EP0629335B1 - Appareil de reproduction de son d'ambiance - Google Patents

Appareil de reproduction de son d'ambiance Download PDF

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EP0629335B1
EP0629335B1 EP93904278A EP93904278A EP0629335B1 EP 0629335 B1 EP0629335 B1 EP 0629335B1 EP 93904278 A EP93904278 A EP 93904278A EP 93904278 A EP93904278 A EP 93904278A EP 0629335 B1 EP0629335 B1 EP 0629335B1
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reproduced
encoded
gain
decoder
frequency
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EP0629335A1 (fr
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Michael Anthony Gerzon
Geoffrey James Barton
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Trifield Productions Ltd
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Trifield Productions Ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/15—Aspects of sound capture and related signal processing for recording or reproduction
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/11—Application of ambisonics in stereophonic audio systems

Definitions

  • the present invention relates to techniques for directionally encoding and reproducing sound, and particularly, but not exclusively, to the technique known as surround sound and to the provision of improved surround sound decoders and reproduction systems using such decoders.
  • the present invention is applicable to a number of different surround sound techniques, including Ambisonics, and to other encoding techniques.
  • Ambisonics was developed in the 1970's and early 1980's based on the idea of encoding information about a 360° directional surround-sound field within a limited number of recording or transmission channels, and decoding these through a frequency-dependent psychoacoustically optimised decoder matrix.
  • the matrix is adapted to the specific arrangement of loudspeakers in the listening room, so as to recreate through that specific layout the directional effect originally intended.
  • Examples of Ambisonic systems are described and claimed in the earlier British patents numbers 1494751, 1494752, 1550627 and 2073556, all assigned to National Research Development Corporation.
  • Ambisonic techniques are also described in a number of published papers including the paper by M.A. Gerzon, "Ambisonics in Multichannel Broadcasting and Video” published at pp 859-871 of J Audio Eng. Soc., Vol. 33, No. 11, (1985 Nov.).
  • a decoder for decoding directionally encoded audio signals for reproduction via a loudspeaker layout over a listening area, comprising:
  • a surround sound decoder including matrix decoding means for decoding a signal having pressure-related and velocity-related components thereby providing output signals representing feed signals for a plurality of loudspeakers, in which the values of the coefficients of the matrix decoding means are such that the magnitude r V of the real part of the ratio of the reproduced velocity vector gain to the reproduced pressure gain varies with azimuthal direction at at least some frequencies.
  • the decoder is an Ambisonic decoder.
  • velocity vector having magnitude r V energy vector having magnitude r E and pressure signal P are formally defined and discussed in relation to the Ambisonic decoding equations in the detailed description and theoretical analysis below.
  • a decoding matrix has been used which is frequency-dependent so as to ensure that r V equals 1 at low frequencies and that r E was larger at high frequencies.
  • the velocity vector had a magnitude which was substantially constant for all directions and the pressure signal P had exactly the same directional gain pattern (as a function of encoded azimuth ⁇ ) at low and high frequencies, apart from a simple adjustment of overall gain with frequency.
  • the present invention provides an Ambisonic decoder arranged to satisfy the Ambisonic decoding equations in the case where the r V varies with azimuth, and, preferably, the directional gain pattern of the pressure signal P varies with frequency.
  • the back-sound gain divided by front-sound gain for the pressure signal will have a smaller value at low frequencies (for which r V typically equals 1) than at higher frequencies (for which typically r E is maximised with a greater value for front- stage sounds than for back-stage sounds).
  • the present inventors have recognised that the directional gain pattern of the pressure signal P (which for layouts of speakers at identical distances is the sum of the speaker feed signals) can be varied with frequency while still giving solutions of the Ambisonic decoding equations and that this gives an extra degree of freedom which may be used to optimise the performance of the Ambisonic decoder.
  • r V vary substantially with encoding azimuth ⁇ rather than to be substantially constant with azimuth as it was in the prior art Ambisonic decoders. This is particularly important in improving the stability of images.
  • the degree of image movement with lateral movement of the listener is proportional to 1-r E , so that the greater the value of r E the less the movement and hence the greater the image stability.
  • r E tends to be maximised when r E substantially equals r V .
  • r E varies with the encoding azimuth and so it is found that the best performance is obtained by having r V also vary with encoding azimuth so as to track the value of r E in so far as this is possible.
  • r E varies with direction with higher harmonic components than r V so that only rarely can r E and r V have exactly the same values for all azimuths. Nonetheless it is found that fairly close matching of the two quantities generally gives improved high frequency results.
  • the encoded directional signal is modified to increase the relative gains of sounds in those directions in which the magnitude r E of the reproduced energy vector is largest.
  • an Ambisonic decoder including decoder matrix means arranged to decode a signal having components W, X, Y where W is a pressure-related component and X and Y are velocity-related components, as herein defined, the matrix decoding means producing thereby output signals representing loudspeaker feeds for a plurality of loudspeakers, further comprises transformation means arranged to apply a transformation to the signal components W, X, Y thereby generating transformed components W', X', Y' for decoding by the decoding matrix means.
  • the transformation may be a Lorentz transformation.
  • the sound field is preferably encoded using the so called B-format.
  • B-format encodes horizontal sounds into three signals, W, X and Y where W is an omnidirectional signal encoding sounds from all azimuths with equal gains equal to 1 and X and Y correspond to figure-of-eight polar diagrams with maximum gains ⁇ 2 aligned with the orthogonal X and Y axes respectively.
  • W is an omnidirectional signal encoding sounds from all azimuths with equal gains equal to 1
  • X and Y correspond to figure-of-eight polar diagrams with maximum gains ⁇ 2 aligned with the orthogonal X and Y axes respectively.
  • This format is shown in Figure 2.
  • the Lorentz transformation means are arranged to apply a forward dominance transformation.
  • a forward dominance transformation has the effect of increasing front sound gain by a factor ⁇ while altering the rear sound gain by an inverse factor 1/ ⁇ .
  • a forward dominance transformation is particularly valuable with decoders in accordance with the first aspect of the invention, since as noted above such decoders can otherwise give excessive gains for rear sounds.
  • a surround sound decoder in a surround sound decoder include decoder matrix means arranged to decode a signal and provide output signals corresponding to a plurality of loudspeaker feed signals, the decoder is arranged to output signals representing feed signals for an arrangement of loudspeakers surrounding a listener position comprising at least two pairs of loudspeakers disposed symmetrically to the two sides of the listener position and for at least one further loudspeaker positioned between one of the said pairs of loudspeakers.
  • the decoder is an Ambisonic decoder.
  • the at least one further loudspeaker is positioned centrally between the two front-stage loudspeakers.
  • an Ambisonic decoder including matrix decoding means for decoding a signal having pressure-related and velocity-related components and thereby providing output signals representing feed signals for a plurality of loudspeakers, in which the values of the coefficients of the matrix decoding means are such that the directional gain pattern of the pressure-related component P of the reproduced signal is different at different frequencies, the decoder not being a 2.5 channel Ambisonic decoder responsive to 3-channel surround sound at some audio frequencies and to a 2-channel surround sound at some other frequencies.
  • a decoder including an Ambisonic decoder according to any one of the preceding aspects, and further comprising means for decoding a supplementary channel E, thereby providing improved stability of and separation between the front and rear stages.
  • a decoder including an Ambisonic decoder according to any one of the preceding aspects, and further comprising means for decoding a supplementary channel F thereby providing a further output signal for use in cancelling crosstalk between front and rear stages.
  • E is encoded with gain k e (1-c e (1-cos ⁇ )) for
  • c e lies between substantially 3 and 3.5 and more preferably is equal to substantially 3.25.
  • This aspect of the present invention provides a decoder which gives a further improvement in frontal-stage image stability. While the Ambisonic decoders of the preceding aspects of the present invention in themselves give a significant improvement in stability even greater stability may still be desirable when, for example, the decoder is to be used in an HDTV system.
  • the present inventor has found that by adding at least one additional channel signal E to provide a feed for a centre-front loudspeaker, a system results which offers much of the image stability attainable with the three or four-speaker frontal stereo systems known in the art, while retaining the flexibility of the Ambisonic decoders in providing optimally decoded results via a variety of speaker layouts. Moreover, it is found that the 5 or 6-speaker Ambisonic decoders of the present invention provide a far better basis for such a system enhanced by a supplementary channel than do the conventional 4-speaker Ambisonic decoders known in the art.
  • two supplementary channels E, F are used in addition to channels W, X, Y to provide a format termed by the inventor "enhanced B-format". This enhanced format is described fully in the detailed description below.
  • the present invention in its different aspects is applicable to audio signals that are directionally encoded.
  • Directionality of sounds can be encoded in Various ways, using two or more related audio signal channels. In all of these methods, each encoded direction of sound is mixed into the audio signal channels with gains (which may be real or complex, and may be independent of or dependent on frequency) on each signal channel whose values as a function of direction characterise the directional encoding being used.
  • directional encoding is characterised by the relative gains with which a sound is mixed into the audio signal channels as a function of intended direction.
  • Examples of directional encoding include the familiar case of conventional amplitude stereophony in which the direction of sounds in two stereo channels is encoded by the relative amplitude gain in two channels normally intended for reproduction via respective left and right loudspeakers.
  • the means of encoding gains as a function of direction often used in studio mixing is the device known as a stereo panpot which allows alteration of the encoded stereo direction by giving adjustment of the relative gains of a sound in the left and right channels.
  • An alternative means of encoding directionally often used is a coincident stereo microphone recording array, where coincident directional microphones pointing in different directions are used, and sounds recorded in different directions around such a microphone array will be encoded into the two channels with different gains determined by the gain response of the two microphones for that incident sound direction.
  • These preferred methods of surround sound directional encoding with which the invention may be used encode horizontal directions as linear combinations of three signals, W with constant gain 1 as a function of direction, and directional signals X and Y whose gains as a function of encoded direction follow a figure-of-eight or cosine gain law pointing in two orthogonal directions.
  • X may be chosen to have a gain ⁇ 2 cos ⁇ and Y a gain ⁇ 2 sin ⁇ , where ⁇ is the angle of a direction measured anticlockwise from due front in the horizontal plane.
  • the directionality for B-format can be encoded either by a suitable B-format panpot such as has been described by M.A. Gerzon & G.J.
  • any three independent linear combinations of the signals W, X and Y may be used to encode directionality, since B-format signals may be recovered from these by using a suitable inverse 3 x 3 matrix.
  • Any decoder for reproducing B-format may be converted to one for three such linear combination signals by preceding or combining the B-format decoder with an appropriate 3 x 3 matrix having the effect of recovering B-format signals.
  • 360° directionality may also be encoded into just two channels as complex linear combinations of W, X and Y.
  • such directionally encoded 2-channel signals may be derived from B-format signals by means of a phase-amplitude matrix incorporating 90° phase difference networks.
  • the invention is also applicable to the decoding of a broad range of such 2-channel directionally encoded sound signal channels, including the prior art BMX and UHJ systems, which are of this form. It is also applicable to conventional 2-channel amplitude stereophony encoding, since this directional encoding may be converted to a BMX encoding by inserting a 90° phase difference between the two channels.
  • the invention may also be applied to signals in which additional to a 360° azimuthal encoding, directions in three dimensional space are encoded including for example elevated sounds, such as are described in M.A. Gerzon, "Periphony: With-Height Sound Reproduction", J. Audio Eng. Soc., Vol. 21, pp. 2-10 (1973, Jan./Feb.) and in the cited 1985 Gerzon reference. Additional directionally encoded channels may be added, such as a signal Z with vertical figure-of-eight directional gain characteristic, or the directional enhancement signals E and F described elsewhere in this description.
  • the invention is additionally applicable to other systems of directional encoding having substantially the same relative gains between signal channels as the systems described above, even if their overall or absolute gains or phases as a function of encoded direction varies.
  • the decoders of the present invention while being generally applicable have particular advantages when used with audiovisual systems.
  • the present invention also encompasses TV, HDTV, film or other audiovisual systems incorporating a decoder in accordance with any one of the preceding aspects in its sound reproduction stages.
  • Decoders according to the invention may be implemented using any known signal processing technology in ways evident to those skilled in the art, and in particular either using electrical analogue or digital signal processing technology, or a combination of the two.
  • matrix networks or circuits may be implemented using resistors or voltage or digitally-controlled active gain elements to implement matrix gain coefficients in combination with active mixing devices such as operational amplifiers to perform addition or subtraction of signals.
  • Frequency-dependent elements such as cross-over network filters may be implemented using any familiar active filter topology.
  • Good approximations to relative 90° phase difference networks may be implemented by pairs of all-pass networks each comprising cascaded first order all-pass poles of the kind extensively described in the previous literature on, for example, quadraphonic or surround-sound phase-amplitude matrixing or on single sideband modulation using quadrature filters.
  • analogue-to-digital converters may be used to provide signals in the sampled digital domain, and the decoders may be implemented as signal processing algorithms on digital signal processing chips.
  • filters may be implemented using digital filtering algorithms familiar to those skilled in the art, and matrices may be implemented by multiplying digital signal words by gain coefficient constants and summing the results.
  • the digital outputs may be converted back to electrical analogue form by using digital-to-analogue converters.
  • matrix, gain and filter means in decoders may be combined, rearranged and split apart in many ways without affecting the overall matrix behaviour, and the invention is not confined to the specific arrangements of matrix means described in explicit examples, but includes, for example, all functionally equivalent means such as would be evident to one skilled in the art.
  • the outputs of decoders will typically be fed to loudspeakers using intermediary amplifier and signal transmission stages which may incorporate overall gain or equalization adjustments affecting all signal paths equally, and also any gain, time delay or equalization adjustment that may be found necessary or desirable to compensate for the differences in the characteristics of different loudspeakers in the loudspeaker layout or for the differences in reproduction from the loudspeakers caused by the characteristics of the acoustical environment in which the loudspeakers are placed. For example, if the reproduction from one loudspeaker is found to be deficient in a given frequency band relative to the reproduction from the other loudspeakers, a compensating boost equalisation may be applied in that frequency band to feed that loudspeaker without changing the functional performance of the decoder according to the invention.
  • decoders may be fed to loudspeaker layouts using loudspeakers covering only a portion of the audio frequency range, and different loudspeaker layouts may be used for different portions of the audio frequency range.
  • the directionally encoded audio signals may be fed to different decoder algorithms for loudspeaker layouts used in different frequency ranges by means of cross-over networks.
  • the decoder should incorporate or be used with means of adjusting the decoder matrix coefficients in accordance with the loudspeaker layout it is intended to use with the decoder, so that correct directional decoded results may be obtained.
  • prior art Ambisonic decoders for rectangular loudspeaker layouts incorporated gains in two velocity signal paths in decoders as a means of adjusting for different shapes of rectangle, and in commercially available decoders this is implemented by means of a potentiometer adjusting the gain of two velocity signal paths whose settings are calibrated either with pictures of the layout shape or with the ratio of the two sides of the rectangle.
  • British Patent 2073556 filed 1980 discloses the provision of gain adjustments in velocity signal paths in decoders for certain loudspeaker layouts where loudspeakers are disposed in diametrically opposite pairs.
  • loudspeaker layout control means may constitute a number of adjustable matrix coefficients in the decoder linked to a means of adjusting these in accordance with an intended or actual reproduction loudspeaker layout.
  • the adjustment means may constitute potentiometers or digitally or voltage controlled gain elements in analogue implementations or a means of computing or looking up in a table the matrix coefficients in a digital signal processor, and a means incorporating these coefficients in a signal processing matrix algorithm.
  • the method of adjustment may be in response to a control menu specifying the shape of the loudspeaker layout, or one or more controls adjusting analogue parameters defining the loudspeaker layout shape, or a combination of these, or any other well known means of adjusting parameters in signal processing systems.
  • the loudspeaker layout may be determined by geometrical measurements, for example with a measurement tape, or by any known automatic or semi-automated measurement technique such as those Used to determine distance in autofocus cameras. In the automated case, the results of the measurements may be used to compute appropriate matrix coefficients, for example by interpolation between the pre-computed values of matrix coefficients on a discrete range of loudspeaker layouts computed by the methods indicated in the appendices.
  • the layout control adjustment of matrix coefficients has the effect of altering only signals represented reproduced velocity, but not signals representing the reproduced pressure.
  • the layout control adjustment of matrix coefficients has the effect of altering not only signals representing reproduced velocity, but also signals representing the reproduced pressure as well, as may be seen by computing the pressure signal (which is the sum of the loudspeaker output signals) for various loudspeaker layouts disclosed in the appendices.
  • the invention may be used in conjunction with the methods disclosed in British Patent 1552478 to compensate for different loudspeaker distances from a preferred listening position in the listening area.
  • the decoding matrices of the present invention may be combined with time delays and gain adjustments for the output loudspeaker feed signals that compensate for the altered time delay and gains of sounds arriving at the preferred listening position caused by unequal loudspeaker distances from the preferred listening position.
  • FIG. 1 is a block diagram illustrating a typical ambisonic encoding/reproduction chain.
  • An incident sound field is encoded by a SoundField microphone 1 into B-Format signals.
  • the resulting WXY signals are applied to an ambisonic decoder.
  • the ambisonic decoder 2 applies to the WXY signals a decoding matrix which derives output signals from weighted linear combinations of the W X and Y signals. These output signals are then amplified by amplifiers 3 and supplied to speakers 4 arranged in a predetermined format around a listener 5.
  • the sound field microphone 1 is a one-microphone system such as that currently commercially available from AMS as the Mark IV SoundField microphone.
  • FIG 3 shows the horizontal polar diagrams of the B-format signals.
  • B-format can be extended to include four full-sphere directional signals as shown in Figure 4.
  • the ambisonic decoding equation used to derive the loudspeaker feed signal from the WXY signals is determined for a given loudspeaker layout in accordance with certain psychoacoustic criteria formally represented by the so-called Ambisonic equations. As described in further detail below, these define different constraints appropriate to different frequency bands for the reproduced sound in terms of the energy vector and velocity vector together with the scalar pressure signal P.
  • the localisation given by signals emerging with different gains g i from different loudspeakers around a listener can be related to physical quantities measured at the listener location.
  • localisation given at low frequencies by interaural phase localisation theories below about 700 Hz is determined by the vector given by dividing the overall acoustical vector velocity gain of a reproduced sound at the listener by the acoustical pressure gain at the listener.
  • the real part of this vector is used.
  • the resulting vector for natural sound sources, has length one and points at the direction of the sound source.
  • the length r V of this vector should ideally be as close to 1 as possible, especially for sounds intended to be near azimuths ⁇ 90°, and the azimuth direction ⁇ V of this vector is an indication of the apparent sound direction.
  • These vector quantities can be computed from a knowledge of the gains g i with which a sound source is fed to each of the loudspeakers, as follows. Suppose one has n loudspeakers all at equal distances from the listening position; let the i'th loudspeaker be at azimuth ⁇ i and reproduce a sound with gain g i . (although the theory can be developed for complex gains g i . we here assume that g f is real for simplicity). The acoustical pressure gain is then simply the sum of the individual speaker gains. The velocity gain is the vector sum of the n vectors with respective lengths g i pointing towards azimuth ⁇ i (i.e.
  • ⁇ v is termed the velocity vector localisation azimuth, or Makita localisation azimuth, and is the apparent direction of a sound at low frequencies if one turns one's head to face the apparent direction.
  • r v is termed the velocity vector magnitude and ideally equals one for single natural sound sources.
  • the two quantities ⁇ v and r v are indicative of apparent localisation direction and quality according to low-frequency interaural phase localisation theories, with deviations of r v from its ideal value of one indicative of image instability under head rotations, and poor imaging quality particularly to the two sides of a listener.
  • ⁇ E is termed the energy vector localisation azimuth , and is broadly indicative of the apparent localisation direction either between 700 Hz and around 4 kHz, or at lower frequencies in the case that the sounds arrive in a mutually incoherent fashion at the listener from the n loudspeakers.
  • r E is termed the energy vector magnitude of the localisation, and is indicative of the stability of localisation of images either in the frequency range 700 Hz to 4 kHz or at lower frequencies under conditions of phase-incoherence of sound arrivals. As before with r V , the ideal value for a single sound source is equal to 1.
  • r V For frontal stage stereo systems subtending relatively narrow angles (say with stage widths of less than 60°), it is found that the value of r V is not critical providing that it lies between say 0.8 and 1.2, but that the value of r E is an important predictor of image stability.
  • r E For surround sound systems aiming to produce images at each side of a listener, however, making r V equal one accurately at low frequencies becomes much more important, since the low-frequency localisation cue is one of the few cues that can be made correct for such side-stage images, and the accuracy of such localisation depends critically on the accuracy of r V .
  • r V 1 for all reproduced azimuths at low frequencies, typically under 400 Hz, at a central listening location.
  • 400 Hz it is instead desirable that the value of r E be maximised.
  • r E above 400 Hz is designed to be larger across a frontal stage than in side and rear directions, but not to the extent that side and rear sounds become intolerably unstable.
  • r E The optimisation of r E above about 400 Hz is partly a matter of design skill and experience obtained over a period of years, but some of this skill can be codified as informal rules of thumb. It is generally highly undesirable that l-r E should vary markedly in value for sounds at only slightly different azimuths, since such variations will cause some sounds to be much more unstable than other near by ones. In general, it is desirable that r E be maximised at the due front azimuth or across a frontal stage, and it is desirable that the values of r E in other directions vary smoothly.
  • a decoder or reproduction system for 360° surround sound is defined to be Ambisonic if, for a central listening position, it is designed such that
  • equation (13) is desirably not satisfied, although it is still found that satisfying equations (11) or (12) gives useful improvements in phantom image quality.
  • the reproduced acoustical pressure gain P had a directional gain pattern as a function of encoded azimuth ⁇ which was the same at low and high frequencies, apart from a simple adjustment of overall gain with frequency.
  • the values of the decoding matrix is such that while the output from the speakers satisfies the Ambisonic decoding equations, different directional gain patterns for the pressure signal P result at different frequencies.
  • a further characteristic of the decoding matrixes is that they result in the magnitude of the velocity vector r v varying systematically with encoding azimuth ⁇ rather than being substantially constant with azimuth as in previous Ambisonic decoders.
  • r v is made substantially to track r E in a mid-high frequency range of e.g, 700 Hz to 4 kHz while in a low frequency range up to e.g 400 Hz r v is as far as possible equal to one.
  • the decoder is arranged to apply a forward dominance transformation to the B-format signal W,X,Y.
  • X X' 1/2 ( ⁇ + ⁇ -1 ) X +2 -1/2 ( ⁇ - ⁇ -1 )
  • W Y' Y W' X' Y' satisfying the above equation where ⁇ is a real parameter having any desired positive value.
  • forward dominance control is important in various applications of B-format to HDTV. In a production application, it can be used to de-emphasise sounds from the rear of a sound field microphone while still giving a true B-format output. However, it can also be used in different reproduction modes relying on B-format input signals to de-emphasise rear sounds. In particular, in the new B-format Ambisonic surround-sound decoders of the present invention which may otherwise give excessive gain for rear sounds can be compensated for by a judicious application of a compensating forward dominance.
  • both the transformation and the decoder may be carried out by a single matrix.
  • Figures 6-9 show typical speaker layouts which will be considered for 360° surround-sound reproduction.
  • Figure 6 shows a rectangular speaker layout using left-back L B , left-front L F , right-front R F and right-back R B speakers at respective azimuths 180°- ⁇ , ⁇ , - ⁇ and - 180°+ ⁇ , supplemented by an extra centre-front C F loudspeaker.
  • Figure 7 shows a similar 5-speaker layout, except that now the azimuth angles ⁇ F of the front pair differs from that 180° + ⁇ B of the rear pair, so that the L B , L F , R F and R B speakers form a trapezium layout.
  • Figures 8 and 9 show similar rectangle and trapezium speaker layouts respectively, but this time supplemented by a frontal pair of speakers C L and C R at respective azimuths + ⁇ C and - ⁇ C .
  • the decoder architecture we now describe, and the associated methods of solution described in Appendix A works for quite general left/right symmetric speaker layouts, although the numerical details of the solution process can be extremely messy in particular cases, requiring the use of powerful computing facilities.
  • Figure 10 shows the general architecture of an Ambisonic decoder for the speaker layouts of Figures 6 to 9, based on equation (27).
  • a forward-dominance adjustment according to equation (3) so that the relative front/back gain balance and directional distribution of sounds can be adjusted.
  • Each of the three resulting B-format signals is then passed into a phase compensated band-splitting filter arrangement, such that the phase responses of the two output signals are substantially identical.
  • the cross-over frequency of the phase-compensated band-splitting filters will be around 400 Hz, and the sum of low and high frequency outputs will be equal to the original signal passed through an all-pass network with the same phase response.
  • the low-pass filters in Figure 10 might be the result of cascading two RC or digital first order low-pass filters with low frequency gain 1
  • the high-pass filters might be the result of cascading two first order high pass filters with the same time constants, with high frequency gain of -1; these filters sum to a first order all-pass with the same time constant, and have identical phase responses.
  • the resulting low and high frequency signals C F , S C (where it exists), M F , S F , M B and S B are then summed together and fed to output sum and difference matrices to provide speaker feed signals suitable for the speaker layouts of Figures 6 to 9.
  • phase compensation i.e. phase matching
  • the use of phase compensation (i.e. phase matching) of the band-splitting filters in Figure 10 is found to be highly desirable for surround sound decoders, since any "phasiness" errors due to relative phase shifts between signal components are magnified by the large 360° angular distribution of sounds, although in some cases, the use of filters that are not phase matched may prove acceptable.
  • the architecture of Figure 10 can be extended to 3 or more frequency bands by using a three-band phase-splitting arrangement with three decoding matrices, so as to optimise localisation quality separately in three or more bands.
  • a three band decoder might have crossover frequencies at 400 Hz and at or around 5 to 7 kHz so as to optimise localisation in the pinna-colouration frequency region above about 5 kHz.
  • bandsplitting into, say, low and high frequency bands as in Figure 10
  • other bandsplitting arrangements can be used, e.g. an all-pass path feeding high frequency decoding matrix coefficients and a phase-matched low-pass path feeding a decoding matrix whose coefficients are the difference between the low and high frequency coefficients.
  • part or all of the output sum and differencing process might be implemented in the decoding matrices before the band-combining summing process.
  • the forward dominance adjustment might be implemented directly as modified coefficients a C , b C , a F , b F , a B , b B rather than or in addition to an input forward dominance matrix.
  • the decoding matrices in Figure 10 will, in general, have matrix coefficients that vary with the speaker layout in use, so that a typical Ambisonic decoder implemented as in Figure 10 will have a means of causing the matrix coefficients to be altered in response to the measured or assumed speaker layout shape and angles shown in Figures 6 to 9. This may be done by a microprocessor software adjustment of coefficients, or by manual gain adjustment means.
  • Appendix A describes a general method for finding decoder solutions having the properties discussed above and Table 1 lists the values of the matrix coefficients for a given layout, and also describes the performance of the decoder in the different high and low frequency domains. Appendix B goes on to describe specific analytic solutions for particular layouts and Appendix C and Table 2 describe the low and high frequency solutions for nine different 5-speaker layouts.
  • the Ambisonic decoders of the present invention also provide a suitable basis for an enhanced decoder including additional channels providing improved stability of and separation between the front and rear stages.
  • an enhanced decoder including additional channels providing improved stability of and separation between the front and rear stages.
  • E additional channel signal denoted by E which incorporates a feed for a front loudspeaker.
  • a second added channel F can be used largely to cancel front-to-rear stage cross talk (which is largely due to the Y signal) and to widen the frontal stage.
  • the F signal gives a frontal stage reproduction closely approximating 3-channel frontal stereo. Any sounds assigned to such a high-stability frontal stage should also be encoded conventionally inTo the three B-format signals so that users discarding the E and F signals will still get B-format reproduction incorporating those sounds.
  • the present example provides a decoder for an enhanced B-format comprising up to 5 signals W,X,Y, E and F for studio production applications in horizontal surround-sound with enhanced frontal image stability. This encodes signals from azimuth ⁇ into the five channels with respective gains
  • Figure 11 shows a typical architecture for decoding enhanced B-format signals incorporating an Ambisonic decoding algorithm as described earlier for pure B-format signals.
  • F Y
  • the first step in an Enhanced B-format Ambisonic decoder is to derive the "cancelled" signals of equation (54) to feed a conventional 5- or 6- (or greater) speaker B-format Ambisonic decoder, and to take the E signal and to feed it with an appropriately chosen gain via a phase-compensating all-pass network (to match the filter networks in the Ambisonic decoder) to feed centre front loudspeakers directly.
  • the cancellation by E of a central speaker feed for encoded azimuths near ⁇ 60° can be adjusted for a given decoder design by a careful choice of the direct speaker feed gains of the E signal.
  • Figure 11 shows the E and F signals as being simply fed forward and mixed into the C F , S C and S F signal paths in a manner that is (apart from phase compensation) independent of frequency
  • a judicious feed of a small amount of E signal to the M F and M B signal paths, and of the F signal to the S B signal path in small amounts, possibly with a frequency dependence in the gain can yield a small but useful improvement in the overall performance of front-stage stereo sounds.
  • FIG. 11 illustrates the structure of an enhanced B-format decoder only in its most basic form, and that slightly more complex direct feeds of the E and F signals, with the dominant components feeding respectively C F and S C and S F may be used to optimise front-stage performance, possibly using gains that vary somewhat with frequency.
  • the function of the E signal is to increase the "separateness" of the frontal speaker feeds, especially that of centre-front, whereas the F signal has the effect of cancelling out the left/right difference signal from the rear speakers and increasing it at the front, thereby converting signals from true Ambisonic surround-sound signals to ones dominantly reproduced from a frontal stage.
  • the invention may also be applied to signals encoded by methods other than B-format, and in particular to a directionally encoded signal conveyed via two channel encoding, such as the two-channel surround-sound systems known as UHJ, BMX or regular matrix.
  • a directionally encoded signal conveyed via two channel encoding such as the two-channel surround-sound systems known as UHJ, BMX or regular matrix.
  • the 2-channel example described here will be described in terms of BMX encoding, although it will be realised that similar methods apply to other 2-channel encoding methods.
  • r V and ⁇ V in this case have similar psychoacoustic interpretations as in the case that g i are all real gains.
  • the equations (8) to (10) above may still be used to compute the localisation parameters r E and ⁇ E as before, and the equations (11) and (13) still define the desirable ambisonic decoding equations that we ideally wish a decoder to satisfy.
  • Phasiness is an unpleasant subjective effect caused by phase differences between loudspeakers, which causes broadening of illusory sound images, an unpleasant change in perceived tonal quality, and an unplesant "pressure on the ears" sensation.
  • the reproduced pressure signal P was substantially a single signal subjected only to a shelf filtering process to meet tne requirements of lower and higher frequency localisation, whereas the present decoder uses a pressure signal P whose polar diagram varies substantially with frequency, thereby achieving at higher frequencies, a value of r V that is not constant with encoded azimuth, but which instead roughly tracks the variation of r E with encoded azimuth.
  • Figure 13 shows an example of an ambisonic decoder for a rectangular speaker layout, which provides signals W d , X d , and Y d derived via phase-amplitude matrices from input signals M and S, in two separate signals paths at low and at high audio frequencies (typically with a cross-over frequency around 400 Hz) produced from the input signals via phase-compensated band splitting filters as described earlier in connection with figure 10.
  • Such low and high frequency signals W d , X d and Y d may then be fed to an output amplitude matrix, such as in equ. (X6) above, to derive output loudspeaker feed signals suitable for a layout such as shown in figure 12.
  • the band splitting filters need not preceed the phase amplitude matrices, but may alternatively follow them or be placed in the middle of the signal path of the matrices.
  • the relative 90° phase difference networks which are relatively complex and which form a part of any phase-amplitude matrix, to precede the bandsplitting filters.
  • FIG 12 Also shown in figure 12 is an optional "forward dominance" adjustment, which in general will differ from that for B format given earlier, but which performs a similar function of altering the gains and azimuthal distributions of different encoded azimuth directions while maintaining the characteristics of the particular locus of encoded directions characterising the directional encoding scheme for which the decoder is designed.
  • this decoder gives a lower rear-stage phasiness for given values of r E .
  • the above BMX decoder is not only applicable to use with rectangular loudspeaker layouts, since the output amplitude matrix in figure 13 may be replaced with alternative output amplitude matrices described in the prior art in British patents 1494751, 1494752, 1550627 and 2073556 for regular polygon, regular polyhedron or other loudspeaker layouts comprising diametrically opposed pairs of loudspeakers.
  • this matrix may be combined with the phase-amplitude matrices shown in figure 13.
  • the above-described decoder for conventional stereophony may also be used with signals encoded for the Dolby 2-channel cinema encoding method with advantageous results, and for signals encoded for regular matrix encoding.
  • decoders for 2-channel surround-sound encoded signals may be devised which feed 5- or 6-speaker layouts such as shown in figs. 6 to 9 and which satisfy the ambisonic decoding equations.
  • Figure 14 shows the block diagram of a typical 2-channel decoder satisfying the ambisonic decoding equations and capable of feeding five or six loudspeakers arranged as in figures 6 to 9. It will be seen that such a decoder is broadly similar to the B-format decoder of figure 10, except that it is fed by a 2-channel encoded signal, which is then fed to a first phase amplitude matrix which provides four output signal components W 2 , X 2 , Y 2 and B 2 which typically represent "pressure”, “forward component of velocity”, “leftward component of velocity” and “pressure phase shifted by 90°” signals, and these four signals are then bandsplit via phase compensated low and high pass filters and fed into respective low- and high-frequency amplitude matrices. The outputs of these matrices are then handled in an identical manner to that already described in connection with the later stages of figure 10.
  • the broad architecture of a decoder satisfying the ambisonic decoding equations is similar to the three channel B-format case, except that an input phase-amplitude matrix produces four signal components to be processed rather than three. Because much of the signal processing is similar, large parts of the signal processing circuitry or algorithm may be common to use for decoding from different 2-channel and 3-channel sources.
  • the signal X 2 may be multiplied by a real constant times j to obtain a signal Y 2 and the signal W 2 may be multiplied by a real constant times j to obtain a signal B 2 suitable for use in a decoder for 2-channel encoded signals according to the invention having the form shown in figure 14.
  • Eq. (X-38) Unlike in the BMX case, making the constant term in Eq. (X-38) zero is not enough to make Re[v x /P]:Re[v y /P] proportional to x/y.
  • Eqs. (X-42) and (X-43a) and (X43b) provide, for t 2 not equal 0, a more general solution to decoding W 2 and X 2 than known in the prior art but sharing substantially the same decoded azimuths.
  • the decoder may use a nonzero value of t 2 giving a value of r V which varies with direction, preferably being chosen so as to be larger across the frontal stage of encoded directions than across the rear stage, as in the BMX example given above.
  • decoders for 2-channel encoded signals of the form of Eqs. (X-29) having larger r V and r E across a frontal stage than across a rear stage will, as in the BMX example given earlier, have an undesirable gain variation with direction, with the less well localised rear stage sounds being reproduced with louder energy than the frontal stage.
  • the encoded 2-channel signals it is possible to subject the encoded 2-channel signals to a linear transformation which has very little effect on the encoding specification of directions except that the directions themselves are altered slightly and changed in gain.
  • transformed signals W 2 ', X 2 ' may replace W 2 and X 2 , for example in Eqs. (X-44), whenever it is desired to adjust the reproduced directional gains.
  • this directional transformation may be implemented as a complex 2 x 2 matrix on the directionally encoded signals before decoding, it is generally preferred if this matrix is either combined with the phase amplitude matrix in the decoder that derives the signals fed to the low and high frequency amplitude decoding matrices, or is implemented as a real linear matrix on signals such as W 2 , X 2 , Y 2 and B 2 .
  • Such preferred implementations avoid having to use additional phase amplitude matrixing, which is generally more costly and harder to do well than simple amplitude matrixing, due to the use of and need for relative 90 degree phase difference networks.
  • the invention may be applied using the methods and principles described in more complicated cases than those decoders explicitly described in the examples.
  • the invention may be used with loudspeaker layouts having seven, eight, nine or more loudspeakers disposed in a left/right symmetrical arrangement around a listening area.
  • the structure of such decoders is identical to that described with reference to Figures 10, 11 and 14, except that additional pairs of signals M i and S i are provided for feeding any left/right symmetric additional pairs L i and R i of speakers.
  • Such layouts with further loudspeakers again preferably have a greater number of loudspeakers across the frontal reproduced stage than across the rear reproduced stage so that the reproduced value of r E is larger for frontal stage sounds than for rear stage sounds, and again it is preferred that the values of r V and r E as a function of encoded sound direction should roughly track each other.
  • the B-format enhancement signals E and F may as before be added to and subtracted from respective M i and S i signals so as to increase the separation among front stage loudspeakers and between front and rear stages.
  • Such decoders are designed by exactly the same methods described in Appendix A and D in the 5 and 6 speaker case.
  • the quantities P, V x and V y of equation (28) are all left/right symmetric real linear combinations of W, X and Y.
  • g 1 so as to avoid repeating a lot of factors g in the analysis; however, it will be necessary to multiply the overall decoder coefficients thus obtain in equations (27) by an overall gain g afterwards, in order to obtain a desired overall gain of reproduction.
  • equation (33) For a suitable choice of ⁇ , to be determined, the left hand side of equation (33) can be factorised, and the right hand side of equation (33) can also be factorised provided only that the coefficients of W 2 and X 2 are of opposite signs.
  • k B (2 - 1 ⁇ 2 - k F s F )/s B .
  • a first step in finding solutions is to determine what values of k F cause either coefficient of W 2 or X 2 in equation (33) to equal zero, and to ensure that either k F exceeds the larger such value or is smaller than the smaller such value in order that the signs of the W 2 and X 2 coefficients differ.
  • a low frequency solution can be found by varying k F and C until equations (45b) are found to be satisfied; such values can be found by a numerical "hill climbing" or Newton's algorithm method.
  • this example also illustrates a typical defect encountered with 5-speaker and 6-speaker decoders designed according to the methods herein - namely that those directions for which r E is largest (and for which high frequency localisation is best) are reproduced with the lowest gain and those for which r E is smallest (and for which localisation is poorest) are reproduced with the highest gain. This is clearly undesirable.
  • the degree of forward dominance applied will be that which compensates for the difference in total energy gain between due front and due back sounds, at high frequencies, thereby giving equal gains in the front and back stages.
  • additional forward dominance need not be implemented as a separate pre-decoder adjustment as shown in Figure 10 (although such additional adjustment can be a useful listener control), but may preferably be implemented as altered coefficients a C , b C , a F , b F , a B and b B in the decoder matrices implementing equations (27) - there is no need to alter the Y coefficients since these are unaffected by forward dominance adjustments.
  • Table 2 lists a range of low and high frequency designs for 9 different 5-speaker layouts computed by the methods of appendices A and B, including forward dominance to compensate for front/rear gain variations and a gain adjustment of the high frequency decoder to ensure that it has the same gain at reproduced azimuths ⁇ 45° as the low frequency decoder.
  • r E at high frequencies for front stage sound is clearly enhanced by the use of the 5-speaker decoder, as seen in Table 2, as compared to similar 4-speaker rectangle decoders, thanks to the significant output from the C F speakers, with r E typically being increased from 0.7071 for a square layout to around 0.835 when a C F speaker is added. This almost halves the degree of image movement for front stage sounds. It will be seen that the value of r E at the sides and back is not drastically reduced, although the average value for r E over the whole 360° stage is not increased, and in fact is slightly reduced.
  • r E at the front is not brought up to ideal values very close to 1
  • the use of a 5-speaker Ambisonic decoder provides an improved image stability, as compared to previous designs, without giving an unacceptable loss of the rest of the surround sound 360° stage.
  • a 5-speaker Ambisonic decoder designed as here described matches TV use a great deal better than earlier decoders, and makes good use of just three transmission channels, although there is still a need for enhancing front-stage results by adding extra transmission channel signals.
  • equations (A15b) and (A15c) are a pair of simultaneous linear equations for M F and M B in terms of W and X, once one has chosen the constants k C , k F , C and ⁇ .
  • C F can be then derived from equation (A5), and S C , S F and S B are given via equation (27), where k B is given by equation (A8).
  • the forward dominance should be adjusted to minimise front/back reproduced gain variations, especially at higher frequencies, and the relative gain of the low and high frequency decoders should be adjusted so as to give broadly similar reproduced gain at all frequencies for at least front-stage sounds, as already described in connection with 5-speaker B-format decoders. This will yield the final 6-speaker B-format ambisonic decoder equations which typically may be implemented as in Figure 10.
  • 6-speaker designs will give a significantly larger r E across the frontal stage than a 5-speaker design, with only a small reduction of r E , spread across the rear and side stages, in other directions.
  • 6-speaker designs are often better than 5-speaker designs in their subjective performance. The price paid for this improved localisation quality performance is the need to use larger amounts of forward dominance in 6-speaker designs (typically over 6 dB) to compensate reproduced gain variations than is needed for 5-speaker designs.
  • Table 3b 6-speaker ambisonic decoder design of table 3a with forward dominance and high-frequency gain adjustments.

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Claims (38)

  1. Décodeur (2) pour décoder des signaux audio codés de manière directionnelle, en vue d'une reproduction par une disposition de haut-parleurs (4) sur une zone d'écoute, comprenant :
    une entrée (21) pour recevoir les signaux audio codés de manière directionnelle,
    des moyens matriciels (22, 23) pour modifier les signaux audio, et
    une sortie (24) pour délivrer le signal audio modifié, dans une forme appropriée pour une reproduction par les haut-parleurs,
    les coefficients des moyens matriciels étant tels qu'à une position d'écoute prédéterminée, dans la zone d'écoute, la direction du vecteur de vitesse reproduite et les directions de vecteurs d'énergie reproduites sont sensiblement égales l'une à l'autre et sensiblement indépendantes de la fréquence, dans une large plage de fréquence audio,
        caractérisé en ce que les coefficients de gain des moyens matriciels (22, 23) sont tels que la valeur du vecteur de vitesse reproduite rv d'un signal audio décodé varie systématiquement avec une direction de son codé, à des fréquence dans la gamme de, et au-dessus de, une fréquence audio moyenne prédéterminée.
  2. Décodeur suivant la revendication 1, caractérisé en ce que les moyens matriciels comprennent :
    un premier moyen matriciel (22) agissant à des fréquences audio basses, en dessous d'une fréquence d'équilibrage,
    un second moyen matriciel (23) agissant à des fréquences audio élevées, au-dessus de la fréquence d'équilibrage, le second moyen matriciel étant différent en résultat du premier moyen matriciel, et
    des moyens d'équilibrage (25) pour effectuer la transition autour de ladite fréquence d'équilibrage, entre le premier moyen matriciel et le second moyen matriciel, et
    en ce que la large gamme de fréquence dans laquelle la direction du vecteur de vitesse reproduite et la direction du vecteur d'énergie reproduite sont sensiblement égales l'une à l'autre et sensiblement indépendantes de la fréquence entoure ladite fréquence d'équilibrage et couvre de préférence plusieurs octaves, et
    en ce que la valeur du vecteur de vitesse reproduite, rv variant systématiquement avec la direction de son codée, à des fréquences dans la gamme de, et au-dessus de, la fréquence d'équilibrage.
  3. Décodeur suivant la revendication 1 ou 2, caractérisé en ce qu'au dessus de la fréquence audio moyenne prédéterminée, rV est nettement plus grande pour une direction codée antérieure que pour une direction codée postérieure diamétralement opposée.
  4. Décodeur suivant la revendication 2 ou 3, caractérisé en ce que la fréquence d'équilibrage se situe entre 150 Hz et 1 kHz et de préférence entre 200 Hz et 800 Hz.
  5. Décodeur suivant la revendication 2 ou 3 ou 4, caractérisé en ce pour toute les directions de son codées, la valeur du vecteur de vitesse reproduite rV est nettement plus grande en dessous de la fréquence d'équilibrage qu'au-dessus de la fréquence d'équilibrage.
  6. Décodeur suivant la revendication 2 ou 3 ou 4 ou 5, caractérisé en ce que, à des fréquences en dessous d'une gamme de transition d'équilibrage, autour de ladite fréquence d'équilibrage, la valeur du vecteur de vitesse reproduite rV est sensiblement indépendante de la direction de son codée.
  7. Décodeur suivant la revendication 6, caractérisé en ce que, à des fréquences en dessous de ladite gamme de transition d'équilibrage, la valeur du vecteur de vitesse reproduite rV est sensiblement égale à 1 pour toutes les directions de son codées.
  8. Décodeur suivant l'une quelconque des revendications précédentes, caractérisé en ce que, à des fréquences au-dessus de la fréquence audio moyenne prédéterminée précitée, la valeur du vecteur d'énergie reproduite rE varie, en fonction de la direction de son codée, d'une manière largement semblable à la valeur du vecteur de vitesse reproduite rV .
  9. Décodeur suivant l'une quelconque des revendications précédentes, caractérisé en ce que, des moyens de commande sont prévus pour régler les coefficients de gain des moyens matriciels afin d'adapter le décodeur pour une pluralité d'agencements de disposition de haut-parleurs, les moyens de commande modifiant le coefficient de gain de façon à modifier des composantes, y compris des composantes de pression, du signal audio reproduit.
  10. Décodeur suivant l'une quelconque des revendications 1 à 8, caractérisé en ce que les moyens matriciels sont agencés pour décoder le signal en vue d'une reproduction par une disposition de haut-parleurs présentant un plus grand nombre de haut-parleurs de reproduction sur un étage antérieur de directions que sur un étage postérieur de directions diamétralement opposé, les coefficients de gain des moyens matriciels étant tels qu'à sensiblement toutes les fréquences, la valeur du vecteur d'énergie reproduite rE de sons codés pour être reproduits à partir de directions de vecteurs dans l'étage antérieur précité est nettement supérieure à la valeur du vecteur d'énergie reproduite rE de sons codés pour être reproduits à partir de directions de vecteur diamétralement opposées, dans l'étage postérieur.
  11. Décodeur suivant la revendication 10, caractérisé en ce que la disposition de haut-parleurs est sensiblement symétrique gauche/droite autour d'un axe ou plan avant, à travers la position d'écoute prédéterminée.
  12. Décodeur suivant la revendication 11, caractérisé en ce que la disposition de haut-parleurs comprend trois haut-parleurs disposés sur l'étage antérieur et deux haut-parleurs disposés sur un étage postérieur.
  13. Décodeur suivant la revendication 11, caractérisé en ce que la disposition de haut-parleurs comprend quatre haut-parleurs disposés sur l'étage antérieur et deux haut-parleurs disposés sur un étage postérieur.
  14. Décodeur suivant l'une quelconque des revendications précédentes, caractérisé en ce que les signaux audio codés de manière directionnelle comprennent des composantes de signal sonore qui représentent la pression sonore et des composantes de vitesse sonore directionnelles orthogonales.
  15. Décodeur suivant l'une quelconque des revendications précédentes, caractérisé en ce que les moyens matriciels sont agencés pour décoder des signaux audio codés de manière directionnelle comprenant au moins trois combinaisons indépendantes de façon linéaire d'un signal omnidirectionnel W avec gain uniforme pour toutes les directions, et au moins deux signaux directionnels X et Y pointant dans des directions perpendiculaires, représentant des sons codés avec des caractéristiques de gain directionnelles en cardioïde ou en cosinus.
  16. Décodeur suivant la revendication 15, caractérisé en ce que le signal de pression reproduit, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aWW + bWX de W et de X dont les proportions relatives aW/bW varient avec la fréquence, en ce que le signal reproduit de vitesse pointant en avant, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aXW + bXX de W et de X dont les proportions relatives ax/bx ne varient pas avec la fréquence, et en ce que le signal reproduit de vitesse pointant latéralement, à la position d'écoute prédéterminée, est proportionnel à Y à toutes les fréquences.
  17. Décodeur suivant l'une quelconque des revendications 1 à 13, caractérisé en ce que les moyens matriciels sont agencés pour décoder des signaux audio codés de manière directionnelle comprenant deux combinaisons linéaires complexes indépendantes d'un signal omnidirectionnel W, avec un gain uniforme pour toutes les directions et d'au moins deux signaux directionnels X et Y, pointant dans des directions perpendiculaires, représentant des sons codés avec des caractéristiques de gain directionnelles en cardioïde ou en cosinus.
  18. Décodeur suivant la revendication 17, caractérisé en ce que le coefficient de gain des moyens matriciels sont tels que le signal de pression reproduit, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aWW + bWX + jcWY de W, X et Y, dont les proportions relatives aW/bW varient avec la fréquence et, en ce que le signal reproduit, qui représente la vitesse pointant en avant, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aXW + bXX + jcXY de W, X et Y, et en ce que le signal reproduit qui représente la vitesse pointant latéralement, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire - jaYw - jbYX + cYY de W, X et Y, les coefficients aw , bW, cW, aX , bX , cX, aY , bY et cY étant réels et j = □ (- 1) représentant une différence de phase de 90° par rapport à une large bande passante.
  19. Décodeur suivant la revendication 17 ou 18, caractérisé en ce que les moyens matriciels comprennent de plus des moyens matriciels de phase-amplitude agencés pour produire au moins trois combinaisons linéaires complexes W2 , X2 , Y2 et, de préférence ou en variante, une quatrième combinaison linéaire B2 de deux signaux d'entrée codés de manière directionnelle, de façon à ce que W2 et X2 aient un gain directionnel de la forme de a2 + b2X + c2jY pour des gains réels a2 , b2 et c2 qui peuvent être différents de W2 et X2 , et en ce que Y2 et B2 sont respectivement proportionnels à jX2 et à jW2 ou à des combinaisons linéaires réelles de ceux-ci, et en ce que les signaux sont fournis par des moyens d'équilibrage, avec des réponses de phase assorties, à au moins deux moyens matriciels d'amplitude qui correspondent à des plages de fréquences différentes dans la bande audio afin de procurer à la sortie du décodeur des signaux audio modifiés.
  20. Décodeur suivant l'une quelconque des revendications 15 à 19, caractérisé en ce que les moyens matriciels comprennent en outre des moyens matriciels linéaires supplémentaires, agencés pour appliquer une transformation linéaire supplémentaire de façon à ce que les directions de vecteurs reproduites délivrées soient en relation avec les directions codées d'entrée, suivant une transformation de direction.
  21. Décodeur suivant la revendication 20, caractérisé en ce que la transformation de directions est une transformation de Lorentz.
  22. Décodeur suivant la revendication 20 ou 21, caractérisé en ce que l'effet de la transformation matricielle supplémentaire consiste à rendre sensiblement égaux tous les gains d'énergie reproduite de sons codés à l'avant et à l'arrière.
  23. Décodeur suivant la revendication 20 ou 21 ou 22, caractérisé en ce que la transformation matricielle linéaire supplémentaire est mise en oeuvre sous la forme d'une matrice linéaire agissant sur les signaux codés de manière directionnelle ou sur des combinaisons linéaires de ceux-ci.
  24. Décodeur suivant la revendication 20 ou 21 ou 22, caractérisé en ce que la transformation matricielle linaire supplémentaire est combinée aux moyens matriciels précités ou aux premier et second moyens matriciels.
  25. Décodeur suivant la revendication 15 ou 16, comportant au moins trois haut-parleurs sur un étage antérieur reproduit, caractérisé en ce que les signaux audio codés de manière directionnelle comprennent de plus des signaux proportionnels à E et /ou F,
    E ayant une caractéristique de gain directionnel sensiblement égale à zéro en dehors d'un étage antérieur codé de directions codées et un gain proportionnel à une combinaison linéaire de W et de X ayant un gain positif pour des sons au centre de l'étage antérieur codé, sur un étage antérieur de directions codées, et
    F ayant un gain sensiblement proportionnel à celui de Y sur un étage antérieur de directions codées et un gain sensiblement proportionnel à celui de - Y sur un étage postérieur de directions codées, et
    le codeur comprenant des moyens pour ajouter E aux, et soustraire E des, composantes de signal contenant W et X, de façon à localiser plus précisément, dans des haut-parleurs séparés d'étage antérieur, des sons d'étage antérieur codés, et/ou des moyens pour ajouter F à des, et soustraire F de, composantes de signal contenant Y, de façon à réduire de la diaphonie entre des étages de sons antérieurs et postérieurs reproduits.
  26. Décodeur suivant la revendication 24, caractérisé en ce que E a un gain de polarité opposée pour des sons aux bords de l'étage antérieur codé plutôt que pour des sons codés vers le centre de l'étage antérieur codé.
  27. Système audio comprenant :
    un décodeur,
    une multiplicité de haut-parleurs disposés autour d'une zone d'écoute, et
    un amplificateur pour amplifier le signal de sortie du décodeur afin de commander les haut-parleurs,
    le décodeur décodant des signaux audio codés de manière directionnelle, en vue d'une reproduction par une disposition de haut-parleurs sur la zone d'écoute,
       le décodeur comprenant :
    une entrée pour recevoir les signaux audio codés de manière directionnelle,
    des moyens matriciels pour modifier les signaux audio précités, et
    une sortie pour délivrer le signal audio modifié, dans une forme appropriée pour une reproduction par les haut-parleurs,
    les coefficients des moyens matriciels étant tels qu'à une position d'écoute prédéterminée, dans la zone d'écoute, la direction du vecteur de vitesse reproduite et les directions de vecteurs d'énergie reproduites sont sensiblement égales l'une à l'autre et sensiblement indépendantes de la fréquence dans une large plage de fréquences audio,
       caractérisé en ce que les coefficients de gain des moyens matriciels sont tels que la valeur du vecteur de vitesse rV reproduite d'un signal audio décodé varie sensiblement avec la direction de son codée, à des fréquences dans la gamme et au-dessus d'une fréquence audio moyenne prédéterminée.
  28. Système suivant la revendication 27, caractérisé en ce que la disposition de haut-parleurs comprend un nombre supérieur de haut-parleurs de reproduction sur un étage antérieur de directions et un nombre inférieur de haut-parleurs sur un étage postérieur diamétralement opposé de directions, et en ce que le coefficient de gain des moyens matriciels du décodeur sont tels qu'à sensiblement toutes les fréquences, la valeur du vecteur d'énergie reproduite rE de sons codés pour être reproduits à partir de directions de vecteurs dans l'étage antérieur est nettement supérieure à la valeur du vecteur d'énergie reproduite rE de sons codés pour être reproduits dans l'étage postérieur, à partir de directions de vecteurs diamétralement opposées.
  29. Système suivant la revendication 28, caractérisé en ce que la disposition de haut-parleurs est sensiblement symétrique gauche/droite autour d'un axe ou plan avant, à travers la position d'écoute prédéterminée.
  30. Système suivant la revendication 29, caractérisé en ce que la disposition de haut-parleurs comprend trois haut-parleurs disposés sur l'étage antérieur et deux haut-parleurs disposés sur un étage postérieur.
  31. Système suivant la revendication 29, caractérisé en ce que la disposition de haut-parleurs comprend quatre haut-parleurs disposés sur l'étage antérieur et deux haut-parleurs disposés sur un étage postérieur.
  32. Système audiovisuel comprenant dans ses étages audio un décodeur suivant l'une quelconque des revendications 1 à 26 ou un système audio suivant l'une quelconque des revendications 27 à 31.
  33. Procédé de décodage de signaux audio codés de manière directionnelle, en vue d'une reproduction par l'intermédiaire d'une disposition de haut-parleurs sur une zone d'écoute, comprenant une application du signal audio codé à des moyens matriciels agencés pour décoder le signal, et
    une sortie du signal dans une forme appropriée pour une reproduction subséquente par les haut-parleurs,
    le coefficient des moyens matriciels étant tel qu'à une position d'écoute prédéterminée, dans la zone d'écoute, la direction du vecteur de vitesse reproduite et la direction du vecteur d'énergie reproduite sont sensiblement égales l'une à l'autre et sensiblement indépendantes de la fréquence dans une large plage de fréquences audio,
       caractérisé en ce que la valeur du vecteur de vitesse reproduite rV d'un signal audio décodé varie systématiquement avec la direction de son codée, à des fréquences dans gamme, et au-dessus, d'une fréquence audio moyenne prédéterminée.
  34. Procédé suivant la revendication 33, caractérisé en ce que
    des fréquences audio basses du signal audio codé, en dessous d'une fréquence d'équilibrage prédéterminée, sont décodées par un premier moyen matriciel et en ce que des fréquences audio élevées, au-dessus de la fréquence d'équilibrage, sont décodées par un second moyen matriciel différent en résultat du premier moyen matriciel,
    la large plage de fréquence audio dans laquelle la direction du vecteur de vitesse reproduite et la direction du vecteur d'énergie reproduite sont sensiblement égales l'une à l'autre et sensiblement indépendantes d'une fréquence entoure la fréquence d'équilibrage, et
    la valeur du vecteur de vitesse reproduite rv varie sensiblement avec la direction de sons codés, à des fréquences dans la gamme et au-dessus de la fréquence d'équilibrage.
  35. Procédé de codage et de décodage d'un signal audio, caractérisé en ce que le signal audio est codé en tant qu'au moins trois combinaisons indépendantes de façon linéaire d'un signal omnidirectionnel W, avec un gain uniforme pour toutes les directions, et de deux signaux directionnels X et Y pointant dans des directions perpendiculaires, les signaux X et Y ayant des caractéristiques de gain directionnel en cardioïde ou en cosinus, et en ce que le signal est décodé subséquemment par un procédé suivant la revendication 33 ou 34.
  36. Procédé de codage et de décodage d'un signal audio suivant la revendication 35, caractérisé en ce que le signal de pression reproduit, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aWW + bWX de W et de X dont les proportions relatives aW/bW varient avec la fréquence, en ce que le signal reproduit de vitesse pointant vers l'avant, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aXW + bXX de W et de X dont les proportions relatives à aX/bX ne varient pas avec la fréquence, et en ce que le signal reproduit de vitesse pointant latéralement, à la position d'écoute prédéterminée, est proportionnel à Y à toutes les fréquences.
  37. Procédé de codage et de décodage d'un signal audio caractérisé en ce que le signal audio est codé en tant que deux combinaisons linéaires complexes indépendantes d'un signal omnidirectionnel W, avec un gain uniforme pour toutes les directions, et d'au moins deux signaux directionnels X et Y pointant dans des directions perpendiculaires et représentant des sons codés avec des caractéristiques de gain directionnel en cardioïde ou en cosinus, et en ce que le signal est décodé subséquemment par un procédé suivant la revendication 33 ou 34.
  38. Procédé de codage et de décodage suivant la revendication 37, caractérisé en ce que le signal de pression reproduit, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aWW + bWX + jcWY, de W, X et Y, dont les proportions relatives aw/bw varient avec la fréquence et, en ce que le signal reproduit, qui représente la vitesse pointant en avant, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire aXW + bXX + jcXY de W, X et Y et en ce que le signal reproduit qui représente la vitesse pointant latéralement, à la position d'écoute prédéterminée, est à toutes les fréquences une combinaison linéaire - jaYW - jbYX + cYY de W, X et Y, les coefficients aW , bW , cW , aX , bX , cX, aY, bY et cY étant réels et pouvant être dépendants de la fréquence et j = □ (- 1) représentant une différence de phase de 90° par rapport à une large bande passante.
EP93904278A 1992-03-02 1993-03-02 Appareil de reproduction de son d'ambiance Expired - Lifetime EP0629335B1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
GB9204485 1992-03-02
GB929204485A GB9204485D0 (en) 1992-03-02 1992-03-02 Surround sound apparatus
PCT/GB1993/000429 WO1993018630A1 (fr) 1992-03-02 1993-03-02 Appareil de reproduction de son d'ambiance

Publications (2)

Publication Number Publication Date
EP0629335A1 EP0629335A1 (fr) 1994-12-21
EP0629335B1 true EP0629335B1 (fr) 1999-05-12

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DE (1) DE69324912D1 (fr)
GB (1) GB9204485D0 (fr)
WO (1) WO1993018630A1 (fr)

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IT1283803B1 (it) * 1996-08-13 1998-04-30 Luca Gubert Finsterle Sistema di registrazione dei suoni a due canali e sistema di riproduzione dei suoni tramite almeno quattro diffusori con
US6711266B1 (en) * 1997-02-07 2004-03-23 Bose Corporation Surround sound channel encoding and decoding
US6072878A (en) * 1997-09-24 2000-06-06 Sonic Solutions Multi-channel surround sound mastering and reproduction techniques that preserve spatial harmonics
EP0905933A3 (fr) * 1997-09-24 2004-03-24 STUDER Professional Audio AG Méthode et système pour mélanger des signaux audio
US7428440B2 (en) 2002-04-23 2008-09-23 Realnetworks, Inc. Method and apparatus for preserving matrix surround information in encoded audio/video
WO2003092260A2 (fr) * 2002-04-23 2003-11-06 Realnetworks, Inc. Procede et appareil destines a preserver des informations d'ambiance sonore au moyen d'une matrice en mode audio/video code
US7391869B2 (en) * 2002-05-03 2008-06-24 Harman International Industries, Incorporated Base management systems
FI118247B (fi) 2003-02-26 2007-08-31 Fraunhofer Ges Forschung Menetelmä luonnollisen tai modifioidun tilavaikutelman aikaansaamiseksi monikanavakuuntelussa
DE10355146A1 (de) * 2003-11-26 2005-07-07 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Erzeugen eines Tieftonkanals
EP2205007B1 (fr) * 2008-12-30 2019-01-09 Dolby International AB Procédé et appareil pour le codage tridimensionnel de champ acoustique et la reconstruction optimale
GB2467534B (en) * 2009-02-04 2014-12-24 Richard Furse Sound system
US9736609B2 (en) 2013-02-07 2017-08-15 Qualcomm Incorporated Determining renderers for spherical harmonic coefficients
EP2782094A1 (fr) 2013-03-22 2014-09-24 Thomson Licensing Procédé et appareil permettant d'améliorer la directivité d'un signal ambisonique de 1er ordre
EP2866475A1 (fr) 2013-10-23 2015-04-29 Thomson Licensing Procédé et appareil pour décoder une représentation du champ acoustique audio pour lecture audio utilisant des configurations 2D
US20150264483A1 (en) * 2014-03-14 2015-09-17 Qualcomm Incorporated Low frequency rendering of higher-order ambisonic audio data
US9338552B2 (en) 2014-05-09 2016-05-10 Trifield Ip, Llc Coinciding low and high frequency localization panning

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US5172415A (en) * 1990-06-08 1992-12-15 Fosgate James W Surround processor

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EP0629335A1 (fr) 1994-12-21
GB9204485D0 (en) 1992-04-15
WO1993018630A1 (fr) 1993-09-16
DE69324912D1 (de) 1999-06-17

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