US005912976A Ulllted States Patent   Patent Number: Klayman et al.   MULTI-CHANNEL AUDIO ENHANCEMENT SYSTEM FOR USE IN RECORDING AND PLAYBACK AND METHODS FOR PROVIDING SAME 5,912,976 Date of Patent: Jun. 15, 1999 OTHER PUBLICATIONS Schroeder, M.R., “An Arti?cial Stereophonic Effect Obtained from a Single Audio Signal”,Journal of theAua'io Engineering Society, vol. 6, No. 2, pp. 74—79, Apr. 1958. KuroZumi, K., et al., “A NeW Sound Image Broadening Control System Using a Correlation Coef?cient Variation  Inventors? Arnold I- Klaymalb Humingmn Beach; Alan D- Kraemel‘, Tllstln, both of Cahf Method”, Electronics and Communications in Japan, vol. _ _ _ 67—A, No. 3, pp. 204—211, Mar. 1984.  Asslgnee' SRS Labs’ Inc" Irvlne’ Cahf' Sundberg, J., “The Acoustics of the Singing Voice”, The  Physics of Music, pp. 16—23, 1978. A 1 N ' 08/743 776 pp '  5 O" Filed; I C] 6 ’ Ishihara, M., “A NeW Analog Signal Processor For A Stereo Nov, 7, 1996 00 Enhancement System”, IEEE Transactions on Consumer Electronics, vol. 37, No. 4, pp. 806—813, Nov. 1991. %5g Ci ................................................... 3.81/18 331/1 Allison, R.’ “The Loudspeaker / Living Room System”,  Field - - of- Search ................................................. .................................. .. 381/1, 17, 18, H04R Audio) 381/19, 20, 22, 23, 307, 300, 27 18_22, “ PIOV~ . . ,, . _ Vaughan, D., HoW We Hear Direction ,Aua'io, pp. 51 55, Dec. 1983.  References Cited (List continued on next page.) U ' S ' PATENT DOCUMENTS 3,170,991 3,229,038 P rimar . y Examiner—l\/l1nsun . . Oh H arve y 2/1965 Glasgal . 1/1966 Richter. Attorney, Agent, or Firm—Knobbe, Martens, Olson & Bear LLP (List continued on neXt page.)  FOREIGN PATENT DOCUMENTS 0 0 0 O 097 320 367 354 982 A3 1/1984 270A2 6/1989 569 A2 10/1989 517 A2 2/1990 O 357 402 A2 3/1990 35 014 2 /1966 33 31 352 A1 3/1985 4029936 10/1940 43.12585 5/1943 58-144989 9/1983 59-27692 2/1984 61-33600 2/1986 61466696 10/1986 European European European European Pat~ PatPat' Pat‘ OffOff O?Off‘ ABSTRACT An audio enhancement system and method for use receives a group of multi-channel audio signals and provides a simulated surround sound environment through playback of only tWo output signals. The multi-channel audio signals comprise a pair of front signals intended for playback from a forWard sound stage and a pair of rear signals intended for ' European Pat. Off. . - . Germany ' Japan _ Japan _ playback from a rear sound stage. The front and rear signals d.? d. . b . b. f are mo .1 e in pairs y separating an am 1ent component 0 each pair of signals from a direct component and processing at least some of the components With a head-related transfer Japan Japan Japan Japan function. Processing of the individual audio signal compo nents is determined by an intended playback position of the corresponding original audio signals. The individual audio signal components are then selectively combined With the Finland. . . - _ g g?ggg? ' W0 87/O6O9O 10/1987 WIPO _ WO 94/16548 7/1994 WIPO . WO 96/34509 10/1996 WIPO . g original audio signals to form tWo enhanced output signals ' for generating a surround sound experience upon playback. 48 Claims, 10 Drawing Sheets i 1 MU —CHANNEL Aumo GNAL SOU RCE MULTl-CHANNEL MlXED OUTPUTS \MMERSION PROCESSOR BF cP Lp 41k 42 26y Bomb-H za’ RP POWER AMPUFlER - RECORDlNG DEVlCE 5,912,976 Page 2 US. PATENT DOCUMENTS 3,246,081 3,249,696 3,665,105 3,697,692 3,725,586 3,745,254 3,757,047 3,761,631 3,772,479 3,849,600 3,885,101 3,892,624 3,925,615 3,943,293 4,024,344 4,063,034 4,069,394 4,118,599 4,139,728 4,192,969 4,204,092 4,209,665 4,218,583 4,218,585 4,219,696 4,237,343 4,239,937 4,303,800 4,308,423 4,308,424 4,309,570 4,332,979 4,349,698 4,355,203 4,356,349 4,393,270 4,394,536 4,408,095 4,479,235 4,489,432 4,495,637 4,497,064 4,503,554 4,567,607 4,569,074 4/1966 5/1966 5/1972 10/1972 4/1973 7/1973 9/1973 9/1973 11/1973 11/1974 5/1975 7/1975 12/1975 3/1976 5/1977 12/1977 1/1978 10/1978 2/1979 3/1980 5/1980 6/1980 8/1980 8/1980 8/1980 12/1980 12/1980 12/1981 12/1981 12/1981 1/1982 6/1982 9/1982 10/1982 10/1982 7/1983 7/1983 10/1983 10/1984 12/1984 1/1985 1/1985 3/1985 1/1986 2/1986 Edwards . Van Sickle . Chowning . Ha?er . Iida . Ohta et al. . Ito et al. . Ito et al. . Hilbert . Ohshima . Ito et al. . Shimada . Nakano . Bailey . Dolby et al. . Peters . Doi et al. . Iwahara et al. . Haramoto et al. . Iwahara . Bruney . Iwahara . Poulo . Carver . Kogure et al. . Kurtin et al. . Kampmann . DeFreitas . Cohen . Bice, Jr. . 4,589,129 4,594,610 4,594,729 4,594,730 4,622,691 4,648,117 4,696,036 4,703,502 4,748,669 4,856,064 4,866,774 4,866,776 4,888,809 4,933,768 4,953,213 5,033,092 5,046,097 5,105,462 5,146,507 5,208,860 5,228,085 5,251,260 5,319,713 5,325,435 5,371,799 5,400,405 5,572,591 5,677,957 5,734,724 5,742,688 5,771,295 5,799,094 5/1986 6/1986 6/1986 6/1986 11/1986 3/1987 9/1987 10/1987 5/1988 8/1989 9/1989 9/1989 12/1989 6/1990 8/1990 7/1991 9/1991 4/1992 9/1992 5/1993 7/1993 10/1993 6/1994 6/1994 12/1994 3/1995 11/1996 10/1997 3/1998 4/1998 6/1998 8/1998 Blackmer et al. . Patel . Weingartner . Rosen . Tokumo et al. . Kunugi et al. . Julstrom . Kasai et al. . Klayman . Iwamatsu . Klayman . Kasai et al. . Knibbeler . Ishikawa ................................... .. 381/1 Tasaki et al. . Sadaie . Lowe et al. . Lowe et al. . Satoh et al. . Lowe et al. . Aylward . Gates . Waller, Jr. et al. . Date et al. . Lowe et al. . Petroff . NumaZu .................................... .. 381/1 Hulsebus .. Kinoshita . 381/17 381/17 Ogawa . 381/17 Waller .. 381/18 Mouri ...................................... .. 381/18 Carver . Fischer . OTHER PUBLICATIONS Iwahara . Cohen . Robinson . Stevens, S., et al, “Chapter 5: The Two—Earned Man”, Sound And Hearing, pp. 98—106 and 196, 1965. Van Den Berg . Eargle, J ., “Multichannel Stereo Matrix Systems: An Over Shima et al. . view”, Journal of the Audio Engineering Society, pp. Ariga et al. . 552—558 (no date listed). Gri?is . Wilson, Kim, “AC—3 Is Here! But Are You Ready To Pay The Price?”, Home Theater; pp. 60—65, Jun. 1995. Copy of International Search Report dated Mar. 10, 1998 Polk . Bruney . Polk . Bruney et al. . from corresponding PCT application. Kaufman, Richard 1., “Frequency Contouring For Image Polk . Enhancement”, Audio, pp. 34—39, Feb. 1985. Davis . U.S. Patent Jun. 15,1999 Sheet 1 0f 10 5,912,976 ' 10 ,5 / MULTI—CHANNEL AuDIO SIGNAL SOURCE A0A1A2A3A4A5A5A7 B c ,6, 20 K SIGNAL MIXER B c 22 I 24 \ MULTl-CHANNEL MIXED OUTPUTS V MULTl-CHANNEL AUDIO IMMERSION PROCESSOR BP 40 CP 4; LP RP 2i \ II B OUT O_d4___ COUTO—<————— 46‘ 28 r30 ' RECORDING ‘ ; _ A , DEVICE I /32 POWER AMPLIFIER LOIJT % % ROuT K35 U.S. Patent Jun. 15,1999 Sheet 2 0f 10 5,912,976 FIG. 2 ,/50 [.90 52 r ——————————— ~—*—————I Q 54| 56‘ | OASIS‘? é? SOURCE(S) I MULTl-CHANNEL \ | l I I I DIGITAL AUDIO DECODER B I C II I l l | I 55 II /50 I I I AUDIO IMMERSION I ' PROCESSOR I I l l ,_ I l____ ____ff2____ _€4___l BP cP I LP RP I DIGITAL/ANALOG I I CONVERTER fja BP CP LP “72 74 55 70 \ 8 I 4\ J I A II I POWER } AMPLIFIER CouT Low" ‘ ' RECORDING I BOUT 86, 50\ RP 80 82 1 ROUT I 32 / DEVICE U.S. Patent Jun. 15, 1999 \\ 6Em Sheet 4 0f 10 5,912,976 U.S. Patent Jun. 15,1999 Sheet 7 0f 10 501Al PD‘IOJ Cm: m xi 5,912,976 Ami; + w >wvl PIQE 5x5 U.S. Patent Jun. 15,1999 Sheet 8 0f 10 5,912,976 F/G.9 _ _ _ _ _ _ _ _ _ _ _ _ . ~ _ _ _ i _ _ _ _ _ _ _ _ i 2O 1 0O 1 0k FREQ(Hz) i i i i _ _ _ 20k U S Patent Jun. 15,1999 Sheet 9 0f 10 5,912,976 F/ G . I0 352 1 OO 1 0k 20k U.S. Patent Jun. 15,1999 Sheet 10 0f 10 5,912,976 gam a QN 5,912,976 1 2 MULTI-CHANNEL AUDIO ENHANCEMENT SYSTEM FOR USE IN RECORDING AND PLAYBACK AND METHODS FOR PROVIDING SAME duced on rear left and right speakers, one channel is used for a forWard center dialogue speaker, and one channel is used for loW-frequency and effects signals. Audio playback sys tems Which can accommodate the reproduction of all these six channels do not require that the signals be mixed into a FIELD OF THE INVENTION tWo channel format. HoWever, many playback systems, including today’s typical personal computer and tomorroW’s This invention relates generally to audio enhancement systems and methods for improving the realism and dra matic effects obtainable from tWo channel sound reproduc tion. More particularly, this invention relates to apparatus personal computer/television, may have only tWo channel playback capability (excluding center and subWoofer channels). Accordingly, the information present in addi tional audio signals, apart from that of the conventional and methods for enhancing multiple audio signals and stereo signals, like those found in an AC-3 recording, must mixing these audio signals into a tWo channel format for either be electronically discarded or mixed into a tWo reproduction in a conventional playback system. BACKGROUND OF THE INVENTION 15 Audio recording and playback systems can be character mixing method may be to simply combine all of the signals into a tWo-channel format While adjusting only the relative iZed by the number of individual channel or tracks used to input and/or play back a group of sounds. In a basic stereo recording system, tWo channels each connected to a micro phone may be used to record sounds detected from the gains of the mixed signals. Other techniques may apply frequency shaping, amplitude adjustments, time delays or phase shifts, or some combination of all of these, to an distinct microphone locations. Upon playback, the sounds recording by the tWo channels are typically reproduced through a pair of loudspeakers, With one loudspeaker repro ducing an individual channel. Providing tWo separate audio channels for recording permits individual processing of channel format. There are various techniques and methods for mixing multi-channel signals into a tWo channel format. A simple individual audio signal during the ?nal mixing process. The particular technique or techniques used may depend on the 25 these channels to achieve an intended effect upon playback. format and content of the individual audio signals as Well as the intended use of the ?nal tWo channel mix. For example, US. Pat. No. 4,393,270 issued to van den Berg discloses a method of processing electrical signals by modulating each individual signal corresponding to a pre selected direction of perception Which may compensate for placement of a loudspeaker. A separate multi-channel pro cessing system is disclosed in US. Pat. No. 5,438,623 issued to Begault. In Begault, individual audio signals are divided into tWo signals Which are each delayed and ?ltered accord ing to a head related transfer function (HRTF) for the left Similarly, providing more discrete audio channels alloWs more freedom in isolating certain sounds to enable the separate processing of these sounds. Professional audio studios use multiple channel record ings systems Which can isolate and process numerous indi vidual sounds. HoWever, since many conventional audio reproduction devices are delivered in traditional stereo, use of a multi-channel system to record sounds requires that the 35 and right ears. The resultant signals are then combined to sounds be “mixed” doWn to only tWo individual signals. In generate left and right output signals intended for playback the professional audio recording World, studios employ such through a set of headphones. mixing methods since individual instruments and vocals of a given audio Work may be initially recorded on separate tracks, but must be replayed in a stereo format found in The techniques found in the prior art, including those found in the professional recording arena, do not provide an effective method for mixing multi-channel signals into a tWo conventional stereo systems. Professional systems may use 48 or more separate audio channels Which are processed individually before recorded onto tWo stereo tracks. In multi-channel playback systems, i.e., de?ned herein as systems having more than tWo individual audio channels, 45 each sound recorded from an individual channel may be separately processed and played through a corresponding speaker or speakers. Thus, sounds Which are recorded from, or intended to be placed at, multiple locations about a listener, can be realistically reproduced through a dedicated speaker placed at the appropriate location. Such systems have found particular use in theaters and other audio-visual environments Where a captive and ?xed audience experi ences both an audio and visual presentation. These systems, Which include Dolby Laboratories’ “Dolby Digital” system; the Digital Theater System (DTS); and Sony’s Dynamic 55 channel format to achieve a realistic audio reproduction through a limited number of discrete channels. As a result, much of the ambiance information Which provides an immersive sense of sound perception may be lost or masked in the ?nal mixed recording. Despite numerous previous methods of processing multi-channel audio signals to achieve a realistic experience through conventional tWo channel playback, there is much room for improvement to achieve the goal of a realistic listening experience. Accordingly, it is an object of the present invention to provide an improved method of mixing multi-channel audio signals Which can be used in all aspects of recording and playback to provide an improved and realistic listening experience. It is an object of the present invention to provide an improved system and method for mastering professional audio recordings intended for playback on a conventional stereo system. It is also an object of the present invention to provide a system and method to process multi-channel audio Digital Sound (SDDS), are all designed to initially record and then reproduce multi-channel sounds to provide a sur round listening experience. In the personal computer and home theater arena, recorded media is being standardiZed so that multiple signals extracted from an audio-visual recording to provide channels, in addition to the tWo conventional stereo channels, are stored on such recorded media. One such a limited number of audio channels. standard is Dolby’s AC-3 multi-channel encoding standard Which provides six separate audio signals. In the Dolby AC-3 system, tWo audio channels are intended for playback on forWard left and right speakers, tWo channels are repro an immersive listening experience When reproduced through For example, personal computers and video players are emerging With the capability to record and reproduce digital 65 video disks (DVD) having six or more discrete audio channels. HoWever, since many such computers and video players do not have more than tWo audio playback channels 5,912,976 3 4 (and possibly one sub-Woofer channel), they cannot use the full amount of discrete audio channels as intended in a surround environment. Thus, there is a need in the art for a FIG. 5 is a perspective vieW of a personal computer having an audio enhancement system constructed in accor dance With the present invention for creating a surround computer and other video delivery system Which can effec tively use all of the audio information available in such sound effect from tWo output signals. FIG. 6 is a schematic block diagram of the personal systems and provide a tWo channel listening experience Which rivals multi-channel playback systems. The present thereof. computer of FIG. 5 depicting major internal components invention ful?lls this need. SUMMARY OF THE INVENTION 10 An audio enhancement system and method is disclosed for processing a group of audio signals, representing sounds FIG. 8 is a schematic block diagram of a preferred embodiment for processing and mixing a group of AC-3 audio signals to achieve a surround-sound experience from existing in a 360 degree sound ?eld, and combining the group of audio signals to create a pair of signals Which can accurately represent the 360 degree sound ?eld When played through a pair of speakers. The audio enhancement system FIG. 7 is a diagram depicting the perceived and actual origins of sounds heard by a listener during operation of the personal computer shoWn in FIG. 5. 15 a pair of output signals. FIG. 9 is a graphical representation of a ?rst signal equalization curve for use in a preferred embodiment for can be used as a professional recording system or in personal computers and other home audio systems Which include a processing and mixing a group of AC-3 audio signals to achieve a surround-sound experience from a pair of output limited amount of audio reproduction channels. signals. In a preferred embodiment for use in a home audio FIG. 10 is a graphical representation of a second signal reproduction system having stereo playback capability, a multi-channel recording provides multiple discrete audio equalization curve for use in a preferred embodiment for signals consisting of at least a pair of left and right signals, processing and mixing a group of AC-3 audio signals to achieve a surround-sound experience from a pair of output a pair of surround signals, and a center channel signal. The home audio system is con?gured With speakers for repro 25 signals. ducing tWo channels from a forWard sound stage. The left FIG. 11 is a schematic block diagram depicting the and right signals and the surround signals are ?rst processed and then mixed together to provide a pair of output signals for playback through the speakers. In particular, the left and right signals from the recording are processed collectively to provide a pair of spatially-corrected left and right signals to various ?lter and ampli?cation stages for creating the ?rst signal equalization curve of FIG. 9. FIG. 12 is a schematic block diagram depicting the various ?lter and ampli?cation stages for creating the second signal equalization curve of FIG. 10. enhance sounds perceived by a listener as emanating from a forward sound stage. The surround signals are collectively processed by ?rst isolating the ambient and monophonic components of the surround signals. The ambient and monophonic components DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 35 FIG. 1 depicts a block diagram of a ?rst preferred embodiment of a multi-channel audio enhancement system 10 for processing a group of audio signals and providing a of the surround signals are modi?ed to achieve a desired spatial effect and to separately correct for positioning of the playback speakers. When the surround signals are played through forWard speakers as part of the composite output pair of output signals. The audio enhancement system 10 comprises a source of multi-channel audio signal source 16 Which outputs a group of discrete audio signals 18 to a signals, the listener perceives the surround sounds as ema nating from across the entire rear sound stage. Finally, the multi-channel signal mixer 20. The mixer 20 provides a set of processed multi-channel outputs 22 to an audio immer sion processor 24. The signal processor 24 provides a center signal may also be processed and mixed With the left, right and surround signals, or may be directed to a center channel speaker of the home reproduction system if one is 45 channel signal 28 Which can be directed to a recording device 30 or to a poWer ampli?er 32 before reproduction by present. a pair of speakers 34 and 36. Depending upon the signal inputs 18 received by the processor 20, the signal mixer may also generate a bass audio signal 40 containing loW frequency information Which corresponds to a bass signal, BRIEF DESCRIPTION OF THE DRAWINGS The above and other aspects, features, and advantages of the present invention Will be more apparent from the fol loWing particular description thereof presented in conjunc B, from the signal source 16, and/or a center audio signal 42 containing dialogue or other centrally located sounds Which corresponds to a center signal, C, output from the signal tion With the folloWing draWings, Wherein: FIG. 1 is a schematic block diagram of a ?rst embodiment of a multi-channel audio enhancement system for generating a pair of enhanced output signals to create a surround-sound effect. processed left channel signal 26 and a processed right 55 source 16. Not all signal sources Will provide a separate bass effects channel B, nor a center channel C, and therefore it is to be understood that these channels are shoWn as optional FIG. 2 is a schematic block diagram of a second embodi ment of a multi-channel audio enhancement system for signal channels. After ampli?cation by the ampli?er 32, the generating a pair of enhanced output signals to create a surround-sound effect. FIG. 3 is a schematic block diagram depicting an audio and 46, respectively. audio signals. tral or other audio performance. Alternatively, the audio signals 40 and 42 are represented by the output signals 44 In operation, the audio enhancement system 10 of FIG. 1 receives audio information from the audio source 16. The enhancement process for enhancing selected pairs of audio audio information may be in the form of discrete analog or signals. digital channels or as a digital data bitstream. For example, FIG. 4 is a schematic block diagram of an enhancement 65 the audio source 16 may be signals generated from a group of microphones attached to various instruments in an orches circuit for processing selected components from a pair of 5,912,976 5 6 source 16 may be a pre-recorded multi-track rendition of an mere personal preference. For eXample, the processing per formed Within the region 90 may be accomplished Wholly Within a digital signal processor (DSP), Within softWare audio Work. In any event, the particular form of audio data received from the source 16 is not particularly relevant to the loaded into a computer’s memory, or as part of a micro operation of the enhancement system 10. processor’s native signal processing capabilities such as that found in Intel’s Pentium generation of micro-processors. For illustrative purposes, FIG. 1 depicts the source audio signals as comprising eight main channels AO—A7, a single bass or loW-frequency channel, B, and a single center channel signal, C. It can be appreciated by one of ordinary skill in the art that the concepts of the present invention are equally applicable to any multi-channel system of greater or feWer individual audio channels. As Will be explained in more detail in connection With FIGS. 3 and 4, the multi-channel immersion processor 24 modi?es the output signals 22 received from the miXer 20 to 10 Referring noW to FIG. 3, the immersion processor 24 from FIG. 1 is shoWn in association With the signal miXer 20. The processor 24 comprises individual enhancement mod ules 100, 102, and 104 Which each receives a pair of audio signals from the miXer 20. The enhancement modules 100, 102, and 104 process a corresponding pair of signals on the output signals, L0,” and Rom, are acoustically reproduced. stereo level in part by isolating ambient and monophonic components from each pair of signals. These components, along With the original signals are modi?ed to generate resultant signals 108, 110, and 112. Bass, center and other signals Which undergo individual processing are delivered The processor 24 is shoWn in FIG. 1 as an analog processor along a path 118 to a module 116 Which may provide level operating in real time on the multi-channel miXed output signals 22. If the processor 24 is an analog device and if the audio source 16 provides a digital data output, then the adjustment, simple ?ltering, or other modi?cation of the received signals 118. The resultant signals 120 from the module 116, along With the signals 108, 110, and 112 are create an immersive three-dimensional effect When a pair of 15 output to a miXer 124 Within the processor 24. processor 24 must of course include a digital-to-analog In FIG. 4, an eXemplary internal con?guration of a converter (not shoWn) before processing the signals 22. Referring noW to FIG. 2, a second preferred embodiment of a multi-channel audio enhancement system is shoWn Which provides digital immersion processing of an audio preferred embodiment for the module 100 is depicted. The module 100 consists of inputs 130 and 132 for receiving a 25 source. An audio enhancement system 50 is shoWn com prising a digital audio source 52 Which delivers audio information along a path 54 to a multi-channel digital audio decoder 56. The decoder 56 transmits multiple audio chan pair of audio signals. The audio signals are transferred to a circuit or other processing means 134 for separating the ambient components from the direct ?eld, or monophonic, sound components found in the input signals. In a preferred embodiment, the circuit 134 generates a direct sound com nel signals along a path 58. In addition, optional bass and center signals B and C may be generated by the decoder 56. ponent along a signal path 136 representing the summation signal M1+M2. A difference signal containing the ambient components of the input signals, MIL-M2, is transferred Digital data signals 58, B, and C, are transmitted to an audio immersion processor 60 operating digitally to enhance the received signals. The processor 60 generates a pair of enhanced digital signals 62 and 64 Which are fed to a digital to analog converter 66. In addition, the signals B and C are fed to the converter 66. The resultant enhanced analog along a path 138. The sum signal M1+M2 is modi?ed by a circuit 140 having a transfer function F1. Similarly, the difference signal MIL-M2 is modi?ed by a circuit 142 having a transfer function F2. The transfer functions F1 and F2 may be identical and in a preferred embodiment provide spatial 35 enhancement to the inputted signals by emphasiZing certain frequencies While deemphasiZing others. The transfer func tions F1 and F2 may also apply HRTF-based processing to the inputted signals in order to achieve a perceived place ment of the signals upon playback. If desired, the circuits signals 68 and 70, corresponding to the loW frequency and center information, are fed to the poWer ampli?er 32. Similarly, the enhanced analog left and right signals, 72, 74, are delivered to the ampli?er 32. The left and right enhanced signals 72 and 74 may be diverted to a recording device 30 for storing the processed signals 72 and 74 directly on a recording medium such as magnetic tape or an optical disk. Once stored on recorded media, the processed audio infor mation corresponding to signals 72 and 74 may be repro duced by a conventional stereo system Without further enhancement processing to achieve the intended immersive effect described herein. 140 and 142 may be used to insert time delays or phase shifts of the input signals 136 and 138 With respect to the original signals M1 and M2. The circuits 140 and 142 output a respective modi?ed sum and difference signal, (M1+M2)P and (M1—M2)P, along paths 144 and 146, respectively. The original input signals M1 and M2, as Well as the processed signals (M1+M2)P and (M 1—M2)P are fed to multipliers Which adjust the gain of the The ampli?er 32 delivers an ampli?ed left output signal 80, LOUT, to the left speaker 34 and delivers an ampli?ed received signals. After processing, the modi?ed signals eXit the enhancement module 100 at outputs 150, 152, 154, and 156. The output 150 delivers the signal KlMl, the output 152 delivers the signal K2F1(M1+M2), the output 154 deliv right output signal 82, ROUT, to the right speaker 36. Also, an ampli?ed bass effects signal 84, BOUT, is delivered to a sub-Woofer 86. An ampli?ed center signal 88, COUT, may be delivered to an optional center speaker (not shoWn). For near ?eld reproductions of the signals 80 and 82, i.e., Where a listener is position close to and in betWeen the speakers 34 55 ers the signal K3F4(M1—M2), and the output 156 delivers the signal K4M2, Where K1—K4 are constants determined by the setting of multipliers 148. The type of processing performed by the modules 100, 102, 104, and 116, and in particular the and 36, use of a center speaker may not be necessary to circuits 134, 140, and 142 may be user-adjustable to achieve achieve adequate localiZation of a center image. HoWever, in far-?eld applications Where listeners are positioned rela tively far from the speakers 34 and 36, a center speaker can be used to ?X a center image betWeen the speaker 34 and 36. The combination consisting largely of the decoder 56 and the processor 60 is represented by the dashed line 90 Which may be implemented in any number of different Ways depending on a particular application, design constraints, or a desired effect and/or a desired position of a reproduced sound. In some cases, it may be desirable to process only an ambient component or a monophonic component of a pair of input signals. The processing performed by each module may be distinct or it may be identical to one or more other 65 modules. In accordance With a preferred embodiment Where a pair of audio signals is collectively enhanced before mixing, 5,912,976 7 8 each module 100, 102, and 104 Will generate four processed signals for receipt by the mixer 24 shown in FIG. 3. All of the signals 108, 110, 112, and 120 may be selectively combined by the mixer 124 in accordance With principles system disclosed herein Will be described for use With Dolby AC-3 recorded media. It can be appreciated, hoWever, that the same or similar principles may be applied to other common to one of ordinary skill in the art and dependent upon a user’s preferences. channels to create a surround sound experience. Moreover, While a computer system 200 is shoWn and described in FIG. By processing multi-channel signals at the stereo level, i.e., in pairs, subtle differences and similarities Within the paired signals can be adjusted to achieve an immersive effect AC-3 recorded media may be a television, a combination created upon playback through speakers. This immersive effect can be positioned by applying HRTF-based transfer standardiZed audio recording techniques Which use multiple 5, the audio-visual playback device for reproducing the television/personal computer, a digital video disk player 10 functions to the processed signals to create a fully immersive positional sound ?eld. Each pair of audio signals is sepa rately processed to create a multi-channel audio mixing system that can effectively recreate the perception of a live 15 360 degree sound stage. Through separate HRTF processing (CPU) 220, a mass storage memory and a temporary random of the components of a pair of audio signals, e.g., the ambient and monophonic components, more signal condi tioning control is provided resulting in a more realistic immersive sound experience When the processed signals are access memory (RAM) system 222, an input/output control acoustically reproduced. Examples of HRTF transfer func tions Which can be used to achieve a certain perceived aZimuth are described in the article by E. A. B. ShaW entitled “Transformation of Sound Pressure Level From the Free Field to the Eardrum in the Horizontal Plane”, coupled to a television, or any other device capable of playing a multi-channel audio recording. FIG. 6 is a schematic block diagram of the major internal components of the processing unit 202 of FIG. 5. The unit 202 contains the components of a typical personal computer system, constructed in accordance With principles common to one of ordinary skill, including a central processing unit 25 J.Acoust.Soc.Am., Vol. 56, No. 6, December 1974, and in the article by S. Mehrgarat and V. Mellert entitled “Trans device 224, all interconnected via an internal bus structure. The unit 202 also contains a poWer supply 226 and a recorded media player/recorder 228 Which may be a DVD device or other multi-channel audio source. The DVD player 228 supplies video data to a video decoder 230 for display on a monitor. Audio data from the DVD player 228 is transferred to an audio decoder 232 Which supplies multiple channel digital audio data from the player 228 to an immer sion processor 250. The audio information from the decoder 232 contains a left front signal, a right front signal, a left formation Characteristics of the External Human Ear”, J.Acoust.Soc.Am., Vol. 61, No. 6, June 1977, both of Which are incorporated herein by reference as though fully set surround signal, a right surround signal, a center signal, and forth. Although principles of the present invention as described enhances the audio information from the decoder 232 in a manner suitable for playback With a conventional stereo above in connection With FIGS. 1—4 are suitable for use in playback system. Speci?cally, a left channel signal 252 and professional recording studios to make high-quality recordings, one particular application of the present inven a loW-frequency signal, all of Which are transferred to the immersion audio processor 250. The processor 250 digitally a right channel signal 254 are provided as outputs from the 35 tion is in audio playback devices Which have the capability to process but not reproduce multi-channel audio signals. For example, today’s audio-visual recorded media are being processor 250. A loW-frequency sub-Woofer signal 256 is also provided for delivery of bass response in a stereo playback system. The signals 252, 254, and 256 are ?rst provided to a digital-to-analog converter 258, then to an encoded With multiple audio channel signals for reproduc ampli?er 260, and then output for connection to correspond tion in a home theater surround processing system. Such surround systems typically include forWard or front speakers ing speakers. Referring noW to FIG. 7, a schematic representation of speaker locations of the system of FIG. 5 is shoWn from an for reproducing left and right stereo signals, rear speakers for reproducing left surround and right surround signals, a overhead perspective. The listener 212 is positioned in front of and betWeen the left front speaker 206 and the right front center speaker for reproducing a center signal, and a sub Woofer speaker for reproduction of a loW-frequency signal. 45 Recorded media Which can be played by such surround systems may be encoded With multi-channel audio signals erated from an AC-3 compatible recording in accordance With a preferred embodiment, a simulated surround experi ence is created for the listener 212. In particular, ordinary through such techniques as Dolby’s proprietary AC-3 audio encoding standard. Many of today’s playback devices are playback of tWo channel signals through the speakers 206 not equipped With surround or center channel speakers. As a consequence, the full capability of the multi-channel recorded media may be left untapped leaving the user With an inferior listening experience. Referring noW to FIG. 5, a personal computer system 200 is shoWn having an immersive positional audio processor constructed in accordance With the present invention. The computer system 200 consists of a processing unit 202 coupled to a display monitor 204. A front left speaker 206 and front right speaker 208, along With an optional sub speaker 208. Through processing of surround signals gen and 208 Will create a perceived phantom center speaker 214 from Which monophonic components of left and right sig nals Will appear to emanate. Thus, the left and right signals from an AC-3 six channel recording Will produce the center phantom speaker 214 When reproduced through the speakers 55 206 and 208. The left and right surround channels of the AC-3 six channel recording are processed so that ambient surround sounds are perceived as emanating from rear phantom speakers 215 and 216 While monophonic surround sounds appear to emanate from a rear phantom center reproducing audio signals generated by the unit 202. A speaker 218. Furthermore, both the left and right front signals, and the left and right surround signals, are spatially listener 212 operates the computer system 200 via a key enhanced to provide an immersive sound experience to board 214. The computer system 200 processes a multi channel audio signal to provide the listener 212 With an eliminate the actual speakers 206, 208 and the phantom speakers 215, 216, and 218, as perceived point sources of sound. Finally, the loW-frequency information is reproduced by an optional sub-Woofer speaker 210 Which may be placed Woofer speaker 210 are all connected to the unit 202 for immersive 360 degree surround sound experience from just the speakers 206, 208 and the speaker 210 if available. In accordance With a preferred embodiment, the processing 65 at any location about the listener 212. 5,912,976 9 10 signals SL and SR are ?rst fed through ?xed-gain ampli?ers FIG. 8 is a schematic representation of an immersive processor and mixer for achieving a perceived immersive 330 and 334, respectively, before transmission to the mixers surround effect shoWn in FIG. 7. The processor 250 corre 280 and 284. Finally, the loW-frequency effects channel, B, sponds to that shoWn in FIG. 6 and receives six audio channel signals consisting of a front main left signal ML, a front main right signal MR, a left surround signal SL, a right surround signal SR, a center channel signal C, and a loW frequency effects signal B. The signals ML and MR are fed to corresponding gain-adjusting multipliers 252 and 254 Which are controlled by a volume adjustment signal Mvolume. is fed through an ampli?er 336 to create the output loW frequency effects signal, BOUT. Optionally, the loW fre quency channel, B, may be mixed as part of the output signals, LOUT and ROUT, if no subWoofer is available. The enhancement circuit 250 of FIG. 8 may be imple mented in an analog discrete form, in a semiconductor 10 substrate, through softWare run on a main or dedicated The gain of the center signal C may be adjusted by a ?rst microprocessor, Within a digital signal processing (DSP) multiplier 256, controlled by the signal Mvolume, and a chip, i.e., ?rmWare, or in some other digital format. It is also possible to use a hybrid circuit structure combing both second multiplier 258 controlled by a center adjustment signal Cvolume. Similarly, the surround signals SL and SR are ?rst fed to respective multipliers 260 and 262 Which are 15 analog and digital components since in many cases the source signals Will be digital. Accordingly, an individual ampli?er, an equalizer, or other components, may be real iZed by softWare or ?rmWare. Moreover, the enhancement controlled by a volume adjustment signal Svolume. The main front left and right signals, ML and MR, are each fed to summing junctions 264 and 266. The summing junction 264 has an inverting input Which receives MR and 306 and 320, may employ a variety of audio enhancement a non-inverting input Which receives ML Which combine to produce ML— R along an output path 268. The signal techniques. For example, the circuit devices 270, 306, and 320 may use time-delay techniques, phase-shift techniques, ML— R is fed to an enhancement circuit 270 Which is signal equaliZation, or a combination of all of these tech niques to achieve a desired audio effect. The basic principles characteriZed by a transfer function P1. A processed differ ence signal, (ML— R)P, is delivered at an output of the circuit 270 to a gain adjusting multiplier 272. The output of circuit 270 of FIG. 8, as Well as the enhancement circuits of such audio enhancement techniques are common to one 25 the multiplier 272 is fed directly to a left mixer 280 and to an inverter 282. The inverted difference signal (MR—ML)P is transmitted from the inverter 282 to a right mixer 284. A circuit 250 uniquely conditions a set of AC-3 multi-channel signals to provide a surround sound experience through playback of the tWo output signals LOUT and ROUT. Speci?cally, the signals ML and MR are processed collec tively by isolating the ambient information present in these signals. The ambient signal component represents the dif ferences betWeen a pair of audio signals. An ambient signal component derived from a pair of audio signals is therefore summation signal ML+MR exits the junction 266 and is fed to a gain adjusting multiplier 286. The output of the multi plier 286 is fed to a summing junction Which adds the center channel signal, C, With the signal ML+MR. The combined signal, ML+MR+C, exits the junction 290 and is directed to both the left mixer 280 and the right mixer 284. Finally, the original signals ML and MR are ?rst fed through ?xed gain adjustment circuits, i.e., ampli?ers, 290 and 292, of ordinary skill in the art. In a preferred embodiment, the immersion processor 35 respectively, before transmission to the mixers 280 and 284. The surround left and right signals, SL and SR, exit the multipliers 260 and 262, respectively, and are each fed to summing junctions 300 and 302. The summing junction 300 often referred to as the “difference” signal component. While the circuits 270, 306, and 320 are shoWn and described as generating sum and difference signals, other embodiments of audio enhancement circuits 270, 306, and 320 may not distinctly generate sum and difference signals at all. This can be accomplished in any number of Ways has an inverting input Which receives SR and a non-inverting using ordinary circuit design principles. For example, the input Which receives SL Which combine to produce SL—SR along an output path 304. All of the summing junctions 264, isolation of the difference signal information and its subse 266, 300, and 302 may be con?gured as either an inverting ampli?er or a non-inverting ampli?er, depending on Whether a sum or difference signal is generated. Both inverting and simultaneously at the input stage of an ampli?er circuit. In addition to processing of AC-3 audio signal sources, the circuit 250 of FIG. 8 Will automatically process signal sources having feWer discrete audio channels. For example, quent equaliZation may be performed digitally, or performed 45 non-inverting ampli?ers may be constructed from ordinary operational ampli?ers in accordance With principles com if Dolby Pro-Logic signals are input by the processor 250, i.e., Where SL=SR, only the enhancement circuit 320 Will mon to one of ordinary skill in the art. The signal SL—SR is fed to an enhancement circuit 306 Which is characteriZed by operate to modify the rear channel signals since no ambient component Will be generated at the junction 300. Similarly, if only tWo-channel stereo signals, ML and MR, are present, a transfer function P2. A processed difference signal, (SL SR)P, is delivered at an output of the circuit 306 to a gain adjusting multiplier 308. The output of the multiplier 308 is then the processor 250 operates to create a spatially fed directly to the left mixer 280 and to an inverter 310. The enhanced listening experience from only tWo channels through operation of the enhancement circuit 270. inverted difference signal (SR—SL)P is transmitted from the inverter 310 to the right mixer 284. A summation signal 55 In accordance With a preferred embodiment, the ambient information of the front channel signals, Which can be SL+SR exits the junction 302 and is fed to a separate enhancement circuit 320 Which is characteriZed by a transfer represented by the difference ML—MR, is equaliZed by the function P3. A processed summation signal, (SL+SR)P, is circuit 270 according to the frequency response curve 350 of delivered at an output of the circuit 320 to a gain adjusting multiplier 332. While reference is made to sum and differ ence signals, it should be noted that use of actual sum and correction, or “perspective”, curve. Such equaliZation of the ambient signal information broadens and blends a perceived difference signals is only representative. The same process ing can be achieved regardless of hoW the ambient and monophonic components of a pair of signals are isolated. The output of the multiplier 332 is fed directly to the left mixer 280 and to the right mixer 284. Also, the original FIG. 9. The curve 350 can be referred to as a spatial sound stage generated from a pair of audio signals by selectively enhancing the sound information that provides a 65 sense of spaciousness. The enhancement circuits 306 and 320 modify the ambi ent and monophonic components, respectively, of the sur 5,912,976 11 12 round signals SL and SR. In accordance With a preferred embodiment, the transfer functions P2 and P3 are equal and both apply the same level of perspective equalization to the respect to that applied to ML—MR. This is required since the normal frequency response of the human ear for sounds directed at a listener from Zero degrees aZimuth Will empha corresponding input signal. In particular, the circuit 306 siZe sounds centered around approximately 2.75 kHZ. The emphasis of these sounds results from the inherent transfer function of the average human pinna and from ear canal equaliZes an ambient component of the surround signals, represented by the signal SL—SR, While the circuit 320 equaliZes an monophonic component of the surround resonance. The perspective curve 352 of FIG. 10 counteracts the inherent transfer function of the ear to create the per signals, represented by the signal SL+SR. The level of equalization is represented by the frequency response curve 352 of FIG. 10. The perspective equaliZation curves 350 and 352 are displayed in FIGS. 9 and 10, respectively, as a function of 10 maintain the perception of a broad rear sound stage as if displayed in log format. The gain level in decibels at reference signal since ?nal ampli?cation of the overall output signals occurs in the ?nal mixing process. Referring initially to FIG. 9, and according to a preferred embodiment, the perspective curve 350 has a peak gain at a pointAlocated at approximately 125 HZ. The gain of the perspective curve 350 decreases above and beloW 125 HZ at a rate of approxi mately 6 dB per octave. The perspective curve 350 reaches a minimum gain at a point B Within a range of approximately out of phase to the corresponding mixers 280 and 284 to reproduced by phantom speakers 215 and 216. gain, measured in decibels, against audible frequencies individual frequencies are only relevant as they relate to a ception of rear speakers for the signals SL—SR and SL+SR. The resultant processed difference signal (SL— R)P is driven 15 By separating the surround signal processing into sum and difference components, greater control is provided by alloW ing the gain of each signal, SL—SR and SL+SR, to be adjusted separately. The present invention also recogniZes that cre ation of a center rear phantom speaker 218, as shoWn in FIG. 20 7, requires similar processing of the sum signal SL+SR since the sounds actually emanate from forWard speakers 206 and 208. Accordingly, the signal SL+SR is also equaliZed by the circuit 320 according to the curve 352 of FIG. 10. The resultant processed signal (SL+SR)P is driven in-phase to achieve the perceived phantom speaker 218 as if the tWo phantom rear speakers 215 and 216 actually existed. For 1.5—2.5 kHZ. The gain increases at frequencies above point B at a rate of approximately 6 dB per octave up to a point 25 audio reproduction systems Which include a dedicated center C at approximately 7 kHZ, and then continues to increase up to approximately 20 kHZ, i.e., approximately the highest channel speaker, the circuit 250 of FIG. 8 can be modi?ed so that the center signal C is fed directly to such center frequency audible to the human ear. Referring noW to FIG. 10, and according to a preferred embodiment, the perspective curve 352 has a peak gain at a speaker instead of being mixed at the mixers 280 and 284. The approximate relative gain values of the various signals Within the circuit 250 can be measured against a 0 dB 30 point A located at approximately 125 HZ. The gain of the reference for the difference signals exiting the multipliers perspective curve 350 decreases beloW 125 HZ at a rate of 272 and 308. With such a reference, the gain of the ampli ?ers 290, 292, 330, and 334 in accordance With a preferred embodiment is approximately —18 dB, the gain of the sum approximately 6 dB per octave and decreases above 125 HZ at a rate of approximately 6 dB per octave. The perspective curve 352 reaches a minimum gain at a point B Within a 35 range of approximately 1.5—2.5 kHZ. The gain increases at frequencies above point B at a rate of approximately 6 dB per octave up to a maximum-gain point C at approximately 10.5—11.5 kHZ. The frequency response of the curve 352 decreases at frequencies above approximately 11.5 kHZ. Apparatus and methods suitable for implementing the 40 45 are disclosed in Us. Pat. Nos. 4,738,669 and 4,866,744, issued to Arnold I. Klayman, both of Which are also incor porated by reference as though fully set forth herein. In operation, the circuit 250 of FIG. 8 uniquely functions to position the ?ve main channel signals, ML, MR, C, SR, and SL about a listener upon reproduction by only tWo speakers. 55 the speakers 206 and 208 shoWn in FIG. 7. This is accom 60 signal strength for the various signals of FIG. 8 is also affected by the volume adjustments and the level of mixing applied by the mixers 280 and 284. 65 produce a much improved audio effect because ambient sounds are selectively emphasiZed to fully encompass a listener Within a reproduced sound stage. Ignoring the rela tive gains of the individual components, the audio output plished through selective equaliZation of the ambient signal Accordingly, the audio output signals LOUT and ROUT applied to the signal SL —SR to broaden and spatially enhance the ambient sounds from the signals SL and SR. In addition, hoWever, the equaliZation curve 352 modi?es the signal SL—SR to account for HRTF positioning to obtain the per ception of rear speakers 215 and 216 of FIG. 7. As a result, the curve 352 contains a higher level of emphasis of the loW and high frequency components of the signal SL—SR With at desired levels. In fact, if the level adjustment of multi pliers 308 and 332 are desirably With the rear signal input levels, then it is possible to connect the enhancement circuits directly to the input signals SL and SR. As can be appreciated by one of ordinary skill in the art, the ?nal ratio of individual perception of a Wide forWard sound stage emanating from information to emphasiZe the loW and high frequency com ponents. Similarly, the equaliZation curve 352 of FIG. 10 is to the type of sound reproduced and tailored to a user’s personal preferences. An increase in the level of a sum signal emphasiZes the audio signals appearing at a center stage positioned betWeen a pair of speakers. Conversely, an increase in the level of a difference signal emphasiZes the ambient sound information creating the perception of a Wider sound image. In some audio arrangements Where the parameters of music type and system con?guration are knoWn, or Where manual adjustment is not practical, the multipliers 272, 286, 308, and 332 may be preset and ?xed As discussed previously, the curve 350 of FIG. 9 applied to the signal ML—MR broadens and spatially enhances ambient sounds from the signals ML and MR. This creates the user preferences and may be varied Without departing from the spirit of the invention. Adjustment of the multipliers 272, 286, 308, and 332 alloWs the processed signals to be tailored equaliZation curves 350 and 352 of FIGS. 9 and 10 are similar to those disclosed in pending application Ser. No. 08/430751 ?led on Apr. 27, 1995, Which is incorporated herein by reference as though fully set forth. Related audio enhancement techniques for enhancing ambient information signal exiting the ampli?er 332 is approximately —20 dB, the gain of the sum signal exiting the ampli?er 286 is approxi mately —20 dB, and the gain of the center channel signal exiting the ampli?er 258 is approximately —7 dB. These relative gain values are purely design choices based upon signals LOUT and ROUT are represented by the folloWing mathematical formulas: 5,912,976 13 14 The enhanced output signals represented above may be and B, and points B and C. In a surround sound environment, a gain separation much larger than 9 dB may tend to reduce a listener’s perception of mid-range de?nition. Implementation of the perspective curve by a digital signal processor Will, in most cases, more accurately re?ect the design constraints discussed above. For an analog implementation, it is acceptable if the frequencies corre (2) magnetically or electronically stored on various recording media, such as vinyl records, compact discs, digital or analog audio tape, or computer data storage media. Enhanced audio output signals Which have been stored may then be reproduced by a conventional stereo reproduction sponding to points A, B, and C, and the constraints on gain separation, vary by plus or minus 20 percent. Such deviation from the ideal speci?cations Will still produce the desired enhancement effect, although With less than optimum system to achieve the same level of stereo image enhance ment. Referring to FIG. 11, a schematic block diagram is shoWn of a circuit for implementing the equalization curve 350 of FIG. 9 in accordance With a preferred embodiment. The circuit 270 inputs the ambient signal ML—MR, corresponding 15 to that found at path 268 of FIG. 8. The signal ML—MR is ?rst conditioned by a high-pass ?lter 360 having a cutoff frequency, or —3 dB frequency, of approximately 50 HZ. Use of the ?lter 360 is designed to avoid over-ampli?cation of the bass components present in the signal ML—MR. The output of the ?lter 360 is split into three separate signal paths 362, 364, and 366 in order to spectrally shape the ambient signal SL—SR, corresponding to that found at path 304 of FIG. 8. The signal SL—SR is ?rst conditioned by a high-pass ?lter 380 having a cutoff frequency of approxi the signal ML—MR. Speci?cally, ML—MR is transmitted mately 50 HZ. As in the circuit 270 of FIG. 11, the output of along the path 362 to an ampli?er 368 and then on to a summing junction 378. The signal ML—MR is also transmit results. Referring noW to FIG. 12, a schematic block diagram is shoWn of a circuit for implementing the equaliZation curve 352 of FIG. 10 in accordance With a preferred embodiment. Although the same curve 352 is used to shape the signals SL-SR and SL +SR, for ease of discussion purposes, reference is made in FIG. 12 only to the circuit enhancement device 306. In a preferred embodiment, the characteristics of the device 306 is identical to that of 320. The circuit 306 inputs 25 ted along the path 364 to a loW-pass ?lter 370, then to an ampli?er 372, and ?nally to the summing junction 378. Lastly, the signal ML—MR is transmitted along the path 366 the ?lter 380 is split into three separate signal paths 382, 384, and 386 in order to spectrally shape the signal SL—SR. Speci?cally, the signal SL— R is transmitted along the path 382 to an ampli?er 388 and then on to a summing junction to a high-pass ?lter 374, then to an ampli?er 376, and then to the summing junction 378. Each of the separately con ditioned signals ML— R are combined at the summing 396. The signal SL—SR is also transmitted along the path 384 junction 378 to create the processed difference signal (ML MR)P. In a preferred embodiment, the loW-pass ?lter 370 has ?nally to the summing junction 396. Lastly, the signal SL—SR is transmitted along the path 386 to a loW-pass ?lter 398, a cutoff frequency of approximately 200 HZ While the high-pass ?lter 374 has a cutoff frequency of approximately to a high-pass ?lter 390 and then to a loW-pass ?lter 392. The output of the ?lter 392 is transmitted to an ampli?er 394, and 35 7 kHZ. The exact cutoff frequencies are not critical so long as the ambient components in a loW and high frequency range, relative to those in a mid-frequency range of approxi mately 1 to 3 kHZ, are ampli?ed. The ?lters 360, 370, and 374 are all ?rst order ?lters to reduce complexity and cost but may conceivably be higher order ?lters if the level of processing, represented in FIGS. 9 and 10, is not signi? cantly altered. Also in accordance With a preferred embodiment, the ampli?er 368 Will have an approximate gain of one-half, the ampli?er 372 Will have a gain of approximately 1.4, and the ampli?er 376 Will have an then to an ampli?er 400, and then to the summing junction 396. Each of the separately conditioned signals SL—SR are combined at the summing junction 396 to create the pro cessed difference signal (SL—SR)P. In a preferred embodiment, the high-pass ?lter 370 has a cutoff frequency of approximately 21 kHZ While the loW-pass ?lter 392 has a cutoff frequency of approximately 8 kHZ. The ?lter 392 serves to create the maximum-gain point C of FIG. 10 and may be removed if desired. Additionally, the loW-pass ?lter 45 398 has a cutoff frequency of approximately 225 HZ. As can be appreciated by one of ordinary skill in the art, there are many additional ?lter combinations Which can achieve the frequency response curve 352 shoWn in FIG. 10 Without approximate gain of unity. departing from the spirit of the invention. For example, the The signals Which exit the ampli?ers 368, 372, and 376 make up the components of the signal (ML—MR)P. The overall spectral shaping, i.e., normaliZation, of the ambient exact number of ?lters and the cutoff frequencies are not critical so long as the signal SL—SR is equaliZed in accor dance With FIG. 10. In a preferred embodiment, all of the ?lters 380, 390, 392, and 398 are ?rst order ?lters. Also in accordance With a preferred embodiment, the ampli?er 388 signal ML—MR occurs as the summing junction 378 com bines these signals. It is the processed signal (ML—MR)P Which is mixed by the left mixer 280 (shoWn in FIG. 8) as part of the output signal LOUT. Similarly, the inverted signal (MR— L)P is mixed by the right mixer 284 (shoWn in FIG. 8) as part of the output signal ROUT. 55 Referring again to FIG. 9, in a preferred embodiment, the gain separation betWeen points A and B of the perspective curve 350 is ideally designed to be 9 dB, and the gain separation betWeen points B and C should be approximately 6 dB. These ?gures are design constraints and the actual ?gures Will likely vary depending on the actual value of components used for the circuit 270. If the gain of the ampli?ers 368, 372, and 376 of FIG. 11 are ?xed, then the perspective curve 350 Will remain constant. Adjustment of the ampli?er 368 Will tend to adjust the amplitude level of point B thus varying the gain separation betWeen points A Will have an approximate gain of 0.1, the ampli?er 394 Will have a gain of approximately 1.8, and the ampli?er 400 Will have an approximate gain of 0.8. It is the processed signal (SL—SR)P Which is mixed by the left mixer 280 (shoWn in FIG. 8) as part of the output signal LOUT. Similarly, the inverted signal (SR—SL)P is mixed by the right mixer 284 (shoWn in FIG. 8) as part of the output signal ROUT. Referring again to FIG. 10, in a preferred embodiment, 65 the gain separation betWeen points A and B of the perspec tive curve 352 is ideally designed to be 18 dB, and the gain separation betWeen points B and C should be approximately 10 dB. These ?gures are design constraints and the actual ?gures Will likely vary depending on the actual value of components used for the circuits 306 and 320. If the gain of the ampli?ers 388, 394, and 400 of FIG. 12 are ?xed, then 5,912,976 15 16 the perspective curve 352 Will remain constant. Adjustment located Within the front sound stage, and Wherein said center of the ampli?er 388 Will tend to adjust the amplitude level of point B of the curve 352, thus varying the gain separation betWeen points A and B, and points B and C. channel signal is combined With a monophonic component of the main left and right signals by said signal mixer to generate said left and right output signals. Through the foregoing description and accompanying 4. The system of claim 1 Wherein said at least four discrete draWings, the present invention has been shoWn to have important advantages over current audio reproduction and enhancement systems. While the above detailed description has shoWn, described, and pointed out the fundamental novel features of the invention, it Will be understood that various omissions and substitutions and changes in the form and details of the device illustrated may be made by those skilled in the art, Without departing from the spirit of the invention. Therefore, the invention should be limited in its audio signals comprises a center channel signal having scope only by the folloWing claims. What is claimed is: 1. A system for processing at least four discrete audio 10 discrete audio signals When said left and right output signals are acoustically reproduced. 6. The system of claim 1 Wherein said ?rst audio enhancer 15 mately 1 kHZ and above approximately 2 kHZ relative to frequencies betWeen approximately 1 and 2 kHZ. 7. The system of claim 6 Wherein the peak gain applied to boost said ambient component, relative to the gain applied to said ambient component betWeen approximately 1 and 2 kHZ, is approximately 8 dB. audio information intended for playback from a front sound stage, and surround left and right signals containing audio information intended for playback from a rear sound stage, said system generating a pair of left and right output signals system comprising: 8. The system of claim 1 Wherein said second and third 25 a ?rst electronic audio enhancer receiving said main left and right signals, said ?rst audio enhancer processing an ambient component of said main left and right signals to create the perception of a broadened sound image across the front sound stage When said left and right output signals are reproduced by a pair of speak ers positioned Within the front sound stage; a second electronic audio enhancer receiving said sur round left and right signals, said second audio enhancer equaliZes said ambient component of said main left and right signals by boosting said ambient component beloW approxi signals including main left and right signals containing for reproduction from the front sound stage to create the perception of a three dimensional sound image Without the need for actual speakers placed in the rear sound stage, said center stage audio information Which is acoustically repro duced by a dedicated center channel speaker. 5. The system of claim 1 Wherein said ?rst, second, and third electronic audio enhancers apply an HRTF-based trans fer function to a respective one of said discrete audio signals for creating an apparent sound image corresponding to said audio enhancers equaliZe said ambient and monophonic components of said surround left and right signals by boosting said ambient and monophonic components beloW approximately 1 kHZ and above approximately 2 kHZ, relative to frequencies betWeen approximately 1 and 2 kHZ. 9. The system of claim 8 Wherein the peak gain applied to boost said ambient and monophonic components of said surround left and right signals, relative to the gain applied to said ambient and monophonic components betWeen approximately 1 and 2 kHZ, is approximately 18 dB. 10. The system of claim 1 Wherein said ?rst, second, and 35 third electronic audio enhancers are formed upon a semi processing an ambient component of said surround left and right signals to create the perception of an acoustic sound image across the rear sound stage When said left conductor substrate. and right output signals are reproduced by the pair of speakers positioned Within the front sound stage; Ware. 11. The system of claim 1 Wherein said ?rst, second, and third electronic audio enhancers are implemented in soft 12. A multi-channel recording and playback apparatus a third electronic audio enhancer receiving said surround receives a plurality of individual audio signals and processes left and right signals, said third audio enhancer pro cessing a monophonic component of said surround left and right signals to create the perception of an acoustic said plurality of audio signals to provide ?rst and second enhanced audio output signals for achieving an immersive sound image at a center location of the rear sound stage 45 multi-channel recording apparatus comprising: sound experience upon playback of said output signals, said a plurality of parallel audio signal processing devices for modifying the signal content of said individual audio When said left and right output signals are reproduced by the pair of speakers positioned Within the front sound stage; and a signal mixer for generating said left and right output signals from the at least four discrete audio signals by combining the processed ambient component from the main left and right signals, the processed ambient component for the surround left and right signals, and the processed monophonic component from the sur round left and right signals, Wherein said ambient signals Wherein each parallel audio signal processing device comprises: a circuit for receiving tWo of said individual audio signals and isolating an ambient component of said tWo audio signals from a monophonic component of said tWo audio signals; positional processing means capable of electronically 55 out-of-phase relationship With respect to each other. 2. The system of claim 1 Wherein said at least four discrete audio signals comprise a center channel signal containing With respect to a listener; and a multi-channel circuit mixer for combining said pro audio information intended for playback by a front sound stage center speaker, and Wherein said center channel signal is combined by said signal mixer as part of said left and right cessed monophonic components and ambient compo nents generated by said plurality of positional process output signals. 3. The system of claim 1 Wherein said at least four discrete audio signals comprise a center channel signal containing audio information intended for playback by a center speaker applying a head related transfer function to each of said ambient and monophonic components of said tWo audio signals to generate processed ambient and monophonic components, said head related transfer functions corresponding to a desired spatial location components of said main and surround signals are included in the left and right output signals in an ing means to generate said enhanced audio output 65 signals Wherein said processed ambient components are combined in an out-of-phase relationship With respect to said ?rst and second output signals.
* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project