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WO2011069205A1 - Décodeur matriciel amélioré pour son ambiophonique - Google Patents

Décodeur matriciel amélioré pour son ambiophonique Download PDF

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Publication number
WO2011069205A1
WO2011069205A1 PCT/AU2010/001666 AU2010001666W WO2011069205A1 WO 2011069205 A1 WO2011069205 A1 WO 2011069205A1 AU 2010001666 W AU2010001666 W AU 2010001666W WO 2011069205 A1 WO2011069205 A1 WO 2011069205A1
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WO
WIPO (PCT)
Prior art keywords
signals
encoded
channel signals
channel
decoder
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Ceased
Application number
PCT/AU2010/001666
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English (en)
Inventor
Charlie Van Dongen
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Reality IP Pty Ltd
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Reality IP Pty Ltd
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Priority claimed from AU2009906030A external-priority patent/AU2009906030A0/en
Application filed by Reality IP Pty Ltd filed Critical Reality IP Pty Ltd
Priority to US13/379,065 priority Critical patent/US9111528B2/en
Priority to EP10835305.3A priority patent/EP2510709A4/fr
Publication of WO2011069205A1 publication Critical patent/WO2011069205A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • 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

Definitions

  • the present invention relates to an improved matrix decoder for surround sound.
  • the matrix decoder may be associated with a surround sound system wherein at least four audio input signals representing an original sound field are encoded into two channels and the two channels are decoded into at least four channels corresponding to the four audio input signals.
  • a multi-channel system as described above four channels of audio signals are obtained from an original sound field and are encoded by an encoder into two channels.
  • the encoded two channels may be recorded on recording media such as CD, DVD or the like or broadcast via stereo TV or FM radio.
  • the encoded two channels may be reproduced from the recording media or broadcast and decoded by means of a matrix decoder back into four channels approximating the four channels of audio signals obtained from the original sound field.
  • the decoded signals may be applied to four speakers to reproduce the original sound field through suitable amplifiers.
  • the decoder may not be possible for the decoder to reproduce signals that are identical to the original four audio signals.
  • cross-talk between adjacent channels may increase so that it may not be possible to obtain a reproduced sound field that is identical to the original sound field.
  • the present invention may provide a matrix decoder having improved separation between respective channels including between front and rear channels and between left and right channels.
  • the present invention may provide a matrix decoder capable of alleviating cross-talk between the respective channels to thereby improve the quality of the reproduced sound field.
  • the present invention may provide a matrix decoder capable of improving image stability in the reproduced sound field.
  • a decoder for use in a surround sound system wherein at least four audio input signals (FL, FR, RL, RR) representing an original sound field are encoded into two channel signals (L, R) and said encoded two channel signals are decoded into at least four audio output signals (FL', FR', RL', RR') corresponding to said four audio input signals, said encoded two channel signals having an amplitude ratio and a phase relationship, said decoder including:
  • first filter means connected to receive said encoded two channel signals for compensating for variations in perceived loudness relative to frequency associated with said encoded two channel signals due to non linearity in human hearing response at least at some frequencies;
  • control unit responsive to the phase relationship of said compensated two channel signals for producing steering signals
  • matrix means connected to receive said compensated two channel signals for decoding said encoded two channel signals to produce said audio output signals corresponding to said audio input signals, said matrix means including means responsive to said steering signals for varying at least the amplitude ratio of said encoded two channel signals contained in each of said output signals.
  • the first filter means may include an equal loudness weighting contour.
  • the first filter means may include an ITU-R 468 weighting contour and/or a pink noise contour.
  • the first filter means may include an A- weighting or Fletcher- Munson contour.
  • the decoder may include an RMS detector connected to receive the two channel signals for determining a root mean square (RMS) value associated with the two channel signals.
  • the RMS detector may include means for applying a first attack time constant and a second decay time constant in determining the RMS value.
  • the first attack time constant may be substantially faster than the second decay time constant.
  • the decoder may include a second filter means connected to receive the two channel signals for adjusting amplitude of the signals to correct for logarithmic sensitivity of human hearing response.
  • a decoding method for use in a surround sound system wherein at least four audio input signals (FL, FR, RL, RR) representing an original sound field are encoded into two channel signals (L, R) and said encoded two channel signals are decoded into at least four audio output signals (FL', FR', RL', RR') corresponding to said four audio input signals, said encoded two channel signals having an amplitude ratio and a phase relationship, said method including:
  • FIG. 1 is a block diagram showing principles of a "4-2-4" matrix system
  • FIG. 2 shows a configuration of an encoder
  • FIG. 3 shows a block diagram of a decoder according to the present invention
  • FIG. 4 shows a block diagram of front-back steering logic associated with a decoder
  • FIG. 5 shows a block diagram of left-right steering logic associated with a decoder
  • FIG. 6 shows a block diagram of a multi-band decoder according to the present invention
  • FIG. 7 shows a circuit diagram a matrix decoder according to one embodiment of the present invention.
  • FIGS 8A to 8D show examples of equal loudness response curves associated with the first filter means.
  • each microphone 10 1 1 , 12 and 13 are installed in an original sound field 14 in order to produce four channel audio signals FL (front-left), FR (front-right), RL (rear-left) and RR (rear-right) respectively.
  • An optional centre channel may also be produced.
  • the four channel audio signals are supplied to encoder 15 to be transformed or encoded into two signals L and R.
  • the outputs L and R from encoder 15 are applied to a decoder 16 to be transformed or decoded into reproduced four channel signals FL', FR', RL' and RR' approximating the original four channel signals FL, FR, RL and RR.
  • Decoder 16 may include single or multi-band processing as described below.
  • the reproduced four channel signals may be applied through amplifiers (not shown) to four loud speakers 17, 18, 19 and 20 located in a listening space 21 to provide a multi-channel sound field that more closely approximates the original sound field 14 when compared to a prior art two channel system.
  • a variety of two channel systems 22 including CD, DVD, TV, FM radio, etc. may be used to capture or store outputs L and R from encoder 15 and to supply the captured or stored outputs to decoder 16.
  • outputs L and R from encoder 15 may be recorded on a storage medium such as a CD, DVD or magnetic tape and the outputs from the storage medium may be applied to decoder 16.
  • the outputs L and R from encoder 15 or the outputs reproduced from the recording medium may be transmitted to decoder 16 via a stereo TV or an FM stereo radio broadcasting system.
  • Encoder 15 may include any conventional or known encoder including Q sound, Prologic or conventional stereo.
  • encoder 15 shown in FIG. 1 may be configured as shown in FIG. 2 wherein audio signals FL and FR produced by microphones 10 and 1 1 disposed in the front of original sound field 14 and audio signals RL and RR produced by microphones 12, 13 disposed in the rear of original sound field 14 are applied to a matrix circuit 23.
  • Matrix circuit 23 includes a plurality of adders/multipliers and phase shifters arranged to produce L and R output signals as follows:
  • R FR + kFL - jRR - jkRL
  • k denotes a transformation or matrix constant generally having a value approximately 0.414
  • j denotes a 90 degree phase shift.
  • the phase shifters may provide a substantially consistent phase shift over the entire audio frequency band.
  • FIG. 3 shows a block diagram of an improved decoder including a variable matrix 24 having control unit 25 and decoder unit 26 and employing matrix coefficients SL, SR, S F , S B the magnitudes of which may be controlled in accordance with the phase difference between two channel signals L and R.
  • the two channel signals L and R are applied to input terminals 27 and 28 of the decoder from a two-channel media source and hence to input terminals 29 and 30 of variable matrix 24.
  • Input terminals 27 and 28 are also coupled to input terminals 31 and 32 of variable matrix 24 via 90 degree phase shift circuit 33.
  • Variable matrix 24 operates to decode or dematrix the two channel signals L and R to produce four channel signals at its output terminals 34, 35, 36 and 37.
  • Control unit 25 provides steering control signals S L , S R , S F , and S B to decoder unit 26 in accordance with the phase difference between two-channel signals L and R.
  • control signal S F may be used to control the matrix coefficient related to the front channels and control signal SB may be used to control the matrix coefficient related to the rear channels.
  • control signal SR may be used to control the matrix coefficient related to the right channels and control signal SL may be used to control the matrix coefficient related to the left channels.
  • control signal SF operates to decrease the matrix coefficient related to the front channels thus enhancing separation between the front channels.
  • control signal SB operates to increase the matrix coefficient related to the rear channels to reduce separation between rear channels. Concurrently therewith signal levels of the front channels may be increased and those of the rear channels may be decreased to improve separation between the front and rear channels.
  • the control unit 25 may include a phase discriminator for detecting a phase difference between signals L and R or a comparator for detecting a phase relationship between signals L and R in terms of the difference in the levels of a sum signal (L+R) and a difference signal (L-R).
  • a reason for controlling the matrix coefficient associated with the front and rear channels by detecting the phase relationship between signals L and R is that humans have a keen sensitivity to detect the direction of a large sound but sensitivity for a small sound coexisting with the large sound may be relatively poor. Consequently, where there is a large sound in the front and a small sound in the rear playback of four channels may be more efficient if separation between the front channels is enhanced and separation between the rear channels is reduced.
  • Fig. 4 is a block diagram of a steering logic circuit for producing front/back steering values SF, SB.
  • the steering logic circuit includes an equal loudness weighting filter 40 such as a modified Fletcher Munson/A-weighting or ITU-R 468 filter for providing compensation for variations in perceived loudness relative to frequency due to non linearity in human hearing response at least at some frequencies.
  • the equal loudness weighting filter may be modified to include a characteristic similar to a pink noise (1 /f) weighting at low frequencies, to further attenuate high amplitude low audibility sounds that may otherwise unduly influence the steering logic circuit.
  • A-weighting filters are sometimes used for the purpose of compensation.
  • a pink noise filter is preferred for music content over an A-weighting filter because the latter is mainly valid for pure tones and relatively quiet sounds.
  • Pink noise is also known as 1/f noise, wherein power spectral density is inversely proportional to frequency.
  • a pink noise contour gives greater attenuation at low frequencies than a Fletcher Munson/A-weighting or ITU-R 468 weighting filter based on the fact that for equal power, amplitude is inversely proportional to frequency.
  • Use of a pink noise contour may further reduce dominance of low frequency sounds (high amplitude but low audibility) in calculating steering logic values, which are based on amplitude, and results in better placement of sound information that may be important for correct image generation.
  • the steering logic circuit includes a mixer/comparator 41 for adding the compensated channel signals L and R to produce a sum signal (L+R) 42 and for subtracting the two channel signals L and R to produce a difference signal (L-R) 43.
  • the sum and difference signals 42, 43 are applied to RMS detector 44.
  • RMS detector 44 is adapted to compensate for the peak nature of music content.
  • the averaging time constant over which RMS detector 44 measures a 'mean' value of a music signal preferably includes a first or 'attack' time constant and a second or 'decay' time constant.
  • the 'attack' time constant may be substantially faster than the 'decay' time constant.
  • the attack time constant may be 20mS and the decay time constant may be 50mS for a full range RMS detector.
  • an RMS detector including a single time constant may be used.
  • RMS detected outputs 45, 46 are applied to logarithmic amplifier 47 to produce outputs 48, 49 proportional to log
  • Logarithmic amplifier 47 is adapted to correct for logarithmic sensitivity of human hearing response to sound that spans a range of signal amplitudes or levels.
  • Comparator 50 may produce at its outputs 51 , 52 front and back steering factors SF, SB that hinge in a complementary and linear fashion around the centre value 0.707 representing OdB difference between signals 48 and 49.
  • Fig. 5 is a block diagram of a steering logic circuit producing left/right steering values S L , SR.
  • the steering logic circuit includes an equal loudness weighting filter 60 such as a modified Fletcher Munson/A weighting or ITU-R 468 filter. Weighting filter 60 may be similar to weighting filter 40 and may be adapted to compensate for non linearity in human hearing response as described above.
  • the steering logic circuit includes a RMS detector 61 .
  • RMS detector 61 may be similar to RMS detector 44 and may be adapted to compensate for the peak nature of music content as described above.
  • RMS detected outputs 62, 63 are applied to logarithmic amplifier 64 to produce outputs 65, 66 proportional to log
  • Logarithmic amplifier 64 may be similar to logarithmic amplifier 47 described above and is adapted to correct for logarithmic sensitivity of human hearing response to sound that spans a range of signal amplitudes or levels.
  • Comparator 67 may produce at its outputs 68, 69 left and right steering factors S L , S R that hinge in a complementary and linear fashion around the centre value 0.707 representing OdB difference between signals 65 and 66.
  • the encoder of the present invention may include a multi-band modification as shown in Figure 6.
  • Fig. 6 shows a multi-band decoder wherein the audible spectrum may be split into 3 separate bands via band splitter 70.
  • the bands include a low frequency band A below 300 Hz, a mid-frequency band B between 300- 3KHz and a high frequency band C above 3KHz.
  • Band splitter 70 may be interposed between 90 degree phase shift circuit 33 (refer Fig. 3) and variable matrix decoder 24.
  • a separate matrix decoder 24A, 24B, 24C may be used to produce a set of four channel output signals FU, FR', RU and RR' for each frequency band A, B, C.
  • the four channel output signals for each band may be subsequently combined via band mixer 71 .
  • the output FU may be obtained by combining contributions FL'A, FL'B and FL'C produced by matrix decoders 24A, 24B and 24C respectively.
  • the attack time constant may be 30mS and the decay time constant may be 60mS for band A.
  • the attack time constant may be 10mS and the decay time constant may be 30mS for band B.
  • the attack time constant may be 1 mS and the decay time constant may be 5mS for band C.
  • matrix decoders 24A, 24B and 24C may be similarly combined to produce full band decoded outputs FU, FR', RU and RR' for the multi band decoder at its output terminals 72, 73, 74, 75 respectively.
  • Fig. 7 shows a circuit diagram of a matrix decoder including a steering logic circuit 80 for producing front/back steering values S F , S B , a steering logic circuit 81 for producing left/right steering logic and matrix circuits 82 to 85.
  • Steering logic circuit 80 includes an equal loudness weighting filter 40 such as a modified Fletcher Munson filter, comparator 41 , RMS detector 44, logarithmic amplifier 47 and comparator 50 as described above with reference to Fig. 4.
  • Comparator 41 includes parts 41 a, 41 b for producing difference (L-R) and sum (L+R) signals respectively as described above.
  • RMS detector 44 has dual time constants and includes parts 44a, 44b for RMS detecting the difference and sum signals respectively.
  • Logarithmic amplifier 47 includes parts 47a, 47b for correcting the RMS detected difference and sum signals respectively.
  • Comparator 50 includes parts 50a, 50b and 50c for comparing the outputs of parts 47a, 47b and for applying a scaling factor to provide steering factor SF and for inverting the latter to provide steering factor SB.
  • Steering logic circuit 81 includes an equal loudness weighting filter 60 such as a modified Fletcher Munson filter, RMS detector 61 , logarithmic amplifier 64 and comparator 67 as described above with reference to Fig. 5.
  • RMS detector 61 has dual time constants and includes parts 61 a, 61 b for RMS detecting the left and right signals respectively.
  • Logarithmic amplifier 64 includes parts 64a, 64b for correcting the RMS detected left and right signals respectively.
  • Comparator 67 includes parts 67a, 67b and 67c for comparing the outputs of parts 64a, 64b and for applying a scaling factor to provide steering factor SR and for inverting the latter to provide steering factor S L .
  • Matrix circuit 82 includes difference amplifier 86, V2 scaler 87, multipliers 88, 89 and summing amplifier 90.
  • the output FU appearing at the output terminal of summing amplifier 90 and hence at the output of matrix circuit 82 is given by the following equation:
  • Matrix circuit 83 includes difference amplifier 91 , inverter 92, fl scaler 93, multipliers 94, 95 and summing amplifier 96.
  • the output FR' appearing at the output terminal of summing amplifier 96 and hence at the output of matrix circuit 83 is given by the following equation:
  • Matrix circuit 84 includes difference amplifier 97, 2 scaler 98, multipliers 99, 100 and summing amplifier 101 .
  • the output RU appearing at the output terminal of summing amplifier 101 and hence at the output of matrix circuit 84 is given by the following equation:
  • Matrix circuit 85 includes difference amplifier 102, V2 scaler 103, multipliers 104, 105 and summing amplifier 106.
  • the output RR' appearing at the output terminal of summing amplifier 106 and hence at the output of matrix circuit 85 is given by the following equation:
  • Equal loudness weighting filters 40, 60 may include a modified Fletcher Munson - pink noise weighting filter including an ITU-R 468 weighting contour.
  • Weighting filters 40, 60 may be implemented in any suitable manner and by any suitable means.
  • the response of weighting filters 40, 60 may include a frequency response contour as shown in Fig 8D for a single band implementation.
  • the response of weighting filters 40, 60 may include frequency response contours as shown in Figs. 8A to 8C for low band A, mid band B and high band C respectively.
  • RMS detectors 44, 61 may be implemented in any suitable manner and by any suitable means. In one form RMS detectors 44, 61 may be implemented on a digital sound processor such as a Texas Instruments TAS 3108 via Pure Path Studio Software.
  • a matrix decoder as described herein may be applied to a surround sound system utilizing more than four audio input signals to represent an original sound field.
  • a pair of decoders as described herein may be applied to encode eight audio input signals representing an original sound field into four channel signals and the encoded four channel signals may be decoded into eight audio output signals.
  • Such decoders may be applied to an installation including four pairs of loudspeakers or speaker arrays wherein each loudspeaker or speaker array is arranged at a respective corner of a cube or a rectangular cuboid to define upper and lower planes of four loudspeakers or speaker arrays each, namely four loudspeakers or speaker arrays in the front and four loudspeakers or speaker arrays in the back.
  • the upper plane of loudspeakers or speaker arrays may be vertically separated relative to the lower plane of loudspeakers or speaker arrays by approximately 2-3m or other suitable distance depending on usable height in an associated listening zone or auditorium.
  • the encoded four channel signals may be recorded on suitable media such as DVD, BluRay disc or the like and/or broadcast via a HDTV transmission service such as Foxtel that is capable of transmitting at least four channels of audio signals.
  • suitable media such as DVD, BluRay disc or the like and/or broadcast via a HDTV transmission service such as Foxtel that is capable of transmitting at least four channels of audio signals.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Mathematical Physics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Stereophonic System (AREA)

Abstract

L'invention concerne un décodeur et un procédé de décodage, destinés à être utilisés dans un système de son ambiophonique, dans lequel au moins quatre signaux audio d'entrée représentant un champ sonore original sont codés en deux signaux de canal qui sont décodés en au moins quatre signaux audio de sortie correspondant aux quatre signaux audio d'entrée et ont un rapport d'amplitude et une relation de phase. Le décodeur et le procédé comprennent des moyens destinés à : compenser lesdits signaux codés pour les variations d'intensité sonore perçue par rapport à une fréquence associée aux deux signaux de canal codés, en raison de la non-linéarité de la réponse auditive humaine au moins à certaines fréquences ; produire des signaux d'orientation en réponse à la relation de phase desdits signaux compensés ; décoder lesdits signaux codés pour produire des signaux audio de sortie correspondant aux signaux audio d'entrée par variation au moins du rapport d'amplitude desdits signaux codés contenus dans chacun des signaux de sortie, en réponse auxdits signaux d'orientation.
PCT/AU2010/001666 2009-12-10 2010-12-09 Décodeur matriciel amélioré pour son ambiophonique Ceased WO2011069205A1 (fr)

Priority Applications (2)

Application Number Priority Date Filing Date Title
US13/379,065 US9111528B2 (en) 2009-12-10 2010-12-09 Matrix decoder for surround sound
EP10835305.3A EP2510709A4 (fr) 2009-12-10 2010-12-09 Décodeur matriciel amélioré pour son ambiophonique

Applications Claiming Priority (2)

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AU2009906030 2009-12-10
AU2009906030A AU2009906030A0 (en) 2009-12-10 Improved matrix decoder for surround sound

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Cited By (2)

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EP2903301A3 (fr) * 2014-01-29 2015-09-09 The Telos Alliance Amélioration d'au moins un des paramètres, intelligibilité ou volume sonore, d'un programme audio
US11838743B2 (en) 2018-12-07 2023-12-05 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using diffuse compensation

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WO2014015914A1 (fr) * 2012-07-27 2014-01-30 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Appareil et procédé destinés à fournir une description de système de microphone à enceinte de haut-parleur
EP2879131A1 (fr) * 2013-11-27 2015-06-03 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Décodeur, codeur et procédé pour estimation de sons informée des systèmes de codage audio à base d'objets

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US5727068A (en) * 1996-03-01 1998-03-10 Cinema Group, Ltd. Matrix decoding method and apparatus
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Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2903301A3 (fr) * 2014-01-29 2015-09-09 The Telos Alliance Amélioration d'au moins un des paramètres, intelligibilité ou volume sonore, d'un programme audio
US9344825B2 (en) 2014-01-29 2016-05-17 Tls Corp. At least one of intelligibility or loudness of an audio program
US11838743B2 (en) 2018-12-07 2023-12-05 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using diffuse compensation
US11856389B2 (en) 2018-12-07 2023-12-26 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using direct component compensation
US11937075B2 (en) 2018-12-07 2024-03-19 Fraunhofer-Gesellschaft Zur Förderung Der Angewand Forschung E.V Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using low-order, mid-order and high-order components generators
US12369008B2 (en) 2018-12-07 2025-07-22 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using low-order, mid-order and high-order components generators
US12418768B2 (en) 2018-12-07 2025-09-16 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method and computer program for encoding, decoding, scene processing and other procedures related to DirAC based spatial audio coding using diffuse compensation

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US9111528B2 (en) 2015-08-18
EP2510709A4 (fr) 2015-04-08
EP2510709A1 (fr) 2012-10-17
US20120163607A1 (en) 2012-06-28

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