants on this side. In other words, it is difficult to describe the matrix element depending on whether the steering is forward and whether it is left or right. Considerable efforts have been made to ensure that the face passes continuously through the boundaries between the quadrants. Such intermittent lack of continuity is one of the problems with the V1.11 decoder to be described in this application.

3. Frequency dependent device

The matrix element shown in FIG. 1 is real and independent of frequency. All input signals are directed to the output according to the derived angles lr and cs. (In the current technology, low and high frequencies are attenuated at the lr and cs induction at the input by a filter not shown in FIG. 1, but the matrix itself is wideband.)

We found that applying frequency-dependent circuits to the signal after the matrix has several advantages. One of such frequency dependent circuits-the phase shift network 170 on the right side of the output 180 of FIG. 1-is described in US patent application 08 / 742,460, which is no longer described herein.

2 shows a five channel version of an additional frequency dependent circuit. There are no fixed parameters in these circuits. Frequency and level operation depends on the steering values lr, cs. This circuit serves several purposes. First, in both 5-channel and 7-channel decoders, the additional element allows the apparent loudness of the rear channel to be adjusted if the steering is neutral (lr and cs are zero) or forward facing (cs> 0). In US patent application 08 / 742,460 this attenuation is performed as part of the matrix itself and is frequency independent. We have done the theoretical research and listening tests that are best suited for the low frequencies to be reproduced on the listener side. Thus, the decoder shown herein attenuates only high frequencies by the variable low pass filters 182, 184, 188, 190.

This attenuates frequencies above 500 Hz in the back channel by elements 188 and 190 and above 4 kHz by elements 182 and 184 using the background control signal 186, which will be described later herein, if steering is almost always neutral or forward. Is done by letting The rear steered sound, which sometimes appears, reduces the attenuation, which is a property that automatically distinguishes the encoded surrounding components from conventional two channel components.

The other five-channel versions of the elements 192, 194 change the spectrum of the sound when steering is directed backwards using the c / s signal 196 (cs < 0) so that the speaker is located on the actual side. Let it be as if you are behind the listener. The changed left surround and right surround signals appear at terminals 198 and 200, respectively. A more detailed description of this circuit will be given in the next chapter.

3 shows a seven-channel version of the frequency dependent device. The first filter set 182, 184, 188, 120 is again controlled by the background control signal 168, as if the steering was neutral or forward attenuating the upper frequencies of the side and rear outputs. This attenuation also allows for more forward sound images and can be adjusted to the listener's preferences. By moving the steering represented by the c / s signal 196 backwards, the additional circuits 202, 204, 206, 208 operate differently from the rear outputs to the side outputs. As the steering is moved backwards, the attenuation previously mentioned in the side speakers is first removed by the elements 204 and 206 to produce the side sound. As the steering is moved further to the rear, the attenuation of the elements 204 and 206 recovers and increases. The result is that the sound will vary slowly from the front speaker to the side speaker (s) and thus to the rear speaker, with a delay of about 10 ms made by the delay elements 202 and 208. Since low frequencies are not affected by this circuit, the low frequency loudness of the side speakers (giving wider perception) is not affected by changes in the sound. Again, further details of the circuit group of FIG. 3 will be described in the next chapter of the present application.

4. General description of the encoder

4 shows a block diagram of an encoder designed to automatically mix five input channels into two input channels. This configuration is very different from the encoder described in US patent application Ser. No. 08 / 742,460. The purpose of this new design is to preserve the sound balance of the first five channels, providing a phase / amplitude signal that allows the first five channels to be extracted by the decoder. Previous encoders had a similar purpose, but had improvements in the method used to achieve these goals. Preservation of sound balance is very important for encoders. One of the main objectives of the encoder is to automatically generate two-channel mixing for five-channel recordings played on a common two-channel system with the same artistic performance as the first five channels. This new encoder design includes an active element that ensures that sound balance is preserved.

Unlike the encoder of the 1997 application, the new design allows the input signal to be panned between any of the five inputs of the encoder. For example, the sound can be panned from the front left input to the rear right input. If the two-channel signal is decoded by the decoder described in this application, it will be very close to the original sound. Also, decoding through the original surround decoder will be similar to the original.

Detailed description of the encoder will be described later.

5. Design Objectives for Decoder Active Matrix Devices

The most basic object of the present invention is the same as that of our previous decoder, in particular the decoder described in US patent application Ser. No. 08 / 742,460. The present invention is directed to reducing the directionally encoded audio component of the output that is not directly related to reproducing the output in the intended direction, and directly reproducing the output in the intended direction to keep the overall strength of the signal constant. While increasing the directionally encoded audio component of the associated output, while maintaining a high separation between the left and right channel components of the non-directional signal independent of the steering signal, whether or not a directionally encoded signal is present or not. Or, if present, a surround sound decoder having a variable matrix value configured to efficiently maintain a constant loudness defined by the omnidirectional overall audio intensity level, regardless of the intended direction.

Most of these purposes are superficially shared by all matrix decoders. What is new in this application depends, in part, on how to more clearly compose the rule and in part when it does not apply the rule. However, much of the methodology of US patent application Ser. No. 08 / 742,460 continues. One of the most important of the previous objectives is the apparent maintenance of high separation between the left and right channels of the decoder under all conditions. It was not possible to maintain separation at the back because all previous four channel decoders were provided with only the signal back channel. The 5-channel decoder in other configurations is a compromise between many schemes of separation. The decoder described in this application approaches this purpose in a similar manner to V1.11 but does not fit the additional purpose well.

US patent application 08 / 742,460 also describes many similar improvements to the design of circuits to improve the accuracy of steering signals and to the design of variable phase shift networks to switch the phase of one of the rear channels during strong rear steering. These features of the V1.11 decoder remain in the new design but are not complete with this application.

In Fig. 4, the front input signals L, C, and R are applied to the input terminals 50, 52, and 54, respectively. L and R are directed to adders 278 and 282 respectively, and C is first attenuated by the exponent of attenuator 372 before being applied to the inputs of both adders 278 and 282. The low frequency effective signal LFE passes through a gain 2.0 of the element 374 and is applied to both adders 278 and 282.

The surround input signals LS and RS are applied through two input terminals 62 and 64, respectively, and the LS signal passes through an attenuator 278 having gains fs (l, ls), and the RS signal is gain fs ( pass through an attenuator 380 with r, rs). Their output is normally passed into cross-coupled devices 384 and 386 with a gain factor of crx of 0.383. The outputs of summers 392 and 394 are applied to the inputs of adders 278 and 282. These elements are located 40 ° left and right, respectively, behind the center in the decoded space.

The other signal branch passes the LS and RS signals through an attenuator 376 with a gain fc (l, ls) and an attenuator 382 with a gain fc (r, rs), respectively, and then center back as described above. Passes a similar arrangement of crosslinking elements 396, 398, 402, 404, 406, 408 with outputs representing the rear left and rear right inputs at 45 ° left and right. However, these signals pass through the phase shifter elements 234 and 246, respectively, and the left and right signals from the adders 278 and 282 pass through the phase shifter elements 286 and 288, respectively. Each of these phase shifter elements is a full pass filter, and the phases represent φ (f) -90 ° for elements 286 and 288 and φ (f) -90 ° for elements 234 and 246. The calculation of the components required for these filters is well known in the art and is not described in detail here. The outputs of summers 406 and 408 cause delays in adders 278 and 282 by 90 ° at all frequencies after passing through the all-pass filter network as shown in FIG. The output of global pass network 234, 286 is combined with summer 276 to produce an A (or left) output signal at terminal 44, and the output of filter 246, 288 is summer 28. Are combined to produce a B (or right) output signal at terminal 46.

The gain functions fs and fc are designed to allow strong surround signals represented by phases with different sounds, with weak surround signals passing through a 90 ° phase shifted path to maintain a constant intensity for the slowed "music" signal. The crx value also changes, changing the angle from which the surround signal is heard.

6. Design improvements after US patent application Ser. No. 08 / 742,460

One of the most significant improvements of the present invention to US patent application Ser. No. 08 / 742,460 is the change in the center matrix and front left and right matrix elements when the signal is steered in the center direction. The inventors have discovered that there are two problems with the central channel when they have been previously encoded and decoded. The most important problem in a five-channel matrix system is that the use of a central channel can create inherent contradictions in maintaining the purpose of left / right separation. If the matrix is to produce a sensitive output from conventional two-channel stereo components, the two input channels must be driven by the sum of the left and right input channels without the left and right components. Thus, both left decoder inputs and right decoder inputs will be played by the center speaker and only the original sound in the left (or right) channel will be played from the center. The result is that these sounds will be pulled into the middle of the thread. This degree of occurrence occurs according to the loudness of the center channel.

In US Pat. Nos. 4,862,502 and 5,136,650, matrix elements having a minimum value of 3 dB compared to the left and right channels are used. When the input to the decoder was reduced, the loudness of the center channel was the same as that of the left and right channels. The forward steer towards the center matrix was increased by 3 dB. This high loudness effect strongly reduces the width of the front image. Instruments played on the left and right sides of the sound image are always drawn towards the center of the sound image.

In US patent application Ser. No. 08 / 742,460, a central matrix element having a minimum value of 4.5 dB less than the initial value was used. This minimum value was chosen as the basis for the listening test. This attenuation spreads pleasantly to the front image when the input component is irrelevant to orchestral music. The forward image is not severely narrowed. In US patent application Ser. No. 08 / 742,460, the forward shifted steering of these matrix elements eventually affects the values used in the Dolby matrix.

The V1.11 decoder, which has shown a reduction in the center channel loudness that solves the spatial problem, is surprising, but the intensity balance in the input signal is not preserved through the matrix. Mathematical analysis does not merely show the V1.11 error, but the Dolby and older decoders also have errors. Paradoxically, the center channel is too strong in terms of reproducing the width of the front image, making the intensity balance preservation too weak. This problem is particularly acute for Mandaly's decoder-standard Dolby decoder. In a standard Dolby decoder, the back channel is stronger than our decoder in US Pat. No. 4,862,502. As a result, the center channel must be strong to preserve the intensity balance. The lack of intensity balance in the center channel has a continuous problem for Dolby decoders. Dolby recommends that sound mix technicians should always listen to the balance through the matrix, so that the lack of intensity balance of the matrix can be compensated for during the mixing process. Unfortunately, modern movies are mixed for five channel releases, and automatic encoding to two channels can lead to problems with dialogue levels.

Further analysis and listening tests on movies and music require different solutions to the balance problem. For the movie, the inventors found from US patent application Ser. No. 08 / 742,460 that it is most useful to preserve the front left and right matrix elements. These devices are capable of removing center channel information from the front left and right channels. This minimizes talk leakage into the front left and right channels. In the new "film" design, the intensity balance is corrected by changing the center matrix element so that the center channel loudness increases more rapidly than standard decoders, such as steering moving forward (cs is zero or more). In particular, the final value of the center matrix element need not be higher than the standard decoder, and this condition is only reached when the center channel is active. The center channel needs to be stronger than the standard decoder when it is at about the same level as the center channel and left and right channels.

With the "film" strategy, the center channel loudness is increased to preserve the intensity balance in the input signal, minimizing the center channel component at all other outputs. This strategy is considered ideal for film, the main use of the central channel is for dialogue, and does not think of dialogue in locations other than the center. The main drawback of this strategy is the frequent central steering, which narrows the forward image, such as occurs in many forms of pop music. However, the advantage for the film is more valuable than this drawback, due to the minimization of talk leakage to the front channel and a good intensity balance.

For music, we tried different strategies. In this case, we increased the center channel loudness to the median value of steering of cs 22.5 ° at the same rate as in US patent application 08 / 742,460. In order to restore the sound balance, the inventors alternated the front left and right matrix elements so that the center component of the input signal was not completely removed. The number of center channel components in the front left and right is adjusted so that the sound intensity from all outputs of the decoder matches the sound intensity in the input signal without excessive loudness at the center.

With this strategy, all three front speakers reproduce the center channel information present in the original encoded component. The most useful version of this strategy is to limit the steering action at the point where the center output is 6 dB stronger than any of the other two front inputs. This is accomplished simply by limiting the cs value.

This new strategy allows the center channel component to be introduced from all three front speakers and provides superiority for all music types by limiting steering when the center is 6 dB stronger than the front left and right. Both the encoded five-channel mix and the original two-channel mix decode into a stable center, satisfying the separation between the center channel and the left and right sides. Unlike previous decoders, the separation between center and left and right was not intentionally completed. The intended signal coming from the left is removed from the center channel in another way. For music, for high side separation and stable front image music, this strategy is more valuable than this leakage of complete separation. Listening tests for this setting in the movie show that the stability of the sound image obtained is very good no matter what dialogue comes from the front left and right speakers. This result is not distracting to be satisfactory. It does not obscure the artistic value of the movie for the listener who listens to the movie with the decoder set for music. Listening to music recordings with a decode set for film is very uncertain.

The next most important improvement in this application is the increase in separation between the front and rear channels when the signal is steered in the front left or rear right direction. The V1.11 decoder with the matrix element of US Pat. No. 4,862,502 is for the front channel under these conditions. These matrix elements do not completely remove the rear steered signal even if steered in the intermediate way between the fully rear position-rear left and rear right. The front left or right output had less 9dB output than the corresponding rear output when the steering went to rear left or rear right (not full rear). In the present invention, the front matrix element is modified to remove sound from the front when the steering is somewhere between the rear left and rear right.

7. Improvement of rear matrix element

Improvements in the rear matrix element are not immediately apparent for typical listeners. These improvements compensate for the continuous varying error of the matrix element across the boundaries between the quadrants. It also improves the intensity balance between the steered and unsteered signals under various conditions. A mathematical description of the matrix device will be given after including these improvements.

8. Detailed description of active matrix device

Matlab language

The mathematics used to describe the matrix element is not based on the continuous function of the variables cs and lr. In general, there are absolute conditions and other nonlinear corrections to the equation. For this reason, a matrix element using a programming language will be described. Matlab language provides a simple way to examine graphical expressions. Matlab is very similar to Fortran or C language. The main difference is that the variables in Matlab are vectors, and each variable can represent an array number in the sequence. For example, in the following, the variable x is

x = 1: 10;

In the present specification, the mat wrap generates a ten-digit bunch having a value of nine. The variables x all contain values of 10. It is described as having 1 to 10 matrix vectors. Each number in each vector can be accessed or manipulated. For example, an expression

x (4) = 4;

Will set the fourth number of the vector x to the value 4. The variable can also represent two magnitude matrices. Each device in the matrix can be assigned in a similar manner.

X (2.3) = 10;

A value of 10 will assign the second row and third column of matrix X.

The detailed description of the matrix element is almost identical to the description disclosed in Reference [2] below. The literature is slightly improved. The main difference is

1. Reference [2] includes the "tv matrix" feature. This feature reduces the level of rear output when steering is forward or neutral. In this application, this feature is achieved through a frequency independent circuit. Thus, we have omitted the "tv matrix" correction.

2. The central matrix element portion has been modified to include a strategy that limits the action of the "movie" strategy, the "music" strategy, and the "music" setting. Reference [2] only describes the "music" setting without a movie.

9. Matrix Decoder in Expressions and Schematic Representations

In reference [1], the present inventors have shown a matrix decoder that can be described in terms of n x 2 matrix elements, where n is the number of output channels. Each output can be viewed as a linear combination of two inputs. The effect of linear combination is given by the elements in the matrix. In this specification, elements are identified by simple combinations of letters. Reference [1] describes 5-channel and 7-channel decode. The conversion from 5-channel to 7-channel is performed independently of the frequency of the decoder, and only the 5-channel decoder is described here.

Only six element-center elements, two front left elements and two rear left elements are described. The right element can be found from the left side by simply changing the same left and right sides. These devices

CL: Matrix element for left input channel to center output

CR: Matrix element for the right input channel to the center output

LFL: Left input channel to front left output

LFR: Right input channel to front left output

LRL: Left input channel to rear left output

LRR: Right input channel to rear left output

These devices are not constant. Their value varies according to two magnitude functions in the direction of appearance of the input sound. The phase / amplitude decoder determines the direction of appearance of the input by comparing the ratio of the amplitudes of the input signals. For example, the degree of steering in the right / left direction is determined from the ratio of the left input channel amplitude to the right input channel amplitude. In a similar manner, the degree of steering in the forward / backward direction is determined from the ratio of the sum of the amplitudes of the input channels to the difference. We do not discuss methods for determining these steering directions here, but the logic 7 decoder is significantly different from the standard decoder. We assume that the steering direction is determined. We will represent these directions in degrees with respect to the angle-left / right direction (lr) and with respect to the front / rear (center / surround) direction (cs). The two steering directions are variable. If both lr and cs are zero, the input signal is unsteered, i.e., the two input channels are uncorrelated.

When an input consisting of a single signal is encoded directly, the two steering directions have their maximums. However, under these conditions they are not independent. The advantage of representing the steering value as an angle is only when it is a single signal and the absolute value of the two steering values should be 45 ° in total. When the input contains any uncorrected components along a strongly steered signal, the sum of the absolute values of the steering values should be less than 45 °.

lr-cs <45

If the value of the matrix element is greater than or equal to the two-dimensional plane formed by the steering value, the center of the plane will have a value (0.0) and the logical value for the sum of the steering values will not exceed 45. In particular, there is a possibility that the total exceeds 45 due to the behavior of the nonlinear filter. US patent application Ser. No. 08 / 742,460 claims a circuit for restricting the lease of lr or cs so that their sum does not exceed 45 °. This claim will not be described herein any further. We will assume that the mathematical behavior of the matrix element is well done during the overrun. The matrix device was graphed with a random number of zeros when the OR of the input variables was exceeded. This makes it possible to directly observe the behavior of the device along the trajectory produced by the boundary trajectory-strongly steered signal. The graph is made of matlab. In Matlab language, the unsteered position is (46.46) because Matlab requires an angle variable that is larger than the actual angle value. This will not cause excessive confusion.

Previous designs for matrix decoders tend to consider only the behavior of the matrix into the strongly steered signal, i.e. the operation of the matrix element surrounding the surface boundary. This is a fundamental error in opinion. In fact, when studying the signal, we find that the boundary between the film or the music is very rarely found. Most of the signals sway around the mid-center of the plane slightly ahead. The behavior of the matrix under these conditions is very important for the sound. When comparing the device of the present invention with the previous device, it can be seen that the complexity of the surface in the intermediate region is increased. This complexity is a response for improvement in sound.

This complexity has a price. Our first 1987 design—see 1980 patent—was simple with a facility with analog components. The new device is designed to be almost completely described as a one-dimensional lookup table that is moved to a digital installation. It is possible to design analog versions with similar performance.

In the present application, we contrast many different versions of the matrix device. Our fastest device is the 1989 patent. These devices were used in the first surround processor and are identical to those of standard (Dolby) surround processors in the left, center and right channels (surround channels). In our design, the surround channel is symmetrically processed to the center channel. In a standard (Dolby) decoder, the surround channel is handled differently and will be described later in this application.

The device shown here is not always a calibrated scale. In general, they are present so that the unsteered value of the non-zero matrix element for a given channel is one. In particular, the elements are scaled such that the maximum value of each element is one or less. In some cases, the scale of the device in the final product is further changed by the calibration process. It is assumed that the matrix elements shown here are constantly scaled.

10. Front left matrix element in 1989 patent

It is assumed that cs and lr are steering directions in the center surround and left and right axes, respectively.

In the 1989 patent, the equation for the front matrix device is given by

About the front left quadrant

LFL = 1-0.5 * G (cs) -0.41 * G (lr)

LFR = -0.5 * G (cs)

About the front right quadrant

LFL = 1-0.5 * G (cs)

LFR = -0.5 * G (cs)

For rear left quadrant (cs is negative)

LFL = 1-0.5 * G (cs) + 0.41 * G (lr)

LFR = -0.5 * G (cs)

About the rear right quadrant

LFL = 1-0.5 * G (cs)

LFR = -0.5 * G (cs)

The function G (x) was determined experimentally in the 1989 patent and mathematically specified in the 1991 patent. As x, a variable of 0 to 1 varies from 0 to 45 degrees. G (x) is 1- | r | / | l | where steering is forward left quadrant (both lr and cs are both positive) r | and | l | may be expressed in the same manner as the left and right input amplitudes). In addition, G (x) may be described in the form of steering angle using a variable equation. One of these is given in the 1991 patent, the other is described later in this specification. 5 and 6 are graphs of three-dimensional illustrated LFL and LFR matrix devices for lr and cs.

In Reference [1], these elements were improved by adding a requirement that the sound intensity of the unsteered components be constant regardless of the steering direction. Mathematically this means that the square root of the sum of the LFL and LFR matrix elements is constant. In the literature this objective is a view that the steering direction—that is, the sum of the squares of these devices should be changed when steered to the left all the way up to 3 dB. Fig. 7 shows the sum of the squares of these devices, which do not meet the requirements of constant loudness. In FIG. 7, the value is constant at .71 along the axis from unsteer to the right. Unsteer to the left rises to 3 dB with a value of 1, while unsteer to the center or rear drops to 3 dB with a value of 0.5. This part of the graph is hidden by the peak on the left. The rearward level is the same as the level in the central direction.

In US patent application Ser. No. 08 / 742,460 and reference [1], the inventors corrected the amplitude error of FIG. 7 by replacing the function G (x) with a matrix equation with sine and cosine. 8 shows a graph of the sum square of the corrected elements LFL and LFR described by the following equation.

The constant value of .71 corresponds to the right half of the plane, rising towards the left vertex.

About the front left quadrant

LFL = cos (cs) + 0.41 * G (lr)

LFR = -sin (cs)

About the front right quadrant

LFL = cos (cs)

LFR = -sin (cs)

Against the rear left quadrant

LFL = cos (-cs) -0.41 * G (lr)

LFR = sin (-cs)

About the rear right quadrant

LFL = cos (-cs)

LFR = sin (-cs)

11. Improvement of front left matrix element

In March 1996, we made many changes to these matrix devices. The basic function dependence was maintained but an additional rise was added along the cs axis forward. The reason for the boost was to improve performance with stereo music, which was panned forward. The purpose of cutting at the rear was to increase the separation between the front and rear channels when stereo music was fanned backwards.

About the front left quadrant

LFL = cos (cs) -0.41 * G (lr) * boost1 (cs)

LFR = (-sin (cs)) * boost1 (cs)

About the front right quadrant

LFL = (cos (cs)) * boost1 (cs)

LFR = (-sin (cs)) * boost1 (cs)

Against the rear left quadrant

LFL = cos (-cs) + 0.41 * G (lr) / boost (cs)

LFR = (sin (cs)) / boost (cs)

About the rear right quadrant

LFL = (cos (cs)) / boost (cs)

LFR = (sin (cs)) / boost (cs)

The function G (x) is identical to one in the 1989 patent. When expressed in degrees as an input, it can be expressed in the same manner as the following expression.

G (x) = 1-tan (45-x)

The function boost1 (cs), like the one used in March 1997, was a total 3dB linear rise applied over the first 22.5 ° steering. The boost (cs) given by corr (x) in the Matlab code is shown below (common lines are indicated by the sign%).

Rise function calculation of -3 dB at% 22.5 °

% corr (x) rises to 3 dB and stops. corr1 (x) rises and then falls again

for x = 1:24; % x has a value from 1 to 24 representing 0 to 23 °.

corr (x) = 10 ＾ (3 * (x-1) / (23 * 20);% up to 3 dB over this range

corr1 (x) = corr (x);

end

for x = 25:46; % corr1 descends again over a range of 24 to 45 °

corr (x) = 1:41;

corr1 (x) = corr (48-x);

end

9 shows the results obtained from the above equation for the illustration of the LFL. The rise in steering moving towards the center is applied to both sides along the left with the lr = 0 axis and the center boundary. In addition, the level reduction during steering moves backwards.

Circuit performance in March 1997 may be improved. The first problem is steering behavior along the boundary between left and center and between right and center. When panning a single strong signal from the left to the center, the value of the LFL matrix element can increase to a maximum of half of the left and center yarns as shown in FIG. This increase in value is an unintended consequence of a deliberate rise in value for the left and right main outputs as the center signal is added to stereo music.

When the stereo signal is panned forward, the front left and right outputs should be at levels that cancel out for removal by the matrix of corrected components from these outputs. However, the method used to increase the level under these conditions should only occur when the lr component of the input is minimal, ie there is no left or right steering. The method of selecting this incremental facility in March 1997 is independent of the lr value, resulting in an increase in level when a strong signal is panned across the boundary.

Rise is only needed along the lr = 0 axis, when lr is zero, the matrix element must not rise. This problem can be solved by using an additional form of matrix element instead of multiplexing. We define a new steering index of defined cs values with the following Matlab code.

Assume lr and cs> 0-front left quadrant

(cs and lr follow matlab transforms varying from 1 to 46)

% bounded c / s search

if (cs <24)

bcs = cs- (lr-1);

if (bcs <1)% This limits the maximum

bcs = 1;

end

else

bcs = 46-cs- (lr-1)

if (bcs <1)

bcs = 11

end

end

If cs <22.5 and lr = 0 (matlab switchover cs <24 and lr = 1), bcs is equal to cs. However, when lr increases, bcs will decrease to zero. If cs> 22.5, bcs also decreases as lr increases.

It is necessary to find the correction function. We find the difference between lr = 0 and the raised matrix element and the unrised one. We call this difference cos _- tbl _- plus and sin _- tbl _- plus. Use matlab code.

a = 0: 45; % Is a vector only in 1 ° steps. α has a value of 0 to 45 °.

a1 is

The% defines the rising table as well as the sine and cosine tables.

sin _- tbl = sin (a1);

cos _- tbl = cos (a1);

cos _- tbl_plus = cos (a1) * corr1 (a + 1);

cos _- tbl _- plus = cos _- tbl _- plus-cos _- tbl:% This is what we use

cos _- tbl _- minus = cos (a1) / corr (a + 1);

sin _- tbl _- plus = sin (a1) * corr1 (a + 1)

sin _- tbl _- plus = sin _- tbl _- plus-sin _- tbl:% This is what we use

sin _- tbl _- minus = sin (a1) * corr (a + 1);

The vectors sin _- tbl _- plus and cos _- tbl _- plus are the difference between sine and cosine, which are elevated sine and cosine.

LFL = cos (cs) -0.41 * G (lr) + cos _- tbl _- plus (bcs)

LFR = -sin (cs) c-sin _- tbl _- plus (bcs)

LFL and LFR in the anterior right quadrant are similar but without the + 0.41 * G (lr) form. These new limitations lead to the matrix device shown in FIG.

In FIG. 10, the new device has a correction amplitude along the center at the right border as well as the left at the center border.

Steering in the rear quadrant is not optimal. The matrix element is given as follows when the steering is rearward.

LFL = cos _- tbl _- minus (-cs) + 0.41 * G (-cs)

LFR = sin _- tbl _- mimus (-cs)

This matrix device is almost identical to the device in the '89 patent. Consider the case when a strong signal is panned from left to back. The '89 device is designed so that the cancellation of the output from the front left output is complete only when this signal is sufficient at the rear (cs = -15, lr = 0). However, in a Logic 7 decoder, it is desirable for the output at the left front output to be zero when the encoded signal reaches the left back direction (cs = -22.5 and lr = 22.5). If the signal is panned to be more sufficient at the rear, the left front output can remain zero. In the matrix device used in March 1997, the output of the front left channel is about -9 dB when the signal is panned in the left rear position. This level difference is sufficient to achieve good performance of the matrix, but in fact is not good.

This performance can be improved by replacing the LFL and LFR matrix devices in the left rear quadrant. Note how the matrix element is changed along the boundary between left and rear. The mathematical method given in Ref. [1] is used to find the behavior of devices along boundaries. Assume that the amplitude of the left front output decreases with the function F (t) when t changes from 0 (left) to -22.5 ° (left rear). The method is given in matrix elements.

LFL = cos (t) * F (t)-/ + sin (t) * (sqrt (1-F (t) ^ 2))

LFR = (sin (t) * F (t) +/- cos (t) * (sqrt (1-F (t) ^ 2))

If F (t) = cos (4 * t) and the correction signal are selected, this is simplified as follows.

LFL = cos (t) * cos (4 * t) + sin (t) * sin (4 * t)

LFR = (sin9t) * cos (4 * t) -cos (t) * sin (4 * t)

In FIG. 11 a plot of the coefficients LFL (straight line curve) and LFR (dashed curve) versus t is shown. (Some defects in the middle are due to the fact that all angles are integers at 22.5 °.)

These devices work well if the front left output is gently reduced from 0 to 22.5 °. The output should remain zero when steered continuously from 22.5 ° to 45 °. Along this part of the boundary, it becomes as follows.

LFL = -sin (t)

LFR = cos (t)

Note that the matrix element is at a considerable distance from the matrix element along the lr = 0 boundary. In Ref. [1], the values are as follows.

LFL = cos (cs)

LFR = sin (cs)

Such matrix elements are designed to work properly with strongly steered signals—where cs and lr have maximum values. The previous matrix element is useful for signals where lr is nearly zero and the stereo signal is panned backwards. There is a need for a method in which the original matrix element is transformed into a new matrix element when lr and cs reach the boundary. Linear insertion method is used. In the processors used in Recicon Products, multiples can be costly and form new better plans—the minimum lr and cs formed by the martlab segment are:

% New boundary parameter search

bp = x;

if (bp> y)

bp = y;

end

And, search for new correction function dependent on bp

for x = 1: 24

ax = 2 * pi * 46 * 360;

front_boundary_tbl (x) = (cos (ax) -sin (ax)) / (cos (ax) -sin (ax));

end

for x = 25:46

ax = 2 * pi * (x-1) * 360;

front_boundary_tbl (x) = (cos (ax) -sin (ax)) / (cos (ax) + sin (ax));

end

In this quadrant, then LFL and LFR are defined as follows.

LFL = cos (cs) / (cos (cs) + sin (cs) -front_boundary_tbl (bp) + 0.41 * G (lr)

LFR = sin (cs). (Cos (cs) + sin (cs))-front_boundary_tbl (bp)

Note the modification to cos (cs) + sin (cs). Dividing cos (cs) by this device yields the same function 1-0.5 * G (cs) as the Dolby matrix in this quadrant. Dividing sin (cs) by this element yields the initial function -0.5 * G (cs).

Similarly obtained in the right rear quadrant.

LFL = cos (cs) / (cos (cs) + sin (cs)) = 1-0.5 * G (cs)

LFR = sin (cs) / (cos (cs) + sin (cs)) = 0.5 * G (cs)

Schematic representations of these values are shown in FIGS. 12 and 13. Note the large correction along the left rear boundary. This causes the front left output to be zero when the steering moves from left to left rear. The output remains at zero when steering is gradually peaking backwards. The function in the right rear quadrant along the lr = 0 axis is equivalent to the Dolby matrix.

In Fig. 13, notice the large peak on the left side of the rear boundary. This works in conjunction with the LFL matrix element, which maintains the front output at zero along this boundary as the steering moves from left rear to maximum rear. In other words, in the rearward direction and rear left quadrant according to lr = 0, the device is identical to the Dolby matrix.

One of the main goals of the design of the Logic 7 matrix is to make the loudness output from the non-steered components at the input of the decoder constant regardless of the direction of the steered signals present at the same time. As described above, this means requires that the square of the sum of the matrix elements for each output be 1 regardless of the steering direction. As explained above, this requirement must be changed when steered strongly in the output direction. In other words, if looking from the left front output, the square of the sum of the matrix elements should be increased to 3 dB when the steering is shifted to the maximum left.

However, the square of the sum of the matrix elements can still be tested for success by plotting the route. See FIG. 14 and FIG. 15, a design modified with this plot. In Fig. 14, note the 3 dB peak in the left direction and some small peaks shifted to 22.5 ° from where the signal is not steered in the center direction (1 / (sin (cs) + cos (cs) in the rear quadrant for this plot). If you remove)), you can see that the derived sum is exactly one)

This peak is the result of deliberate boost of the left and right outputs during half-forward steering. Note that in this design, the rms sum is very large in another quadrant. The value in the rear left quadrant is not the same as the method used for the device, but the match is very good.

In FIG. 15, it has one value which is not steered on the left side. The center vertex has a value of 0.71, the back vertex has a value of 0.5 and the left vertex has a value of 1.41. Note the peak in the middle of the central axis.

12. Rear Matrix Element During Forward Steering

The back matrix element in the 89 patent (except for introducing a reference according to 0.71 here to show the effectiveness of the standard calibration procedure) is given as follows.

About the front left quadrant

LRL = 0.71 * (1-G (lr))

LRR = 0.71 * -1

Against the rear left quadrant

LRL = 0.71 * (1-G (lr) + 0.41 * G (-cs))

LRR = -0.71 * (1 + 0.41 * G (-cs))

(The right half of the plane is the same, but replaced by LRL and LRR.)

The rear matrix elements in Dolby Pro Logic are as follows.

About the front left quadrant

LRL = 1-G (lr)

LRR = -1

Against the rear left quadrant

LRL = 1-G (lr)

LRR = -1

(The right half of the plane is the same, but LRL and LRR are swapped.)

The Dolby element and the '89 element are modified to be identical in the rear left quadrant when cs = -45 °.

13. A Brief Description of Surround Levels in Dolby Pro Logic

The Dolby element is similar to the '89 patent element except that there is no independent boost on the rear cs. This difference becomes very important when the device has different values for the unsteered signal after the standard modification procedure. In general, the matrix device of the present invention does not consider the modification procedure for such a decoder. All matrix elements with relatively arbitrary scaling can be obtained. In the best case, the device is present at a maximum value of 1.41. In fact, for technical reasons, the matrix elements are all equally equal and have a maximum value less than 1.41. In addition, when the decoder is last used, the gain of each output in the loudspeaker is adjusted. To adjust this, signals encoded in four directions of left, center, right and surround with the same sound intensity are played, and the gain of each output is adjusted until the sound intensity is the same at the listening position. In fact, this means that the actual values of the matrix elements are referenced so that the four outputs of the decoder are identical in full steering. This modification is included in the back element equation above.

With the steered forward or not steered, the 3dB difference of the device is very large. In the non-steered state the device in the '89 patent has a value of 0.71 and the square of the value of the device has a value of one. This is not the actual value of the Dolby rear element when modified. LRL has a non-steered value of 1 and the sum of squares is 2 or 3 dB higher than the '89 output. Note that the modification procedure in the matrix does not correspond to a "Dolby Surround" passive matrix when the matrix is unsteered. Note that the Dolby Surround passive matrix has a rear output of 0.71 * (A _in -B _in ) and the prologic matrix does not have this value. As a result, if the A and B inputs are not related to each other, the rear output is 3dB stronger than the other. If the two speakers are separated at the rear, each speaker is adjusted 3 dB more slowly than one rear speaker, which ensures that all five speakers have the same sound intensity when the decoder inputs are not related to each other. If the matrix element in the '89 patent is used and the decoder inputs are not related to each other, the same modification procedure is 3 dB smaller than the sound intensity at the rear.

As a matter of taste, the problem is how the loudness exists in the rear channel when the inputs are not related to each other. When a surround encoded recording is played, the producer wants to play when the mixed recording is balanced. To achieve this balance, the decoder and encoder must be designed to combine. Standard stereo components, however, aim to reproduce the intensity balance in the first recording, while producing a surround that suits your taste. The problem with the Dolby Matrix element is that in conventional two-channel recording the intensity balance is not preserved through the matrix. The surround channel is too strong and the center channel is too weak.

Let's look at the problem that occurs when you have inputs in a decoder that consists of left and right components that are not related to each other and a central component that is separate and not related to each other.

A _in = L _in + 0.71 * C _in

B _in = R _in + 0.71 * C _in

If A _in and B _in are played through a conventional stereo system, the sound intensity in the room is proportional to L _in ² + R _in ² + C _in ² . If all three components have the same amplitude, the intensity ratio of the left and right components plus the central component is 1: 2.

The decoder reproduces the sound intensity indoors with almost the same intensity ratio, such as stereo, regardless of the intensity ratio of C _in , L _in, and R _in . This can be expressed mathematically. Essentially the equivalent condition of the intensity ratio can be described as a function of the central matrix element along the cs axis. If all other matrix elements are given, the Dolby Matrix element can cause the rear sound intensity to be modified to be 3 dB smaller than the other three outputs when the matrix is fully steered. In other words, 3dB is smaller than the standard crystal and then the central matrix element may have the shape shown in FIG. Standard modifications can be made identical and the results are shown in FIG. 17.

In FIG. 16, assuming that the intensity ratio at the decoder output is the same as the intensity ratio at stereo, it is modified to be 3 dB smaller than the level normally used. The actual value resulting from the unsteered signal and the fully steered signal is less than 1.5 dB in the middle.

In FIG. 17, assuming that the intensity ratio of the stereo given to the matrix element and the correction used in Dolby Pro Logic (dotted curve) are the same, the actual value is 3 dB or more so that all values of cs are lowered.

These two shapes are shown to mix something, which shows that on a Dolby Pro Logic system, the ready mix for playback is louder in the center than the ready mix for playback in stereo. Conversely, a prepared mix for stereo loses timbre clarity when played on a prologic decoder. In short, this is not a manual Dolby Surround decoder. This problem will be mentioned once again when the central matrix element is mentioned.

14. Generate two independent rear outputs

The main problem with both Dolby and '89 devices was that there was only a single rear output. The '91 US patent discloses a method for generating two independent lateral outputs, the equations in which are embodied in the FL quadrant in US patent application Ser. No. 08 / 742,460 and reference [1], 1996. The purpose of the device in this quadrant was to eliminate the output of the steered signal from left to center while maintaining some output from the LR channel for non-steer components present at the same time. In order to achieve this object, it is assumed that the LRL matrix element has the following form.

In the LF quadrant,

LRL = 1-GS (lr) -0.5 * G (cs)

LRR = -0.5 * G (cs) -G (lr)

As can be seen, these matrix elements are very similar to the '89 elements, but additionally have a G (lr) term in the LRR and a GS term in the LRL. In order to provide some unsteered signal strength as the steered signal is removed, G (lr) has been included to add the signal from the B input channel of the decoder to the LR output. The function GS (lr) has been described using the reference that there should be no signal output with a fully steered signal moving from left to center. Although the equation for GS (lr) has been shown to be the same as G ² (lr), a more complex representation of this equation is provided in the '91 US patent. The two representations can be seen as identical.

In reference [1], these devices are modified to provide a boost of (sin (cs) + cos (cs)) to bring them closer to a constant loudness for the non-steered components. Although perfectly successful in the RF quadrant, modifications are not very successful in the LF quadrant. See Figure 18. (The matrix elements in the RF quadrant are the same as the LRL and LRR elements in the '89 US patent.)

In Fig. 18, it should be noted that there is a 3 ms dip along the line from the middle to the left vertex in the FL quadrant, and there is almost a 3 ms boost at the level along the boundary between the left and center. The mountain range in the rear quadrant will be described later. In this figure, the “tv matrix” correction, V1.11, has been removed for better comparison with the present invention in FIG. 20.

18 illustrates various problems with sound strength. First, consider the dip in the sum of squares along the cs = 0 axis. Dip exists because the functional form of G (lr) in LRR is not optimal. The choice of G (lr) is arbitrary—in previous designs, such a function had already existed in the decoder, and its implementation in analog circuits is easy.

Ideally, we choose GS (lr) and GR (lr) as a way to keep the sum of the squares of LRL and LRR constant along the cs = 0 axis with the function GR (lr) in the equation, We want to keep the output at 0 along the boundaries between. This can be done. It is intended that the matrix element in the RF quadrant and the matrix element be the same along the lr = 0 axis. therefore,

LRL = cos (cs) -GS (lr)

LRR = -sin (cs) -GR (lr)

Assume

We want the sum of squares along cs = 0 to be equal to one.

(1-GS (lr)) ² + (GR (lr)) ² = 1

And for the steered signal, we want the output to be zero, or t changes from 0 to 45 degrees,

LRL * cos (t) + LRR * sin (t) = 0

The equation produces a scattering quadratic equation for GR and GS which is numerically explained and graphed in FIG. The use of GS and GR as shown greatly improves the sum of the intensities along the cs = 0 axis as intended. However, the peak is maintained at the sum of squares along the boundary between the left side and the center.

In a practical design, it may not be important to compensate for these errors, but we will do so using the following strategy. We will divide both matrix elements into arguments dependent on the newly combined variable based on cs and lr. The new variable is called xymin. (You don't actually use division, but multiply by the inverse of the arguments described below.) In Matlab notation,

Find the minimum of% x or y

xymin = x;

if (xymin> y)

xymin = y;

end

if (xymin〉 23)

xymin = 23;

end

% xymin varies from 0 to 22.5 degrees

Use xymin to find modifications to the matrix device along the boundary.

In the anterior left quadrant,

LRL = (cos (cs)-GS (lr)) / (1 + .29 * sin (4 * xymin))

LRR = (-sin (cs)-GR (lr)) / (1 + .29 * sin (4 * xymin))

In the front right quadrant

LRL = cos (cs)

LRR = -sin (cs)

In reference [2], these elements are additionally multiplied with the "tv matrix" correction. In this application, FIG. 20 shows a matrix element without "tv matrix" correction. In this application, this modification is handled by a frequency dependent circuit that follows the matrix.

In FIG. 20, the sum of squares is close to 1 and is continuous except for a gentle rise of the level at the rear.

15. Rear Matrix Elements During Back Steering

The back matrix element provided in the '91 US patent was inadequate for a five channel decoder and was modified in Applicants' CP-3 product. Reference [1] and US patent application Ser. No. 08 / 742,460 provide a mathematical method for deriving these devices along the boundaries of the LR quadrant. The method worked along the boundary but produced discontinuities along the lr = 0 axis and the cs = 0 axis. In March 1997, these discontinuities were (nearly) compensated for by further modifications to the matrix device, maintaining their behavior along the steering boundary.

In the devices disclosed in this application, these errors were corrected by interpolation. First-order interpolation repairs discontinuities along the cs = 0 boundary for LRL. The interpolation method ensures that the value corresponds to the value of GS (lr) when cs = 0, and allows the value to rise smoothly to the value given by the equation as cs increases negatively toward the rear. Quadratic interpolation causes LRR to be interpolated to the value of GR (lr) along the cs = 0 axis.

16. Left side rear output during rear steer from right to RR

Consider first the LR and LRR matrix elements when the steering is neutral or at any location between the full right and RR. That is, lr can vary from 0 to -45 degrees and cs can vary from 0 to -22.5 degrees.

Under these conditions, the steered component of the input must be removed from the left output-there must be no output from the RL channel if there is steering towards the right or RR.

The matrix element provided in the '91 US patent achieves this purpose. They are essentially the same as the rear matrix elements in a four-channel decoder with the addition of sin (cs) + cos (cs) modifications for non-steer loudness. When this is done, the matrix element becomes simple sine and cosine:

LRL = cos (-cs) = sri (-cs)

LRR = sin (-cs) = sric (-cs)

A new function of sric (x) equal to sin (x) and sri (x) equal to cos (x) was defined in the range of 0 to 22.5 degrees. We will use these functions again when defining LR matrix elements during left steering.

17. Left / Left Rear Output During Rear Steering from RR to Right

Now consider the same matrix device as cs is above -22.5 degrees. As disclosed in Reference [1] and in two patent applications, the LRL must rise to at least 1 in this range and the LRR must decrease to zero. Simple functions satisfy this. (Remember that cs becomes negative in this equation and varies from -22.5 to -45 degrees):

LRL = (cos (45 + cs) + rboost (-cs)) = (sri (-cs) + rboost (-cs))

LRR = sin (45 + cs) = sric (-cs)

Rboost (cs) is defined in reference [1] and US patent application Ser. No. 08 / 742,460. It is almost equivalent to the function 0.41 * G (cs) in the matrix element above, except that rboost (cs) goes from 0> cs> -22.5, and 0 as cs changes from -22.5 degrees to -45 degrees. To 0.41. The exact function shape is determined by the need to keep the loudness of the rear output constant while the sound pans from LR to full rear.

The LR matrix device during right steering is now perfect.

18. LR element during steering from left to left rear

The operation of the LRL and LRR devices is much more complicated. The LRL device needs to rise quickly from zero to near maximum, with lr decreasing from 45 to 22.5 or zero. The matrix element given in reference [1] must do this, but there is a problem with continuity at the cs = 0 boundary as described above.

In March 1997, I published an answer that turned out to be a function of one variable and several conditional statements. In reference [1], a problem occurred at the cs = 0 boundary because the LRL matrix element in front of the boundary (cs ≧ 0) is given by GS (lr). The function given by reference [1] in the rear face (cs <0) has the same end point, but is different if lr is nonzero or 45 degrees.

The mathematical method in reference [1] provides the following equation for the LR matrix element in the range 22.5 &lt; lr &lt; 45. (Remember that t = 45-lr when rewriting these equations from reference [1].)

LRL = cos (45-lr) * sin (4 * (45-lr))-sin (45-lr) * cos (4 * (45-lr)) = sra (lr)

LRR =-(sin (45-lr) * sin (4 * (45-lr)) + cos (45-lr) * cos (4 * (45-lr))) =-srac (lr)

In this range, new functions sra (lr) and srac (lr) are defined.

If cs ≧ 22.5, lr can vary from 0 to 45. Reference [1] defines LRL and LRR (when lr has a range of 0 &lt; lr &lt; 22.5) (see FIG. 6 in Reference [1]).

LRL = cos (lr) = sra (lr)

LRR = -sin (lr) = -srac (lr)

These two functions sra (x) and srac (x) are now defined at 0 <lr <45.

19. March 1997 version

The March 1997 version uses interpolation to modify the LRR according to the boundary. There are two discontinuities herein. Along the cs = 0 boundary, the LRR at the rear must match the LRR for the forward direction, which represents LRR = -G (lr) along the cs = 0 boundary.

The choice used in March 1997 is to use an interpolation method based on the value of cs in the range of 0 to 15 degrees but with some computational emphasis. In other words, when cs is 0, G (lr) is used to find LRR. Interpolate the value of srac (lr) with cs increasing to 15 degrees.

There may be a discontinuity along the lr = 0 axis. In March 1997, this discontinuity was (slightly) corrected by adding a term to the LRR, found by using the new variable cs_bounded. The modification term is simply sric (cs_bounded). This term will ensure continuity on the lr = 0 axis.

First, cs_bounded is defined by the following Matlab notation.

cs_bounded = lr-cs;

if (cs_bounded <1)% this limits the maximum

cs_bounded = 0;

end

if (45- | lr | 〈cs_bounded)% Use the smaller of the two values

cs_bounded = 45-lr;

end

for cs = 0 to 15

LRR = (-(srac (lr) + (srac (lr) -G (lr)) * (15-cs) / 15) + sric (cs_bounded));

for cs = 15 to 22.5

LRR = (-srac (lr) ÷ sric (cs_bounded));

20. LRL implemented with Logic 7 as in '97 August

In the present invention, LRL is calculated by the same interpolation method as in LRR.

for cs = 0 to 15

LRL = ((sra (lr) + (sra (lr) -GS (lr)) * (15-cs) / 15) + sri (-cs));

for cs = 15 to 22.5

LRL = (sra (lr) + sri (-cs));

21.Rear output during steering from LR to full rear

As the steering moves completely back in the LR, the device adds a modification to the rear loudness, following what is provided in reference [1].

In Matlab notation,

For cs〉 22.5, lr 〈22.5

LRL = (sra (lr) + sri (cs) ÷ rboost (cs))

LRR = -srac (lr)-sric (cs_bounded)

This completes the LRL and LRR matrix elements during left steering. The value for right steering can be found by swapping left and right in the above definition.

22. Central Matrix Device

Both the '89 US patent and Dolby Pro Logic have the following matrix elements.

About forward steering

CL = 1-G (lr) + 0.41 * G (cs)

CR = 1 + 0.41 * G (cs)

About rear steering

CL = 1-G (lr)

CR = 1

Since the matrix element has symmetry with respect to the left-right axis, the values of CL and CR for right steering can be found by exchanging CL and CR. See FIG. 21 for a schematic representation of such a device.

In Figure 21, the middle and right and back vertices of the graph have a value of 1. The central vertex has a value of 1.41. In practice, these devices are referenced so that the maximum value is one.

In the application of US patent application Ser. No. 08 / 742,460 and reference [1], these devices are replaced with sine and cosine.

About forward steering

CL = cos (45-lr) * sin (2 * (45-lr))-sin (45-lr) * cos (2 * (45-lr)) + 0.41 * G (cs)

CR = sin (45-lr) * sin (2 * (45-lr)) + cos (45-lr) * cos (2 * (45-lr)) + 0.41 * G (cs)

These equations were never implemented. In March 1997, the product used the device in the '89 US patent, but used other standardization and other boost functions than G (cs). It was found that it was important to reduce the non-steer level of the center output, and a value of 4.5 dB below the prologic level was chosen. In the March 1997 version, the boost function was selected by listening tests.

In the March 1997 version, cs's boost function starts at zero as before, and rises with cs in such a way that CL and CR increase as cs moves from 0 to 22.5 degrees. The increase is a constant value of k for each k of increase in cs. At the next 20 degrees, the boost function changes the slope so that the matrix element rises and remains constant by another 3 ms. Thus, if the steering is "half front" (8 ms or 23 degrees), the new matrix element is equal to the median of the previous matrix element. As the steering moves forward, the new matrix element and the old matrix element become identical. Thus, the output of the center channel is 4.5㏈ smaller than the previous output if the steering is neutral, but rises to the previous value if the steering is completely centered. See FIG. 22 for a three-dimensional plot of this device.

In Figure 22, it should be noted that the median and the right rear vertex have been reduced by 4.5 ms. As cs increases, the center rises to a value of 1.41 on two slopes.

We found that the central element used in March 1997 was not optimal. Indeed, significant experiments with decoders have shown that switching between stereo (two-channel) playback and playback through the matrix may tend to lose the central location of popular music recordings and dialogue in some movies. Also, as the center channel level changes, listeners who are not equidistant from the front speakers can recognize the exact location of the center voice movement. This problem has been extensively analyzed when deploying the new central matrix device presented herein. As will be described later, there is also a problem when the signal pans from left to center or right to center along the boundary. The matrix element of US patent application Ser. No. 08 / 742,460 provides very low output from the center speaker when the fan is in between.

23. Central channel of the new design

While it is possible to remove strongly steered signals from the center channel output using the matrix method, at any time there is steering in front but not biased left or right, the center channel is the sum of the A and B inputs with some gain factor. Should play. In other words, it is impossible to remove the uncorrelated left and right components from the central channel. The only option is to adjust the loudness of the center speaker. How big should it be?

This question depends on the action of the left and right main outputs. The matrix values provided in the LFL and LFR are designed to remove the center component of the input signal as the steering moves forward. If the input signal is encoded to appear from the front by using a cross mixer, such as stereo width control, the matrix element ('89 element, 1996 AES element, March 1997 element, and the document first described above) The devices provided) all completely restore the original separation.

However, if the input to the decoder consists of uncorrelated left and right channels with the addition of the uncorrelated central channel, i.e.

A _in = L _in + 0.71 * C _in

B _in = R _in + 0.71 * C _in

As the level of C _in is increased _in relation to L _in and R _in , the C components of the L and R front outputs of the decoder are not completely removed unless C _in is large compared to L _in and R _in . In general, there is some C _in remaining in the L and R forward outputs. What will the listeners hear?

There are two ways to calculate what the listener hears. If the listener is exactly equidistant from the left, right, and center speakers, the listeners will hear the sum of the sound pressures from each speaker. This is equivalent to summing three forward outputs. Under these conditions, it is easy to indicate that any reduction in the center component of the left and right speakers produces an absolute loss of sound pressure from the center component regardless of the amplitude of the center speaker. This is because the center speaker is always derived from the A and B inputs, and as its amplitude increases, the amplitude of the L _in and R _in signals must increase with the amplitude of the C _in signal.

However, if the listener is not equidistant from each speaker, the listener is very likely to hear the sum of the sound intensities from each speaker equal to the sum of the squares of the three front outputs. In fact, extensive listening has shown that the sum of the intensity of all speakers is really important, so the sum of the squares of all outputs of the decoder, including the rear output, must be considered.

If you want to design the matrix so that the ratio of the amplitudes of L _in , R _in , and C _in is retained when switching between stereo reproduction and matrix reproduction, the sound intensity of the Cin component from the center output is determined by It should rise precisely in proportion to the decrease in sound intensity and its decrease in rear output. Further complicating is that the left and right front outputs have a level boost up to 3 dB above. This will cause the center to be slightly louder to keep the ratio constant. This requirement can be described as a set of equations for sound intensity. These equations can be explained by the gain function required for the center speaker.

A graph showing the energy relationship for the Dolby Pro Logic decoder under various conditions was previewed. The prologic decoder is not optimal. The same can be done in the new decoder of the present application.

FIG. 23 shows the required center gain if the energy of the center component of the input signal is maintained in the three front channels as steering increases toward the front (continuous curve). As can be seen, the necessary increase in the level of the center channel is very steep. The increase is a multiplicity of milliseconds per millimeter of steering value. It also shows the gain in the standard decoder (dashed curve).

As mentioned above, there are two solutions to this problem. First, the "film" solution will be explained. The solution is not entirely mathematical. In fact it was found that the function shown in FIG. 23 rose very steeply. The change in the level of the center channel is very obvious. We decided to slow down the counting demands weakly to about a centimeter smaller than the ideal. If the median is recalculated, the result shown by the straight line in FIG. 24 is obtained. Indeed, a linear rise can be substituted for the beginning of the curve shown by dashed lines in FIG. In fact, these median results were excellent for the film.

Referring to FIG. 24, the curve without actual interruption rises very steeply. Linear slopes given by dashed lines work better.

Music requires a different solution. The central attenuation shown in FIGS. 23 and 24 is derived assuming matrix elements given in advance for LFL and LFR. If so, what other devices do you use? In particular, do you need to be aggressive about removing the central component from the left and right front outputs?

Listening tests indicate that the previous left and right front matrix elements need not be aggressive about removing the center component while playing music. Acoustic, you don't have to. Energy removed from the left and right voltages should be provided to the central expander. If this energy is not removed, energy comes from the left and right front speakers, and the center speaker does not need to be strong. The sound intensity in the room is the same. The key is to put enough energy in the center speaker to produce a convincing front image for listeners outside the axis while minimizing the reduction in stereo width for listeners equidistant from the front left and right speakers.

As in US patent application Ser. No. 08 / 742,460, errors and trials can find the optimal median loudness. And, it can be described for the matrix elements required in the front left and right to maintain the strength of the C _in component _{in the} solution. As before, it is assumed that the center channel is level reduced by 4.5 dB below the -0.75 dB overall attenuation or below the level at Applicants''89 US patent decoder. -7.5 ㏈ is equivalent to 0.42. The matrix element at the center can be multiplied by this factor, and a new center boost function GC can be defined.

About forward steering

CL = 0.42 * (1-G (lr)) + GC (cs)

CR = 0.42 + GC (cs)

About rear steering

CL = 0.42 * (1-G (lr))

CR = 0.42

Several functions have been tried for GC (cs). One function described below may not be ideal, but is considered good enough. It is specified in degrees with respect to the angle (cs) and is obtained over some trial errors.

In MATLAB notation:

center_max = 0.65;

center_rate = 0.75;

center_max2 = 1;

center_rate2 = 0.3;

center_rate 3 = 0.1;

if (cs <12)

gc (cs-1) = 0.42 * 10 ^ (db * center_rate / (20));

tmp = gc (cs ÷ 1);

elseif (cs <30)

gc (cs + 1) = tmp * 10 ^ ((cs-11) * center_rate3 / (20));

if (gc (cs-1)> center_max)

gc (cs + 1) = center_max;

end

else

gc (cs + 1) = center_max * 10 ^ ((cs-29) * center_rate2 / (20));

if (gc (cs ÷ 1)〉 center_max2)

gc (cs + 1) = center_max2;

end

end

The function (0.42 + GC (cs)) is envisioned in FIG. Note that it quickly rises to a value of 0.42 (4.5 dB lower than Dolby Surround), then rises slowly, and finally rises steeply to value 1.

Assuming functions for LFL, LRL, and LRR, the functions needed for LFR can be described. The ratio of the C _in component to the left and right outputs should be explained. These matrix elements must also provide some boost of the L _in and R _in components and must have a current shape at the left-side center boundary as well as the right-side center boundary.

LFL = GP (cs)

LFR = GF (cs)

CL = 0.42 * (1-G (LR)) + GC (cs)

CR = 0.42 + GC (cs)

Assume

The intensity from front left and right can be calculated as follows:

PLR = (GR ² + GF ² ) * (L _in ² + R _in ² ) + (GP-GF) ² * C _in ²

The intensity from the center

PC = GC ² * (L _in ² + R _in ² ) + 2 * GC ² * C _in ²

The intensity from the back depends on the matrix element used. We will assume that the rear channel is attenuated by 3 dB during forward steering, LRL is cos (cs) and LRR is sin (cs). From a single speaker,

PREAR = (0.71 * (cos (cs) * (L _in + 0.71 * R _in ) -sin (cs) * (R _in + 0.71 * C _in ))) ²

If L _in ² ≒ R _in ² , on two speakers,

PREAR = 0.5 * C _in ² * ((cos (cs) -sin (cs)) ² ) + L _in ²

The total intensity from all three speakers is PLR + PC + PREAR.

PT = (GP ² + GF ² + GC ² ) * (L _in ² + R _in ² ) + ((GP-GF) ² + 2 * GC ² ) * C _in ² + PREAR

C _in intensity for L and R _in the ratio of intensity _in the (L _in ² = R _in ² family)

RATIO = (((GP (cs) -GF (cs)) ² + 2 * (GC (cs) ² ) + 0.5 * (cos (cs) -sin (cs)) ² ))

* C _in ² / ((2 * (GP (cs) ² + GC (cs) ² + GF (cs) ² ) + 1) * L _in ² )

RATIO = (C _in ² / L _in ² ) * ((GP (cs) -GF (cs)) ² + 2 * (GC (cs) ² ) + 0.5 * (cos (cs)

-sin (cs)) ² ) / (2 * (GP (cs) ² + GC (cs) ² + GF (cs) ² ) +1)

In normal stereo, GC = 0, GP = 1, GF = 0. Center LR strength ratio is

RATIO = (C _in ² / L _in ² ) * 0.5

If this ratio is constant regardless of the value of C _in ² / L _in ² for the matrix device of this application,

((GP (cs) -GF (cs)) ² + 2 * (GC (cs) ² ) + 0.5 * (cos (cs) -sin (cs)) ² )

= ((GP (cs) ² + GC (cs) ² + GF (cs) ² ) +0.5)

The equation can be explained numerically. If we assume the GC and GP = LFL as before, the result can be seen in FIG.

In FIG. 26, the continuous curve is the GF graph required for a constant energy ratio with a new “music” median attenuation (GC). The dashed curve is the LFR device, sin (cs) * corr1 from March '97. The dashed curve is an LFR device without sin (cs), the correlation term corr1. Note that GF is close to zero and increases rapidly until cs reaches 30 degrees. In fact, we found that it was best to limit the value at about 33 degrees. In fact, the LFRs derived from these curves have a negative sign.

GF provides the shape of the LFR matrix device along the lr = 0 axis, with cs increasing from 0 to the center. There is a need for a method of mixing this action with the action of the previous LFR element, which must be maintained not only from right to center, but also along the boundary between left and center. If cs≤22.5 degrees, the way to do this is to define a different function between GF and sin (cs). And we limit these functions in several ways. In Matlab notation:

gf_diff = sin (cs) -gf (cs);

for cs = 0: 45;

if (gf_diff (cs)〉 sin (cs))

gf_diff (cs) = sin (cs);

end

if (gf_diff (cs) <0)

gf_diff (cs) = 0);

end

end

% Limited c / s search

if (y <24)

bcs = y- (x-1);

if (bcs <1)% limit maximum

bcs = 1;

end

else

bcs = 47-y- (x-1);

if (bcs <1)%〉 46)

bcs = 1; % 46;

end

end

LFR devices can now be marked with Matlab notation:

% Such proper trick is to perform interpolation to the boundary

% Of course the cost is also split !!!

if (y <23)% This is an easy way to intersect a region.

lfr3d (47-x, 47-y) =-sin_tbl (y) + gf_diff (bcs);

else

tmp = ((47-y-x) / (47-y) * gf_diff (y);

lfr3d (47-x, 47-y) =-sin_tbl (y) + tmp;

end

Note that the sign of gf_diff becomes positive in the above equation. Gf-diff thus cancels the value of sin (cs), reducing the value of the device to zero along the first portion of the lr = 0 axis. See FIG. 27.

In FIG. 27, the value is zero in the middle of the plane (no steering) and remains at 0 with cs increasing to ˜30 degrees along the lr = 0 axis. The value is then lowered to match the previous value along the boundary from left to center and from right to center.

24. Panning Error at Center Output

As you can see, the new central function, if you describe it this way,

CL = 0.42 * (1-G (lr)) + GC (cs)

CR = 0.42 + GC (cs)

lr = 0 acts along the axis, but produces a panning error along the boundary between left and center. The value in reference [1] of 1996 (not implemented at all) gives a flat function of cos (2 * cs) along the left boundary. These values create flat panning between the left and center. Along this boundary, we want the new central function to act similarly.

By adding an additional function of xymin, you can modify the matrix device you are working with (in Matlab notation).

center_fix_tbl = .8 * (corr1-1);

And,

CL = 0.42-0.42 * G (lr) + GC (cs) + center_fix_table (xymin)

CR = 0.42 + GC (cs) + center_fix_table (xymin)

See FIG. 28 for a three-dimensional representation of the CL matrix device. It's not perfect, but it actually works.

In Fig. 28, note the correction for panning along the boundary between the left and center, which is fairly flat.

In FIG. 29, which is a graph of left front (dashed curve) and center (continuous curve) outputs, center steering takes place on the left side of the configuration and full left steering takes place on the right side. In the "music" strategy, when the center is about 6 dB stronger than the left, it is easy to limit the value of cs to about 33 degrees (about 13 in the axis as identified).

25. Technical details of the encoder

There are two purposes of the Logic 7 encoder. First, it must be able to encode a 5.1 channel tape in such a way that the encoded version is decoded by a Logic 7 decoder with minimal subjective change. Second, the encoded output must be stereo compatible. That is, it should sound as close as possible to a manual two-channel mix of the same components. One factor in this stereo compatibility is that when played back on a standard stereo system, the encoder's output must provide the same sense loudness for each sound source in the first 5 channel mix. The exact position of the sound source in stereo should also be as close as possible to the exact position in the first five channels.

In the discussion of the Institute for Broadcast Technique (IRT) in Munich, it became clear that the purpose of stereo compatibility of stereo signals, as described above, cannot be met by a passive encoder. If all channels have the same foreground importance, the five channel recording should be encoded as described above. This encoding requires the surround channels to be mixed at the output of the encoder in such a way as to maintain energy. In other words, the total energy of the encoder outputs must be the same. If all five loudspeakers have the same instrumentation, then this constant energy setting will be required for five-channel music sources and most film sources. Such music sources are not common now, but I think they will be common in the future. Music recording requires a different encoding if a front three-channel foreground instrument is located, which is inherently reflective in the rear channel.

After a series of tests (in IRT and elsewhere), this type of music recording was successfully encoded in stereo-compatible form when the surround channels were mixed at less than 3 dB less intensity than the others. Although this -3 dB level has been adopted as a standard for surround encoding in Europe, the standard specifies that other surround levels can be used for special purposes. The new encoder includes active circuitry to detect strong signals in the surround channel. If such a signal is sometimes present, the encoder uses the full surround level. If the surround input is consistently -6 dB or smaller compared to the front channel, the surround gain is gradually lowered to 3 dB to meet European standards.

These active circuits also existed in the encoder in US patent application Ser. No. 08 / 742,460. However, tests using previous encoders at Munich I's IRT revealed that some sound sources were incorrectly encoded. To solve this problem, a new configuration has been developed. The new encoder is clearly superior in its performance in many difficult components. As a passive encoder, the first encoder was first developed. The new encoder will also operate in passive mode, but originally intended to act as an active encoder. Active circuitry corrects for many small errors inherent in the design. However, it performs better than previous encoders without active modification.

In widespread listening, several other small problems have arisen with the first encoder. Most (but not all) of these issues were raised in the new encoder. For example, if a stereo signal is applied to both the front and rear terminals of the encoder at the same time, the resulting encoder output is biased very far from the front. The new encoder compensates for this effect by slightly increasing the rearward bias. Similarly, it has been found that dialogue can sometimes be lost when the film is encoded with substantial surround content. This problem is greatly improved by the above mentioned changes in intensity balance, but the encoder is also intended to be used with a standard (Dolby) decoder. The new encoder compensates for this effect by slightly increasing the center channel input to the encoder under these conditions.

26. Description of the design

If the attenuation function fcn is equal to 0.71 or -3 dB, the new encoder treats the left, center and right signals the same as the previous design and Dolby encoder.

The surround channel looks more complicated. The functions fc () and fs () direct the surround channel to a path with a phase shift of 90 ° relative to the front channel or to a path without phase shift. In the basic operation of encoder fc, fc is 1 and fs is 0. That is, only paths using a phase shift of 90 ° are activated.

The value of crx is normally 0.38. This controls the amount of negative crossfeed for each surround channel. When only the input to one of the surround channels is present, the A and B outputs have an amplitude ratio of -.38: .91, which has a steering angle of 22.5 ° backwards. In general, the total intensity of the two output channels is one. That is, the sum of the squares of .91 and .38 is 1.

Although the output of the encoder is relatively simple when only one channel is driven, it becomes a problem when both surround inputs are driven simultaneously. Driving the LS and RS inputs with the same signal (shown together on the film), all signals at the summation node are in phase, so the overall level of each output channel is .38 + .91 or 1.29. This output is stronger than the index 1.29 or 2.2 ms. When the two surround channels have similar levels and coincident phases, an active circuit is included in the encoder to reduce the value of the function fc by 2.2 dB.

When the two surround channels have similar levels and out of phase, different errors occur. In this case, two attenuation indices are subtracted, so the A and B outputs have the same amplitude and phase and are at levels of .91-.38 or .58. This error is serious. Previous encoder designs generated unsteer signals in this state, which makes sense. It is not reasonable for the signals applied to the rear input terminals to become center-oriented signals. Thus, an active circuit is provided which increases the value of fs when the two rear channels have similar levels and have opposite phases. The result of mixing the actual path and phase shifted path for the back channel has a 90 ° phase difference between output channels A and B. This result is an unsteer signal, which is what the inventors desire.

As mentioned earlier, the inventors discovered that there was a European standard surround encoder during the discussion at Munique's IRT. The encoder simply reduces the two surround channels by 3 ms and adds these channels to the front channel. Thus, the left rear channel is attenuated and added to the left front channel. The encoder with specific tools in the surround channel has many advantages when encoding or recording multichannel film sound. Loudness and orientation of these instruments are incorrectly encoded. However, this encoder goes well with classical music, where the two surround channels are basically reverberation. 3 kHz attenuation is carefully selected through listening tests to form a stereo compatible encoding. The inventors said that the encoder should include this 3 dB attenuation when classical music is encoded, and can detect this condition through the relative levels of the front and surround channels of the encoder.

The main function of the function fc on the surround channel is to reduce the level of the surround channel by 3 dB in output mixing when the surround channel is much slower than the front channel. Circuitry is provided to compare the front and rear channels, and when the rear is 3 ms less, the value of fc is reduced by a maximum of 3 ms. Maximum attenuation is reached when the rear channel is 8 dB less intense than the front channel. This active circuit seems to work well. This makes the new encoder compatible with European standard encoders for classical music. The operation of the active circuit allows the tools intended to be strong in the back channel to be encoded at full level.

There is another function of the real coefficient mixing path fs for the surround channel. As the sound moves from the left front input to the left rear input, the active circuitry detects that the levels of these two inputs are similar and in phase. In this state, fc is reduced to 0 and fs is increased to 1. Changes in the real coefficients in this encoding result in more precise decoding of this fan type. This function isn't really necessary, but it's great.

There is an additional active circuit that has not yet been removed from the product. The level detection circuit checks the phase relationship between the center channel and the front left and right sides. Some pop recordings using five channels mix vocals to all three front channels. If there is a strong signal on all three inputs, the encoder output will have excessive vocal strength because the phases of the four front channels are added together. If this occurs, the active circuitry increases the attenuation of the center channel by 3 dB to restore the strength balance of the encoder output.

In summary, the active circuit

1.When the phases of two channels match, reduce the level of the surround channel by 2.2㏈,

2. Increase the real coefficient mixing path for the rear channel sufficiently to form an unsteered state when the two rear channels are out of phase,

3. Reduce the surround level by 3㏈ when the surround level is much lower than the front level,

4. Increase the rear channel level and negative phase when the level of the rear channel is similar to the front channel,

5. Configure the surround channel to use real number coefficients when the sound source is panned from the corresponding front input to the rear input,

6. Increase the level of the center channel of the encoder when the level of the center level and the front surround input are approximately equal,

7. Provided to reduce the level of the center channel of the encoder when all three front inputs have a common signal.

Other improvements to the encoder may include features similar to feature 2 above for the front channel. In current encoders, the encoding causes the decoder to place the sound behind when the two front channels have out of phase. We would like to detect this condition and configure the generated output with no steering.

27. Frequency dependent circuit at the decoder

2 is a block diagram of a frequency dependent circuit that follows a five channel version of a decoder. There are three parts: the variable lowpass filter, the variable shelf filter, and the HRTF (head related transfer function) filter. The HRTF filter changes its characteristics in accordance with the rear steering voltage c / s. The first two filters change their characteristics in response to the signal to indicate the average orientation of the input signal to the decoder during the interval between the strongly steered signals.

28. Background control signal

One of the main objectives of current decoders is to be able to optimally form a five channel surround signal from the first two channel stereo signal. It is also highly desirable for the decoder to reconstruct the 5-channel surround recording that was encoded in two channels by the encoder described as part of this application. These two applications differ in how surround channels are perceived. For a typical stereo input, most of the sound should be in front of the listener. Surround speakers contribute to a pleasant surround and ambience, but should not be noticeable. Encoded surround recording requires stronger and more aggressive surround speakers.

It is necessary to discriminate between 2-channel recording and encoded 5-channel recording in order to optimally play both input types without user adjustment. The background control signal is designed to make this determination. The background control signal BCS is similar to and derived from the rear steering signal cs. BCS represents the negative peak value of cs. That is, when cs is more negative than BCS, the BCS is configured to be equal to cs. When cs is more positive than BCS, the BCS value slowly dies out. However, the disappearance of the BCS involves other calculations.

Different types of music consist of a series of strong foreground tones or, in the case of a song, the words. There is a background between the foreground tones. The background can be composed of instruments that play different tones or reverberation. The circuit which derives the BCS signal keeps track of the peak level of the foreground timbre. When the current level is ˜7 dB less than the peak level in the foreground, the level of cs is measured. The value of cs during the gap between the foreground peaks is used to control the disappearance of the BCS. If the component present in the gap between the timbres is reverberation, there is a tendency to have a net rearward bias in the recording that was constructed by encoding the first five channels. This is because the reverberation on the first rear channel is encoded with back bias. In the first two-channel recording the reverberation has no final back bias. Cs for this reverberation will be zero or weakly potential.

The BCS derived in this way tends to reflect the recording form. When there is a valid rear steered component, the BCS is always strongly negative. However, if the reverberation of the recording has a final back bias, the BCS can be negative even if there is no strong steering back. You can use BCS to adjust the filter to optimize the decoder for stereo-surround input.

29. Frequency dependent circuit: 5-channel version

The first in the filter of FIG. 2 is a simple type low pass filter of 6 Hz per octave with adjustable cutoff frequency. When the BCS is positive or zero, this filter is set to a user adjustable value, but is typically about 4 Hz. When the BCS is negative, the cutoff frequency is raised until the filter is inactive when the BCS is behind 22 °. This low frequency filter assumes that the rear output is less forced when the first stereo component is played. This filter has been part of the decoder since at least V1.11, but in the original decoder it was controlled by cs, not BCS.

The second filter is a variable shelf filter. The low frequency part (pole) of this filter is fixed at 500 Hz. The high frequency part (zero point) changes with user adjustment and BCS. This filter implements "sound stage" control in the current decoder. U.S. Patent Application No. In 08 / 742,460, the "sound stage" is implemented via a matrix element using "tv matrix" correction. The first decoders based on this work reduced the overall level of the rear channel when steering was neutral or forward. In the new decoder presented here, the matrix elements do not include a "tv matrix" correction.

In the new decoder, when the sound stage control is set to "rear", the high frequency portion of the shelf filter is set equal to the low frequency portion. In other words, the shelf has no attenuation, and the filter has a uniform response.

When the sound stage control is set to "neutral", the setting for the high frequency zero point changes. When the BCS is positive or zero, the zero shifts to 710 Hz and 3 Hz attenuation occurs at high frequencies. For high frequencies, the result is the same as for the original decoder. There is a 3 dB attenuation when the steering is neutral or forward. However, low frequencies are not attenuated. These come from the side of the room to the fullest level. The result is greater low frequency fullness and encirclement without scattering high frequencies behind the scenes. The high frequency zero moves towards the pole when the BCS is negative, so that the shelf filter has attenuation when the BCS is about 22 ° to the rear.

When the sound stage control is set to "forward", the operation is similar, but the zero moves to 1 ms when the BCS is zero or positive. As a result, the high frequency is attenuated by 6 Hz. Once again, when the BCS goes negative, the attenuation is removed.

The third filter is controlled by cs, not BCS. This filter is designed to emulate the frequency response of the human head and the auricle when the sound source has an azimuth of approximately 150 ° from the front of the listener. This type of frequency response curve is called "head related transfer function" or "HRTF". These frequency response functions have been measured at many angles for many different people. In general, when the sound source is approximately 150 ° from the front, there is a strong notch at a frequency response of about 5 kHz. A similar notch exists when the sound source is in front of the listener. In this case, the notch is at about 8 ms. Sound sources on the listener's side do not generate this notch. The human brain uses the presence of a notch at 5 Hz as one way of detecting that the sound source is behind the listener.

The current standard for 5-channel sound reproduction recommends that the two rear speakers should be at +/- 110 or 120 ° from the front slightly behind the listener. This speaker position provides good envelopment at low frequencies. However, the sound from the listener's side does not generate the same level of stimulus as the sound located completely behind the listener. Very often, film directors want sound effects to come from behind, not from the listener's side.

Often the listening room does not have the proper size or shape to place the speaker completely behind the listener, and the lateral position is the best that can be achieved.

The HRTF filter of the decoder adds a frequency notch of the rear sound source, so that the listener feels as if the speaker is farther than the actual position when listening to the sound. The filter is designed to change with cs. When cs is positive or zero, the filter is maximum. Because of this, the ambient sound and reverberation are felt behind the listener. When cs becomes negative, the filter is reduced. When cs is approximately -15 °, the filter is completely removed and the sound source feels like coming completely out of the side. When cs goes more noteworthy, the filter is applied once again, so the sound source feels like it is behind the listener. The filter is slightly modified to correspond to the HRTF for sound in full rear when the cs is completely in the rear.

30. Frequency dependent circuit: 7 channel version

3 shows a frequency dependent circuit in a seven channel version decoder. They are shown as being composed of three parts, although in practice the second two parts can be combined into one circuit.

The first two parts are identical to the two parts of the five channel decoder and perform the same function. The third part is only in the 7-channel decoder. V1.11 and US patent application no. In 08 / 742,460 the side and rear channels have separate matrix elements. These devices operate so that the lateral and rear outputs are the same except for delay when cs is positive or neutral. The two outputs remained the same until cs became more negative than 22 °. As the steering moves further to the rear, the side output is attenuated by 6 dB and the rear output is increased by 2 dB. Because of this, the sound feels like moving backwards on the listener's side.

In the present decoder, the discrimination between the side output and the rear output is achieved by the variable shelf filter of the side output. The third shelf filter of FIG. 3 has no attenuation when cs is forward or zero. When cs becomes more negative than 22 °, the zero point of the shelf filter moves rapidly toward 1100 Hz, resulting in attenuation of about 7 Hz at high frequencies. Although this shelf filter is described as a filter separate from the shelf filter providing the "sound stage" function, the operation of these two shelf filters can be combined into a single shelf via suitable control circuitry.

While the preferred embodiments of the invention have been described and illustrated herein, many other possible embodiments exist, and these and other variations and modifications will be apparent to those skilled in the art without departing from the spirit of the invention.

Claims (23)

A pair of left and right audio input signals comprising directionally encoded components and non-directional components is redistributed into a plurality of output channels for playback through a speaker surrounding the listening area, and the left and right audio signals A surround sound decoder for determining a directional component and interpolating means for generating at least a left-right steering signal and a center-surround steering signal therefrom. Left and right input terminals for receiving the corresponding left and right audio input signals, Left and right delay means for generating a left and right audio signal delayed from the left and right audio input signals, A number equal to twice the number of the plurality of output channels consisting of a pair of a first element receiving the delayed left audio signal and a second element receiving the delayed right audio signal, each providing an output signal; A plurality of multiplier means for multiplying their input audio signal by a variable matrix coefficient controlled by one or both of the steering signals and One for each of the plurality of output channels, each including a plurality of summation means for receiving a pair of output signals of the multiplier means and generating one of the plurality of output signals at its output; The decoder reduces directionally encoded audio components at the output not directly related to playback in the intended direction to maintain a constant overall intensity for the signal, and directed at the output directly related to playback in the intended direction. Having the variable matrix value configured to increase the encoded audio component, preserving high separation between the left and right channel components of the omni-directional signal independent of the steering signal, and whether there is a directionally encoded signal Otherwise, and if present, a surround sound decoder that effectively maintains a constant loudness defined as the overall audio intensity level of the omni-directional signal, regardless of the intended direction. The method of claim 1, Wherein the plurality of output channels is five and is identified as left front, center, right front, left surround and right surround. The method of claim 2, And further comprising a frequency dependent variable filter that follows the left and right surround outputs to change the frequency response and phase response of the output in the described manner, wherein the change is a surround or detected signal in the left and right audio input signals. Decoder controlled by a plurality of control signals responsive to the presence of a background peripheral component. The method of claim 2, There is also a frequency dependent variable filter means and additional delay means for following the left and right surround outputs for providing side and rear output channels from each surround output to change the frequency and phase response of several outputs in the manner described. Wherein the modification is controlled by a plurality of control signals responsive to the presence of surround or background peripheral components detected in the left and right audio input signals. The method according to claim 3 or 4, The control signal is A center-surround control signal responsive to a ratio of a surround or anti-phase signal included in the left and right audio input signals and a center signal component whose phase coincides; And a background control signal responsive to the presence of an antiphase signal component included in the left and right audio input signals during a period when no strongly steered signal is present. The method according to any one of claims 1 to 4, At least two different operating modes are provided, and matrix coefficients are controlled differently from each other by said steering signal in different operating modes. The method of claim 6, The film operation mode is optimized for playback of surround encoded audio signals derived from film soundtracks and other video sources, and the music operation mode is optimized for playback of music recordings or broadcasts. The method of claim 7, wherein For decoding of film sources, the matrix values for the left and right front outputs are configured to remove as much of the center component of the input signal as possible, wherein the matrix values for the center outputs have attenuation of the center output of at least 4 than the previous standard decoder. ㏈ start large, and are configured to rapidly decrease as the center / surround steering signal becomes more positive, the intermediate matrix value of which must maintain a constant ratio of the uncorrelated and center components of the input signal at the output of the decoder. A decoder characterized by the requirement. The method of claim 7, wherein For decoding of a music source, the matrix value of the center output is such that the center attenuation begins at least 4 dB greater than the standard decoder, gradually decreasing to the maximum value for the standard decoder reaching a center / surround steering signal value of about 20 °. Attenuation is relatively constant when the steering value is increased, and the left and right front matrix values are configured such that the center component of the input signal is not removed as much as possible from these outputs, but is uncorrelated with the input signal at the output of the decoder. Carefully adjusted to maintain the intensity ratio of the component and the center component, the operation of the center, left, and right front elements adds to the center / surround steering value, which produces a roughly 6 dB level difference between the center output and the left or right front output. Decoder which is limited in general. The method of claim 1, And the left and right front matrix elements are configured such that a backward encoded input signal does not produce any output from the front output such that the orientation is located between the left rear and the right rear. The method of claim 1, The left and right front matrix elements are configured such that there is a level increase of about 3 dB for signals that do not have a final left / right component with a center / surround steering value of about 22 °, the level increase being center / surround. Decoder, characterized in that the steering value decreases to zero and increases to 45 ° or decreases to zero when the left / right steering value increases from zero to + -45 °. The method according to claim 4 or 5, And further circuitry for detecting the direction of the background sound between the timbre or syllable of the input component to form a background control signal, wherein the background control signal is a value of the center-surround steering signal when the center-surround steering signal is negative. And the background control signal is slowly set to positive when the direction of the background sound between the timbre and syllable is forward, and the background control signal is set to the negative value or the standard two channel component when the surround encoding component is reproduced. Decoder which tends to maintain a positive or zero value when played back. The method of claim 12, And the background control signal is used to control the relative loudness of the front and rear outputs so that the loudness of the rear output is reduced when the background between the tones is in neutral or positive direction. The method of claim 12, The background control signal controls the variable lowpass filter of the rear output so that the cutoff frequency is set to rise to a user adjustable value when the background directed signal is positive or zero and to a high value when the background directed signal is negative. A surround output is less compulsorily formed when the two-channel component of is played back. The method of claim 12, The background control signal controls the variable shelf filter so that the frequency above 500 Hz of the rear output is attenuated by a user adjustable value when the background control signal is positive or zero, and this attenuation is reduced to zero when the background control signal is negative. And thus the surround output is less forcedly formed when the first two-channel component is reproduced. The method of claim 5, The rear output of the matrix is separated into the lateral output and the rear output by the combination of the additional delay of the rear output and the variable lowpass filter of the side output, wherein the lowpass filter is positive for the center-surround steering signal to be greater than -22 °. When the center-surround steering signal becomes more negative than -22 °, the lowpass frequency is reduced to a final value of 500 Hz when the center-surround steering signal reaches a minimum value of -45 °. Decoder, characterized in that. The method of claim 4, wherein The left and right surround outputs of the five-channel version decoder output the frequency response of the human head / wheel system to a sound source located at an azimuth of more than 150 ° from the front so that the filter has maximum effect when the center-surround steering signal is zero or positive. With an additional variable filter that emulates, when the center-surround steering signal changes from 0 to -15 °, the filter operation decreases to 0, and when the center-surround steering becomes more negative, the filter is once again maximized. And a full rearward correction to correspond slightly to the frequency response of the human head-wheel system to the sound source when the center-surround steering signal reaches a minimum value of -45 °. The intensity ratio of the input signal is maintained in the output signal, the direction of the input signal is preserved in the phase and amplitude relationship of the different components of the output signal, and when the input signal is panned from the input channel to another input channel, the phase / Automatically the five channels of the audio input signal to the two output channels so that the amplitude relationship is decoded as close to the original direction as possible through the decoder according to claim 1 and as close to the original direction as possible when decoded via a standard film decoder. In the encoder circuit for mixing with 5 input terminals for receiving 5 audio input signals respectively identified as left, center, right, left surround and right surround, One of the two output channels is two output terminals provided to an external device, Means for determining an amplitude / phase relationship between the five audio input signals and generating a control signal therefrom; Means for mixing a fixed ratio of each input channel or a variable ratio responsive to the control signal to one of the two output channels. The method of claim 18, And the circuit comprises means for actively correcting the mixing value such that the intensity of the output signal matches the intensity of the input signal when the same signal is based on the rear input in the same phase or antiphase. The method of claim 18, The circuitry comprises means for actively correcting the mixing value such that when the same signal is applied to the rear output in antiphase, the output of the encoder has a relative phase of 90 ° representing an unsteered state to the decoder. Circuit. The method of claim 18, The circuitry comprises means for actively canceling the phase shift network of the rear channel when the signal is panned between one of the front side inputs and the rear input, for example from the front left input to the rear left input. Decoder circuit. The method of claim 18, The circuitry comprises means for actively determining the presence of a common signal at all three front inputs and for adjusting the mixing level of the center channel to maintain the overall strength of this common signal at the output of the encoder. Decoder circuit. The method of claim 18, The circuit comprises means for comparing the level of the rear channel with the level of the three front channels, and when the rear level is less than the front level, the active circuit reduces the mixing level of the rear channel by 3 dB and the circuit A decoder circuit which allows the encoder to optimally encode music when this includes most of the reverberation.