WO2007127821A2 - Procédé et dispositif d'étalonnage d'un système de conformation de faisceau sonore - Google Patents

Procédé et dispositif d'étalonnage d'un système de conformation de faisceau sonore Download PDF

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Publication number
WO2007127821A2
WO2007127821A2 PCT/US2007/067468 US2007067468W WO2007127821A2 WO 2007127821 A2 WO2007127821 A2 WO 2007127821A2 US 2007067468 W US2007067468 W US 2007067468W WO 2007127821 A2 WO2007127821 A2 WO 2007127821A2
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WIPO (PCT)
Prior art keywords
response
speaker
listening position
impulse response
surround
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PCT/US2007/067468
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English (en)
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WO2007127821A3 (fr
Inventor
John L. Melanson
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Cirrus Logic, Inc.
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Publication date
Priority claimed from US11/380,840 external-priority patent/US7606380B2/en
Priority claimed from US11/425,969 external-priority patent/US7804972B2/en
Application filed by Cirrus Logic, Inc. filed Critical Cirrus Logic, Inc.
Publication of WO2007127821A2 publication Critical patent/WO2007127821A2/fr
Publication of WO2007127821A3 publication Critical patent/WO2007127821A3/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/301—Automatic calibration of stereophonic sound system, e.g. with test microphone
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic

Definitions

  • the present invention relates generally to home entertainment devices, and more specifically, to techniques for calibrating an audio or audio/video (A/V) device including a surround sound beam-forming system.
  • A/V audio or audio/video
  • Audio systems in home entertainment systems have evolved along with theatre audio systems to include multi- speaker surround sound capabilities. Only recently have discrete surround signals been available from sources in home entertainment systems and further only recently have encoded sources reached a sufficient level of home use for consumers to justify installation of the requisite equipment. With the development of Digital Versatile Disc (DVD) technology that provides surround audio source information for movies or surround-encoded music, and sophisticated computer games that provide surround audio, surround speaker installation in home environments has become more desirable and frequent. With the recent availability of digital television (DTV) signals, which can include surround audio signals as part of their audiovisual (A/V) information, increasing sales of televisions and/or DTV sets including surround channel outputs are expected.
  • the surround signals may be encoded in a pair of stereo signals, such as early DBX or as in more recent Dolby or THX surround encoding, or may constitute a fully separate audio channel for each speaker, often referred to as discrete encoding.
  • an amplifier unit which may be included in an AV receiver or in a television, provides signals to multiple sets of speakers, commonly in what is referred to as a 5.1, 6.1 or 7.1 arrangement.
  • the 5.1 arrangement includes right, center and left main speakers located in the front of the room, and a right-left pair of surround speakers located in the rear of the room for providing an aural environment in which sounds can be psycho-acoustically located such that they emanate from any horizontal direction.
  • the " .1" suffix indicates that an additional subwoofer is provided for providing low frequency sounds that are typically not sensed as emanating from a particular direction.
  • the 6.1 configuration adds a center channel speaker in the surround speaker set and in a 7.1 configuration, an additional pair of speakers is included over the 5.1 configuration and located even farther back in the room from the surround channel speakers.
  • proper installation of surround channel speakers can be costly and undesirable in many home environments. Wiring must be added and locations with unobstructed paths to the listening area must be available. Since the surround channel audio sources are generated for a particular location of the speakers, they cannot be simply placed at any location in the room and still function properly. It is desirable to position the surround speakers in such a way that the surround sound is diffuse, often limiting possible locations for speaker placement.
  • the term "diffuse" indicates that the sound does not appear to emanate from a single direction, which is generally provided via reflections from or more surfaces that cause the sound to be reflected toward the user from multiple angles.
  • surround channel signals are provided to speakers placed behind the listener.
  • surround channel signal is provided to speakers placed in front of the listener.
  • Simulated surround sound implementations typically use filtering and/or delays to alter mono or stereo audio signals to provide outputs for additional front speakers to generate the surround field.
  • U.S. Patent 6,937,737 describes a simulated surround sound system that provides the right and left surround channel information to each side (right and left) of an additional stereo speaker pair as well as to each side of the main stereo speaker pair. The frequency response of the system is controlled to cause the apparent position of the surround channel information to appear wider than the speaker position.
  • Such systems do not provide surround sound performance approaching that of actual surround sound implementations.
  • the method provides a test signal to multiple speaker drivers through an electronic beam-forming network surround channel input. Then using a microphone, the sound propagated from the multiple speaker drivers is detected at a listening position.
  • a signal relationship between the surround channel input and at least one of the multiple speaker drivers is adjusted in conformity with sound detected at the listening position in order to control propagation of the surround channel information in a direction toward the listening position so that the direct sound is substantially attenuated.
  • the surround channel information is thereby propagated by the multiple speaker drivers according to a directivity pattern having at least one lobe directed away from the listening position, so that the surround channel information is substantially only heard as reflections from the walls or ceiling of the listening room, rather than propagating directly toward the listening position.
  • the adjustment criteria may be to maximize the late vs. early response for the surround channel information at the listening position.
  • the signal relationship adjusted by the calibration method may be a frequency-dependent phase or phase/amplitude relationship provided by adjustable finite impulse response (FIR) filters, or may be another less complex adjustment, such as an amplitude adjustment of a canceling surround channel signal applied to a front directed speaker driver carrying primarily main channel information, but including a canceling surround channel signal for canceling sound directed toward the listening position from a surround channel speaker that is oriented in another direction.
  • FIR finite impulse response
  • the calibration apparatus includes a test signal source, a microphone, a microphone preamplifier circuit and a detector for detecting the signal received by the microphone.
  • the detector may be a correlator and may include multiple correlators for correlating multiple information sources encoded within the test pattern, such as surround and main channel information for multiple pairs of speaker drivers. Alternatively, a single correlator may be use in successive measurements to determine surround channel response for each set of drivers at the listening position and optionally the main channel response, as well.
  • calibration of beam-forming types other than surround beam-forming may be employed.
  • a beam-forming system is calibrated to form a beam as narrow as possible at a specific listening location, so that program information is not as audible in other directions away from the specific listening position.
  • the beam-forming system is calibrated to provide separate program information to two different listening positions to maximize audio separation at one listening position from the program information provided to another listening position.
  • Figures 1A-1D are views of a room incorporating a DTV surround-sound system that includes calibration circuits for calibration of the system according to methods of the present invention.
  • Figure 2 is a block diagram of the DTV surround sound system of Figures IA-ID.
  • Figure 3A is an illustration showing a speaker arrangement that can be employed in the system of Figures 1 and 2.
  • Figure 3B is a graph showing sound pressure level directivity patterns produced by the speaker arrangement of Figure 3A in surround mode.
  • Figures 3C is a graph illustrating a frequency response of speaker driver channels within the system of Figures 1 and 2.
  • Figure 3D is an illustration showing an alternative frequency response of speaker driver channels within the system of Figures 1 and 2.
  • Figure 4A is a block diagram of another system that includes program instructions for performing calibration methods in accordance with an embodiment of the present invention.
  • Figure 4B is a block diagram of a direct and surround channel circuit that is calibrated by a method in accordance with an embodiment of the present invention.
  • Figure 5A is a block diagram of yet another system that is calibrated in accordance with embodiments of the present invention .
  • Figure 5B is an illustration depicting a DTV speaker arrangement that is calibrated by methods in accordance with embodiments of the present invention.
  • Figure 6 is a block diagram of a calibration subsystem for performing calibration methods in accordance with an embodiment of the present invention.
  • Figure 7 is a flowchart depicting a surround mode calibration method in accordance with an embodiment of the present invention.
  • Figures 8A and 8B are graphs showing sound pressure level directivity patterns in night mode and picture-in- picture/split screen mode, respectively.
  • Figure 9 is a flowchart depicting a night mode calibration method in accordance with an embodiment of the present invention.
  • Figure 10 is a flowchart depicting a picture-in- picture/split screen mode calibration method in accordance with another embodiment of the present invention.
  • the present invention encompasses systems and methods for calibrating audio systems that include beam-forming capabilities.
  • the system may be a device such as a video device having speakers included for the rendering of audio content, such as a DTV or computer monitor, may be an audio-only device, such as a stereo system having internal speakers, or beam- forming capabilities may be incorporated within the speaker cabinets attached to an ordinary audio or video device.
  • the surround channel signal (s) are provided via beam-forming that produces reflections via one or more beams directed away from the listener.
  • the beam(s) are formed by a phase-aligned combination of an internal and an external speaker and calibrated by a method according to an embodiment of the present invention using a test signal and a microphone.
  • Special beam-forming modes provide an isolated listening location for night-time viewing ("Night Mode") or for simultaneous viewing of split-screen or picture-in-picture (PIP) program selection in two or more listening locations, and are calibrated by methods in accordance with other embodiments of the invention.
  • the goal is to achieve a substantial attenuation of sound (e.g., 6dB or more) due to a particular input at one or more particular positions by adjusting a directivity pattern produced by multiple speaker drivers.
  • the parent U.S. Patent Applications disclose various systems and speaker configurations for providing surround sound beam-forming without requiring a horizontal speaker array having a large number of elements.
  • the present invention includes methodologies and for calibrating the systems disclosed therein, as well as other simulated surround sound systems.
  • the present invention also includes methodologies for calibrating the night- mode and split-screen/picture-in-picture (PIP) modes of U.S. related Patent Application 11/380,840 entitled “METHOD AND SYSTEM FOR SOUND BEAM-FORMING USING INTERNAL DEVICE SPEAKERS IN CONJUNCTION WITH EXTERNAL SPEAKERS” as well as other systems using speaker drivers located in separate speaker cabinets for beam-forming in such modes.
  • PIP split-screen/picture-in-picture
  • the illustrated system is a DTV 10 that includes an internal set of stereo speakers 14A-B and a set of external speakers 12A-B having inputs coupled to DTV 10 for operating external speakers 12A-B in phase-alignment with internal speakers 14A-B.
  • phase-alignment is understood to define a particular phase relationship between the speakers and not necessarily a zero-time aligned relationship with respect to each channel and speaker. In fact, it is the difference between the time-alignment for surround channels versus main channels that provides the directionality used in the present invention to present diffuse surround channel information and direct main channel information from speakers located substantially near a single wall of a room.
  • the DTV of the present invention uses the vertical offset of speakers within speaker pairs 12A, 14A and 12B, 14B to project a beam 17A, 17B to reflection points 19A,
  • the present invention calibrates the system so that the main channel (front speaker) information is maximized according to the vector sum of direct paths 18A, 18B such that the main speaker information is provided in-phase at listening position 16, while the surround channel (rear speaker) information is substantially attenuated by the vector sum of direct paths 18A, 18B, so that a listener at listening position 16 will hear the surround channel information only as reflected energy from ceiling 15 and room walls. Since each pair of speakers 12A, 14A and 12B, 14B provides a two-lobed pattern, another maximum intensity beam is directed toward the floor of the room. However, the floor in a home environment is typically carpeted, which attenuates the higher frequencies involved in the surround channel beam.
  • the system will generally be calibrated to suppress the reflection from the floor, which is also more subject to obstruction, even if the floor is sound- absorbent. Also, in the configuration shown, the floor path to the listener would be shorter, and thus provide less apparent distance.
  • DTV 10 is mounted on a wall, it is generally desirable to mount external speakers 12A-B slightly below DTV 10 or in general, at approximately mid-height with respect to the total height of the wall.
  • the surround beam-forming calibrated by the present invention generally uses a limited band of frequencies that is above the low-frequency range where beam-forming is not necessary due to the non-directive perception of low frequency acoustic energy and also not practical due to the spacing required in the beam-forming array.
  • Energy below approximately 250Hz is generally provided only in the direct channel, which is either a substantially in-phase signal provided to internal speakers 14A-B and external speakers 12A-B, or the low-frequency information may be provided only to external speakers 12A-B.
  • the low-frequency cut-off frequency can be set in conformity with a typical speaker spacing such that no beam is formed for the common (in-phase) low frequency information.
  • the practical low-frequency cut-off can be "learned" during the calibration process described below and the cut-off frequency adjusted in conformity with the calibration measurement results. Additionally, the calibration method can determine whether it is practical to use the internal speakers 14A-B for low frequency operation. If poor low-frequency response is detected with respect to internal speakers 14A-B, they can be selectively disabled.
  • the speakers employed in DTV devices which must fit the package dimensions and cost point for the DTV components, will generally have poorer low-frequency performance than even a low-cost set of external bookshelf speakers. Additionally, less amplifier power is required for the higher-frequency audio bands, and therefore the amplifiers provided in DTV 10 can be much smaller and dissipate less heat if only the higher-frequency components of the main and surround channel signals are provided to internal speakers 14A-14B.
  • the beam-forming channel is also generally band- limited to remove higher frequencies, for example, those above approximately 2500Hz, for which the spacing between speaker pairs 12A, 14A and 12B, 14B usually extends to multiple wavelengths, and therefore would generate a "combing" effect that would be difficult to remove with calibration.
  • the high-frequency information may be provided to internal speakers 14A-B and removed from the signals provided to external speakers 12A-B.
  • Internal speakers 14A-B are generally provided with signals directly from amplifiers internal to DTV 10.
  • the high-frequency information can be processed via delays or filtering to provide a simulated surround effect from a single speaker used as a tweeter.
  • External speakers 12A-B will generally be powered speakers that receive either a corresponding line-level analog output signal from DTV 10 or a digital signal such as an optical or coaxial SONY/PHILIPS Digital Interface (S/P-DIF) connection.
  • additional amplifiers may be included within DTV 10 that can provide power signals to external "non-powered” speakers.
  • FIG. 1C a “Night Mode”, as illustrated in Figure 1C, the system can be calibrated to neutralize sound in all zones apart from a particular limited listening area 16A and in "Picture-in-Picture (PIP) Mode", for use with split screen viewing or PIP screen presentation of video, two listening areas 16B and 16C can be provided as illustrated by Figure ID, where the goal is not to neutralize sound outside the listening areas 16B and 16C, but to maximize isolation between the two zones, which is generally accomplished by using the right and left stereo channels for the separate audio information, but calibrating the system to neutralize sound for the non-corresponding channel within each of listening areas 16B and 16C.
  • PIP Picture-in-Picture
  • the PIP mode can be accomplished somewhat using the horizontal displacement between the right and left pairs of speakers 12A, 14A and 12B, 14B and directing nulls with respect to the undesired program channel at each listening position.
  • the above frequency limitations on the beam-forming range can be further refined to establish proper operating frequency ranges for the drivers.
  • the speakers can be oriented with the drivers in a horizontally displaced configuration and calibrated to provide further spatial discrimination for night-mode and PIP beam-forming modes .
  • DTV 10 includes a DTV receiver/decoder 22 that receives digital and/or analog television signals from a cable television (CATV) , digital versatile disc (DVD) player, videocassette recorder (VCR) , antenna or other form of signal connection (not shown) and provides video information to a video processor 26 that supplies graphical information to a video display 27.
  • Video processor 26 supports such features as picture-in-picture (PIP) and split- screen modes that are relevant to some of the surround audio beam-forming modes described in detail below.
  • PIP picture-in-picture
  • DTV receiver/decoder 22 also provides audio information to an audio signal processor 30 that includes a surround decode/simulator circuit 32, calibration circuits 38 that receive a signal from an external microphone MIC via a preamplifier PA, and a signal combiner/filter network 34.
  • Calibration circuits 38 are controlled by control elements described in further detail below, by methods and algorithms according to embodiments of the present invention.
  • Microphone MIC is ideally an omni-directional microphone, so that all responses with respect to a given speaker or combination of speakers is detected during calibration.
  • the outputs of signal combiner/filter network 34 are provided to DACs 35 that generate analog output signals for internal speakers 14A-B via corresponding power amplifiers Al and A2, and also to external connectors CNl and CN2 that supply line-level signals to amplifiers A3 and A4, which in turn supply power signals to speakers 12A-12B.
  • DACs 35 and amplifiers A1-A2 may be replaced with pulse-width modulator/filter circuits.
  • connectors CN3 and CN4 may be provided if amplifiers A3 and A4 (or PWM output drives/filters) are incorporated within DTV 10.
  • Surround decode/simulator circuit 32 decodes any encoded main channel, surround channel and other surround-sound information in the audio stream(s) provided from DTV receiver/decoder and may optionally synthesize surround channel information if such surround-sound information is absent from the audio streams (s) .
  • Signal combiner/filter network 34 receives the main and surround channel information for each stereo side and generates the proper signals via digital-to-analog converters (DACs) 35 to amplifiers Al-4 to form the direct beam for the main channel information and the reflected beam for the surround channel information.
  • DACs digital-to-analog converters
  • Calibration circuits 38 tune filters within signal combiner/filter network 34 during a calibration set-up process in order to provide the best response at listening position 16 for the main channel information and to maximize the delay of the reflected energy for the surround channel information, when in surround mode. In the other operating modes, the calibration circuits 38 provide other pattern control tuning consistent with those modes as described in further detail below.
  • FIG. 3A an illustration showing a speaker arrangement that may be employed in the system of Figures 1 and 2 is depicted.
  • internal speaker 12A is used at higher frequencies and the beam-forming midrange frequencies and external speaker 14A is used at lower frequencies and the beam-forming midrange frequencies. Therefore, both speakers are active in the midrange beam-forming frequency range.
  • both speakers will typically have a full-range response, but a full-range response is not a requirement to practice the invention.
  • a simplified combiner 34A is shown for illustrative purposes.
  • Signal combiner 34A receives a main channel signal A and a surround channel signal B.
  • the signal provided to internal speaker 12A is A+B for both the midrange (overlap range) and the high frequency range
  • the signal provided to external speaker 14A is A-B for the midrange and A+B for the low-frequency range.
  • the result of the operation of combiner 34A is that the midrange of the surround channel signal B is provided out- of-phase (as between speakers 12A and 14A) along the direct path to a listener located on-axis between speakers 12A and 14A, thus producing a null with respect to the midrange surround channel information toward the listener.
  • the listener will not hear the surround channel information as emanating from speakers 12A and 14A, but will rather hear the surround channel information as diffuse, coming from a range of reflection points primarily along the ceiling of the room.
  • the main channel midrange information is provided in-phase (as between speakers 12A and 14A) along the direct path, so that the main channel information is heard as emanating from the speakers.
  • the main and surround channel information are combined and are only supplied to one speaker of each vertically-displaced speaker pair, so that no beam-forming is produced in those frequency ranges .
  • FIG. 3B a directivity pattern of the speaker arrangement of Figure 3A is shown for the midrange beam-forming range.
  • Signal A is shown as having a substantially cardioid shape, while signal B is produced in two lobes, one directed at the ceiling and one directed at the floor, due to the vertical displacement of speakers 12A and 14A.
  • Figure 3C illustrates the three band filtering scheme of combiner 34A in which beam-forming is employed in the midrange frequency band Mid.
  • the Low frequency band the sum of the main and surround channel information can be sent to both speakers, since the longer wavelengths will ensure that the drivers act in phase.
  • the Low band might be provided only to external speakers selectively, in response to the results of a calibration or user setting, or as a fixed design feature under the assumption that the external speakers 12A-B will have superior low frequency response.
  • the High frequency band generally only one of the full-range speakers will be used so that "combing" effects do not occur due to interference between the speakers .
  • Figure 3D illustrates the frequency response of drivers 14A and 14B and the crossover filtering scheme of combiner 34A in an alternative system as disclosed in parent U.S. Patent Application 11/383,125 entitled “METHOD AND SYSTEM FOR SURROUND SOUND BEAM-FORMING USING THE OVERLAPPING PORTION OF DRIVER FREQUENCY RANGES", in which driver pairs having differing frequency ranges are operated outside of their specified ranges to provide beam-forming action. Beam-forming is employed in the shaded overlap frequency band shown.
  • the crossover slopes (dotted lines) show the main channel crossover frequency locations, which differ from the overlap frequency range boundaries.
  • FIG. 4A a system in accordance with an embodiment of the present invention is shown.
  • the depicted system employs a digital signal processor (DSP) 41 that performs the calibration of the present invention and the signal combining/filtering functions, as well as frequency-band splitting and any compression/protection algorithms used in the system.
  • Microphone MIC is connected to preamplifier PA, which provides signals to analog-to-digital converter ADC, which provides a detected signal for calibration from a test signal generated by DSP 41.
  • DSP 41 is coupled to a program memory 42 containing program instructions forming a computer program product in accordance with one or more embodiments of the present invention, and further coupled to a data memory 43 for storing data used by the computer program and results produced thereby.
  • DSP 41 The outputs of DSP 41 are depicted as pulse-width modulator (PWM) outputs for each channel, with corresponding low-pass filters and driver transistors 44, generally half- bridge circuits with series LC filters connected to speakers 14A-14B and optionally (non-powered) speakers 12A-12B.
  • PWM pulse-width modulator
  • the calibration, signal combining, filtering and compression operations performed by the algorithms of the computer program within program memory 42 will be described in further detail below in illustrations that apply to discrete circuits as well as the calibration algorithms executed by DSP 41.
  • DSP 41 can also be used to detect the nature of the sounds provided by the audio channel (s) and operate the beam-forming algorithms accordingly. Detection of speech is performed by correlating the stereo signals provided for each channel, since most speech information is presented monophonically (i.e., equal and in-phase levels at each channel). The signals are also further analyzed to detect modulation patterns, e.g. envelope shapes that characteristically different for music and speech. The common information between the channels can be processed to detect speech modulation patterns, and the detected speech information can be increased in amplitude or extracted for presentation in the split-screen or PIP operating modes.
  • modulation patterns e.g. envelope shapes that characteristically different for music and speech.
  • the common information between the channels can be processed to detect speech modulation patterns, and the detected speech information can be increased in amplitude or extracted for presentation in the split-screen or PIP operating modes.
  • DSP 41 then equalizes, compresses, and re-processes the audio information provided by each direct beam to improve intelligibility of speech in each direct beam, while the other direct beam might have speech or music.
  • speech can be favored over music as shown in Table I below, which can be applied to PIP or split-screen modes.
  • the surround beams can be provided with the wide portion of the stereo program (i.e., the uncorrelated information between right and left in each stereo signal source), without detracting much from either program's audio.
  • Calibration of beams in PIP or split-screen modes involves placement of the calibration microphone at each location for individual calibration, the provision of two or more directional microphones for simultaneous calibration, or an assumption that the performance of the listening environment will be symmetrical across a line dividing the two listening areas.
  • the response of the direct beam with respect to the two program channels can be optimized by minimizing the ratio of the other program information to the program associated with the beam being measured.
  • "Night Mode" performance can be optimized to reduce the amount of low frequency information, while retaining speech intelligibility and beam forming capability that restricts the space in which sound can be heard. For that purpose, high-frequency energy may also be attenuated in the ranges where combing can cause significant sidelobes to emerge.
  • Calibration can be performed by placement of the microphone in the listening position and tuning the response of the individual horizontal and vertical array elements to form a narrow beam at the listening position.
  • other positions at angles significantly apart from the listening position direction may be measured and the direct sound present at those positions minimized.
  • FIG. 4B a direct and surround channel circuit or algorithm, that is calibrated by methods in accordance with embodiments of the present invention, is shown in a block diagram. Only one stereo side (right or left) of the system is shown, as the other side will generally be an identical circuit.
  • a Main Channel and Surround Channel signal are provided to processing blocks 4OA and 4OB, that provide respective output signals Out 1 and Out 2 to power driver stages that drive a pair of speakers.
  • switch portion SlA connects output signal Out 1 to the Main Channel signal and switch portion SlB connects output signal Out 2 to the Surround Channel signal, so that the system can be selectively used with placement of speakers at actual rear room positions.
  • switch portion SlA connects output signal Out 1 to the output of processing block 4OA and switch portion SlB connects output signal Out 2 to the output of processing block 4OB, so that the system provides beam-formed surround for use with placement of speakers at the front of the room as described above.
  • Processing blocks 4OA and 4OB are similar processing blocks, but processing block 4OA removes low frequency information from output signal Out 1, which serves as a mid-high frequency output in a frequency selective configuration as described above. Similarly, processing block 4OB removes high frequency information from output signal Out 2, serving as the mid-low frequency output.
  • Each of processing blocks 4OA and 4OB includes two adjustable finite impulse response (FIR) or other suitable digital filters 47A-B and 47C-D, respectively, for calibrating the system maximum surround effect by adjusting the impulse response of each output Outl and Out2 with respect to each input (Main and Surround Channels) .
  • FIR finite impulse response
  • an optional pair of high-pass filters 46A and 46B remove low- frequency information from the Main and Surround Channel signals and a pair of adjustable FIR filters 47A and 47B provide for calibration of the beam-forming system.
  • FIR filters 47A and 47B are summed in-phase by a combiner 48A and then applied to an optional compressor 49A that protects a speaker coupled to output signal Out 1 from damage, or in general preserves overhead as the system works to beam-form over the mid frequency range. Also, in other modes such as Night Mode and PIP mode, compression and frequency-selective compression is applied by compressor 49A in order to reduce the audible volume required for intelligibility of speech and to limit the volume of program material such as music.
  • the Main and Surround Channel signals are summed in-phase by a combiner 48B and out- of-phase by a combiner 48C.
  • the output of in-phase combiner 48B is low-pass filtered by filter 46C and provided to inputs of both of a pair of FIR filters 47C and 47D.
  • the output of in- phase combiner 48B is also filtered by a bandpass filter 46D to provide a midrange output and provided to an input of FIR filter 47C.
  • the output of out-of-phase combiner 48C output is also filtered by a bandpass filter 46E to provide a midrange output and provided to an input of FIR filter 47D.
  • the outputs of FIR filters 47C and 47D are then combined and optionally compressed by compressor 49B, which may be linked to compressor 49A to prevent amplifier clipping as the speaker coupled to output signal Out 2 attempts to provide the correct level of midrange signals which may otherwise rise too high as overall system volume is increased.
  • the resulting output of processing block 4OB is a signal having the sum of the Main and Surround channel signals in a low-frequency band, and the difference between the Main and Surround channel signals in the midrange beam-forming band.
  • Compressor 49B is also used in other modes such as Night Mode and PIP mode for the same reasons as described above with respect to compressor 49A.
  • the channel circuit of Figure 4B is an example of an arrangement of blocks that implement beam-forming that may be calibrated by an embodiment of the present invention, or similar cascaded functions that can be applied in a DSP algorithm and calibrated by another DSP algorithm.
  • DSP algorithm calibrated by another DSP algorithm.
  • alternative implementations are possible and in some instances preferred.
  • all of the filtering functions could be performed within FIR filter blocks, with the in-phase/out-of-phase midrange beam-forming summations performed also within the FIR filter blocks.
  • speaker protection compression can be made part of the filter algorithm, as well.
  • a more generic expression of a channel circuit in accordance with an embodiment of the present invention can be made as a set of FIR filters each receiving either a Main or Surround channel signal and having output summed for forming output signals Out 1 and Out 2. Additional FIR filters for each discrete other speaker may be provided (e.g., center speaker or additional horizontally distributed speakers).
  • FIG. 5A is a block diagram of yet another system that is calibrated by embodiments of the present invention, that has an expanded number of speaker output channels. If additional horizontal resolution is provided by horizontally displacing drivers connected to the speaker channels, then enhanced performance can be obtained, particularly in night and PIP modes.
  • Block 50 illustrates a 5.1 surround speaker configuration, adapted for use in a front-only speaker placement.
  • Channel circuits 52A and 52B provide the right and left channel outputs for the respective pair of beam-forming speakers and can be implemented as described above with respect to Figure 4B.
  • An additional set of FIR filters 55A-B and a combiner 57 combines time-aligned surround channel signals for left and right channels with the center channel, permitting the center channel to form part of the overall beam-forming array.
  • FIG. 5B illustrates an exemplary implementation of a 5.1 or 7.1 DTV system and a consequent speaker arrangement.
  • DTV 10 further includes a center speaker C, along with a center left CL and center right CR speaker.
  • the vertical beam-forming speaker array is provided as described above by internal speakers 14A-B in combination with external speakers 12A-B.
  • a subwoofer/effects channel speaker SUB is located beneath DTV 10.
  • the resultant combination increases the degrees of freedom possible in calibrating maximum surround channel effect via adjustment of the individual FIR filters in channel blocks 52A and 52B as well as additional filters 55A-55F of Figure 5A. Further, the horizontal arrangement of additional speakers C, CL and CR greatly improves pattern control and isolation in Night Mode and PIP mode.
  • a calibration controller 64 in response to a user control of DTV 10 applies the output of a sequence generator 60 to signal combiner/filter network 34. Either one channel can be calibrated at a time, or multiple uncorrelated sequences can be provided to all channels for simultaneous calibration.
  • An adjustable delay 63 applies the sequence signal (s) to a correlator (or multiple correlators) 62 that correlate the sequence (s) with a microphone signal provided from detector 61. The arrangement permits calibration controller 64 to determine the impulse response of each channel at the microphone position.
  • the system can then be calibrated via the adjustment of the filter coefficients within signal combiner/filter network 34 to minimize the reverberant (reflected) energy with respect to the main channel inputs or otherwise adjust the main channel response for best performance, while maximizing the reverberation with respect to the surround channel inputs.
  • a sequence such as a maximal-length sequence (MLS) to extract the impulse response of the system, frequency sweeping, chirping, or white/pink noise techniques may be similarly employed, with correlator 62 replaced with an appropriate filter.
  • MLS maximal-length sequence
  • FIG. 7 a flowchart depicting a calibration method in accordance with an embodiment of the present invention is shown.
  • the illustrated method is for a single channel calibration on each pass, but the multi-channel simultaneous calibration follows the same pattern.
  • a test signal tone, noise, or sequence
  • the listening position is monitored with a microphone (step 71) , and if the channel under test is a main (direct) channel (decision 72), then the response of the channel filter is optimized to obtain the best main beam performance, and optionally, the level of reflected energy is minimized if performance will not be compromised (step 73) .
  • low frequencies can be disabled to that speaker (step 75) .
  • the above determination can be made via further selection of not only the channel in step 70, but selectively disabling the signal path to each speaker from the selected channel by disabling the FIR filter that couples the channel to the associated speaker channel.
  • the frequency range over which beam-forming is practical can optionally be learned and the surround channel response can be limited to that range (step 76) .
  • the frequency range over which beam-forming is practical can be determined by determining a low-end frequency at which the direct beam becomes difficult to suppress at the listening position due to loss in phase-cancellation between the internal and external speakers.
  • the high-end frequency at which the beam splits into additional beams due to combing can also be detected as a change in the ability to suppress the direct beam at the listening position.
  • the response of the channel filter is optimized to maximize the delay of the reflected energy (step 77) to achieve the maximum reverberant effect.
  • the process from steps 70-77 is repeated over each channel (or performed simultaneously) and also iterated until all filter sets have been calibrated and the values stabilized as between all of the channels (decision 78) .
  • the above-described calibration can be performed by summing the response of one driver in each driver pair with a time-delayed version of the other driver's response. As the delay is varied, a delay is reached having the greatest surround effect, which is determined as the above-described maximum of the ratio of late response to early response.
  • the figure-of- merit is the ratio of late to early energy in the signal received at the microphone.
  • a reasonable cut-off time for considering energy late vs. early for a typical room, is energy arriving more than 5 ms after the initial impulse response (direct energy) for a single speaker is considered late energy.
  • the impulse response of the adjustable FIR filters in each channel can then be adjusted to accomplish the delay, which can be a frequency dependent delay for each driver.
  • the direct response can also be calibrated in a similar manner, with the delay determined to minimize the reflected energy and maximize the direct (non-reflected) energy.
  • FIG 8A a directivity graph of the system of the present invention in Night Mode is depicted.
  • a single lobe 8OA is formed by adjustment of each of the FIR filters that couple the input channels, which are summed together as a mono signal, to the speakers.
  • Figure 8B illustrates a directivity graph of the system in PIP or split- screen mode, where two distinct patterns are generated for two different input video program audio information channels. The audio information for each program is summed monophonically and then provided to the right and left main inputs of the above- depicted system.
  • the system is calibrated to produce a lobe 8OB or 8OC with respect to each program channel.
  • the system is calibrated to best minimize the energy from program 1 (i.e., the desired program in lobe 80B) at listening position 2 and vice- versa.
  • test signal (tone, noise or sequence) is generated through all speaker channels (step 90) .
  • the listening position is monitored with a microphone (step 91) , and the response of the channel filters is optimized to minimize the level of reverberant energy and to maximize the direct energy (step 92).
  • the frequency range over which beam-forming is practical can be learned, and the Night Mode response can be limited to that range (step 93) .
  • the frequency range over which beam-forming is practical can be determined by determining a low-end frequency at which the reflected energy becomes difficult to suppress at the listening position.
  • the process from steps 90-93 is repeated over each channel (or performed simultaneously), and also iterated until all filter sets have been calibrated and the values converged as between all of the speaker channels (decision 94).
  • a calibration method for PIP and split-screen Mode is depicted in a flowchart, in accordance with an embodiment of the present invention.
  • the first listening position is monitored with a microphone (step 100) .
  • a test signal tone, noise or sequence
  • a first program channel step 101
  • the response of the channel filters is optimized to minimize the level of reverberant energy and maximize the direct energy (step 102).
  • all speakers are selected with respect to a second program channel, a tone, noise or sequence is generated through the second program channel (step 103) , and the response of the channel filter is optimized to minimize the leakage from the second channel at the first listening position (step 104) .
  • the second listening position may be monitored with a microphone (step 110) , a test signal (tone, noise or sequence) is applied to the second program channel (step 111) and the response of the channel filters optimized to minimize the level of reflected energy and maximize the direct energy at the second listening position (step 112). Then, a test signal (tone, noise or sequence) is generated through the first program channel (step 113) , and the response of the channel filter is optimized to minimize the leakage from the first channel at the second listening position (step 114) .
  • the alternative technique provides improved information regarding the attenuation of first channel sound at the second listening position, but requires a second microphone or repositioning of a single microphone in order to accomplish the calibration.
  • the frequency range over which beam-forming is practical can be optionally learned and the PIP mode response can be limited to that range (step 105) .
  • the frequency range over which beam- forming is practical can be determined by determining a low-end frequency at which the reflected energy becomes difficult to suppress at the program-associated listening position or the direct energy becomes difficult to suppress at the alternate listening position.
  • the process from steps 100-105 is repeated until all filter sets have been calibrated and the values stabilized as between all of the speaker channels (decision 106) .

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

Procédé et circuits d'étalonnage de systèmes de conformation de faisceaux sonores bon marché. Un signal test est envoyé à des circuits d'attaque de hauts-parleurs multiples et est détecté par un signal de microphone dans un microphone positionné sur une position d'écoute. Une relation de signal entre des informations de canal surround fournie aux circuits d'attaque des hauts-parleurs multiples est ajustée conformément au signal détecté de sorte que lesdites informations sont sensiblement atténuées le long d'un chemin direct vers la position d'écoute. Il en résulte que les informations de canal sont propagée selon un schéma de directivité comprenant au moins un lobe primaire dirigé à l'écart de la position d'écoute et que ces informations de canal surround sont diffusées par réflexion avant d'atteindre la position d'écoute. La relation de signal peut être régulée par des filtres numériques multiples qui maximisent la réponse récente par rapport à la réponse anciennes des informations de canal surround.
PCT/US2007/067468 2006-04-28 2007-04-26 Procédé et dispositif d'étalonnage d'un système de conformation de faisceau sonore WO2007127821A2 (fr)

Applications Claiming Priority (6)

Application Number Priority Date Filing Date Title
US11/380,840 2006-04-28
US11/380,840 US7606380B2 (en) 2006-04-28 2006-04-28 Method and system for sound beam-forming using internal device speakers in conjunction with external speakers
US11/383,125 2006-05-12
US11/383,125 US7545946B2 (en) 2006-04-28 2006-05-12 Method and system for surround sound beam-forming using the overlapping portion of driver frequency ranges
US11/425,969 2006-06-22
US11/425,969 US7804972B2 (en) 2006-05-12 2006-06-22 Method and apparatus for calibrating a sound beam-forming system

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WO2007127821A2 true WO2007127821A2 (fr) 2007-11-08
WO2007127821A3 WO2007127821A3 (fr) 2008-12-24

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Cited By (1)

* Cited by examiner, † Cited by third party
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US10863276B2 (en) 2015-08-03 2020-12-08 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Soundbar

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US5870484A (en) * 1995-09-05 1999-02-09 Greenberger; Hal Loudspeaker array with signal dependent radiation pattern
US6778672B2 (en) * 1992-05-05 2004-08-17 Automotive Technologies International Inc. Audio reception control arrangement and method for a vehicle
US7123731B2 (en) * 2000-03-09 2006-10-17 Be4 Ltd. System and method for optimization of three-dimensional audio

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US6778672B2 (en) * 1992-05-05 2004-08-17 Automotive Technologies International Inc. Audio reception control arrangement and method for a vehicle
US5870484A (en) * 1995-09-05 1999-02-09 Greenberger; Hal Loudspeaker array with signal dependent radiation pattern
US7123731B2 (en) * 2000-03-09 2006-10-17 Be4 Ltd. System and method for optimization of three-dimensional audio

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US10863276B2 (en) 2015-08-03 2020-12-08 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Soundbar

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