Abstract:
A downsampled adaptive filter is used to find the impulse response of a home theater system. Downsampling yields higher maximum measurable distance for given filter length. By using a Least-Mean-Square (LMS) adaptive filter, almost anything can be used as the source noise. While downsampling may decrease the resolution of the distance measurement, Adaptive Filtering allows a much broader range of test signals, as opposed to MLS (Maximum Length Sequence) in which the test signal defines the technique (a pseudo-random Maximum Length Sequence.)

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS  
       [0001]     The present application claims priority from Provisional U.S. Patent Application No. 60/612,474 filed on Sep. 23, 2004 (Cirrus Logic Docket No. 1537-DSP), and incorporated herein by reference. The present application is also a Continuation-In-Part of U.S. patent application Ser. No. 11/002,102 entitled “TECHNIQUE FOR SUBWOOFER DISTANCE MEASUREMENT”, filed on Dec. 3, 2004 (Cirrus Logic Docket No. 1538-DSP), and incorporated herein by reference. 
     
    
     FIELD OF THE INVENTION  
       [0002]     The present invention relates to a method and apparatus for calibrating a home theater system. In particular, the present invention is directed toward a technique for measuring the distance between a speaker and a listener location using a downsampled adaptive filter.  
       BACKGROUND OF THE INVENTION  
       [0003]     Home theater systems, which once were expensive luxury items, are now becoming commonplace entertainment devices. Complete Home Theater systems, known as a Home Theater In a Box (HTIB), are available to consumers at reasonable prices. However, properly setting up such Home Theater systems can sometimes be problematic for the consumer.  
         [0004]     Home theater systems provide a number of components, which may be located in various parts of the room. The components include the home theater receiver/amplifier, front stereo speakers (left and right), rear surround sound speakers (left and right), a center speaker, and a subwoofer. Various other combinations of speakers may also be used, including additional or fewer speakers. One such home theater system is described, for example, in U.S. Pat. No. 5,930,370, issued Jul. 27, 1999 to Ruzicka, incorporated herein by reference.  
         [0005]      FIG. 4  depicts a diagrammatic view of the home theater surround sound speaker system (the surround sound system)  10  arranged in accordance with the principles of the present invention. The surround sound system  10  includes a source of a preferably amplified stereo signal, shown in  FIG. 4  as television set  12 . The stereo audio source may be any of a number of audio signal sources. It should, thus, be noted that the source of a stereo audio signal is represented herein as television  12 , but the audio signal source may also be a stereo receiver, a car stereo, a portable compact disk or tape player, a portable boom-box type stereo, or any other source of a stereo signal.  
         [0006]     Television  12  outputs an amplified audio signal to interconnect module  14  via a multi-conductor cable  16 . Multi-conductor cable  16  typically includes two conductor pairs for conducting the left and right channels of the stereo signal output by television  12  to interconnect module  14 . Interconnect module  14  receives the audio signals from television  12  and assembles the component left and right channel signals for selective distribution to particular component speakers of the surround sound system  10 .  
         [0007]     The component speakers typically include a sub-woofer  18 , which receives full range left and right signals, but only reproduces the low frequency components of the audio signal. Interconnect module  14  also outputs an audio signal to front center speaker  20 . Front center speaker  20  receives both the left and right component signals of the stereophonic signal and reproduces the (L+R) summation signal. Preferably, front center speaker  20  is located in proximity to television  12  and projects the acoustic output of the (L+R) summation signal toward the listener  28 .  
         [0008]     Interconnect module  14  also outputs the left channel signal to left satellite speaker  22  and right channel signal to right satellite speaker  24 . Left satellite speaker  22  and right satellite speaker  24  may be relatively small speakers and need only reproduce mid range and/or high frequency signals. Left and right satellite speakers are preferably oriented so that the primary axis of radiation of the speaker points upward along a vertical axis; however, other orientations of the satellite speakers may also provide satisfactory performance. Interconnect module  14  also outputs an audio signal to rear ambience speaker  26 . Rear ambience speaker  26  typically receives an audio signal in the form of a left channel minus right channel (L−R) or a right channel minus left channel (R−L) difference signal. As will become apparent throughout this detailed description, several embodiments of the invention described herein enable interconnect module  14  to generate a variety of signals to be output to left satellite speaker  22 , right satellite speaker  24 , and/or rear ambience speaker  26 . It should be noted at the outset that the term speaker refers to a system for converting electrical input signals to acoustic output signals where the system may include one or a number of crossover networks and/or transducers.  
         [0009]     The components described in  FIG. 4  typically are arranged to optimize the surround sound effect to enhance the listening experience of the viewer  28 . The viewer  28  typically faces television  12  which has front center speaker  20  arranged in proximity to television  12  so that center speaker  20  and television  12  radiate their respective audio and video output in the general direction of viewer  28 . The left satellite speaker  22  typically is arranged to the left side of viewer  28  while right satellite speaker  24  is arranged to the right side of viewer  28 , both satellite speakers typically being located nominally midway between the viewer  28  and television  12 . Rear ambience speaker  26 , which contributes to creating a spacious audio effect, is typically located behind viewer  28 . Rear ambience speaker  26  is depicted as a single speaker, but multiple rear speakers  26  may be included in the system.  
         [0010]     One problem with such systems is that a major aspect of acoustical sound reproduction may depend upon the relative location of each of the speakers in a room, relative to the preferred listening area, as well as room acoustics, speaker orientation, and the like. These aspects are largely outside the control of the manufacturer, as speaker placement can only be suggested by the manufacturer, and room configuration or other criteria may alter such placement by the consumer. In addition, the size of a room in which the system is setup is impossible to predetermine, and thus a great variance results in the placement, orientation, and location of speakers, as well as their relative distance from the preferred listening area and the receiver/amplifier.  
         [0011]     One Prior Art approach for high-end home theater systems has been to hire a skilled acoustician to setup the home theater system. Such a skilled technician can adjust the location and placement of the speakers, and using various components, (adjustable delays, equalizers, and even passive acoustical components), optimize the sound quality for a particular room. Unfortunately, hiring an acoustician to fine tune a home theater system is expensive. Many “consumer grade” home theater systems sell for only a few hundred dollars, which is far less than the cost of even one in-home visit by an acoustician.  
         [0012]     Another approach has been to provide a built-in system for measuring the relative time delay (e.g., location) of speakers within a room using a microphone and some processing equipment so that a consumer can calibrate the system for a given room. Such a system has many advantages, as it reduces the overall cost of installation, provides a better acoustical response to the system (resulting in fewer consumer complaints) and also allows the system to be easily moved to new locations.  
         [0013]     While a number of such systems exist in the present market, one such system is illustrated, for example, by U.S. Pat. No. 6,655,212, issued on Dec. 2, 2003 to Ohta (hereafter “Ohta”), and incorporated herein by reference.  FIG. 1  is a diagram from Ohta, illustrating a configuration of a measurement system including the sound field measuring apparatus. Measurement system  100  comprises a number of components. DSP (Digital Signal Processor)  1  outputs a test signal to D/A converters  2   a ,  2   b , etc. Amplifiers  3   a ,  3   b , etc. receive signals output from D/A converters  2   a ,  2   b , etc. and drive speakers  4   a ,  4   b , etc. Microphone  6  is disposed at a predetermined position (listening position) in an acoustic space  5  where the speakers  4   a ,  4   b , etc. are placed. Amplifier  7  amplifies a signal output from microphone  6  and outputs the signal to A/D converter  8 .  
         [0014]     DSP  1  includes a number of components. Exponential pulse generator  11  generates an output signal to speaker (“SP”) selector  12 , which in turn outputs the signal to a selected one (or more) of D/A converters  2   a ,  2   b , etc. RAM  14  stores a received signal from A/D converter  8 . Calculation section  15  uses the data stored in RAM  14  to calculate the time of arrival of an exponential pulse transmitted via speaker  4   a ,  4   b , etc. Control section  13  operates exponential pulse generator  11  and RAM  14  so as to synchronize start timings. Calculation section  15  includes a rising emphasizing section  151 , a time detecting section  152 , and a calculating section  153 .  
         [0015]     Although not shown, DSP  1  has a signal processing circuit, which, during multi-channel audio reproduction using the speakers  4   a ,  4   b , etc., delays each channel&#39;s signal by a predetermined time period. According to this configuration, the perceived distances between the speakers and the listening position can be made constant by adjusting the time delays to compensate for the actual differences in distance.  
         [0016]     In operation, a system such as that illustrated in  FIG. 1  may send a signal generated by exponential pulse generator  11  (or other sound source) to a speaker  3   a ,  3   b , etc. via speaker selector  12 . Microphone  6  maybe positioned by a consumer at a preferred listening location in the room. Microphone  6  receives the exponential pulse (or other sound) from speaker  3   a ,  3   b , etc. and transmits this signal, via amplifier  7  and A/D converter  8  to RAM  14 . Calculating section  15  may then measure the time delay between the output of the sound pulse from speaker  4   a ,  4   b , etc. and the reception at microphone  6 , and thus calculate the relative distance of the speaker from the preferred listening position. This value may be displayed to the user as a physical distance, and/or may be used as a time delay value internally. Each speaker  4   a ,  4   b , etc. is tested in turn and relative time delays calculated. The home theater system can then adjust the relative time delays of each speaker accordingly to provide optimal sound levels at the preferred listening area.  
         [0017]     Ohta employs what may be referred to as “Gated Noise”. The time of arrival is measured by using an impulse signal in the following manner. An impulse signal is output from a speaker. The signal is then detected by a microphone disposed at a predetermined position (listening position), and an impulse response between the speaker and the microphone (listener) is calculated. The time of arrival means a time period from a time when an impulse response is input, to that when an impulse response reaches the maximum peak value. This is basically a threshold-based technique that is very susceptible to errors from background noise (bad Signal-to-Noise Ratio (SNR)). For example, if a loud noise is made in the background during the setup, the system may interpret this sound as the peak value.  
         [0018]     Another solution, known as Maximum Length Sequences (MLS), is described by Douglas D. Rife and John Vanderkooy, AES Vol. 37, No.6, 1989 June, “Transfer-Function Measurement with Maximum-Length Sequences”, incorporated herein by reference. The basic idea is to apply an analog version of an MLS to a linear system, sample the resulting response, and then cross-correlate that response with the original sequence. The result of the cross correlation is the system impulse response. Once the impulse response is obtained, the system delay is simply the location of the initial peak and the phase is the peak&#39;s polarity (in-phase: positive, out-of-phase: negative). Borish and Agell, “An Efficient Algorithm for Measuring the Impulse Response using Pseudorandom Noise”, J. Audio Eng. Soc., Vol. 31, No. 7, July/August 1983, incorporated herein by reference, also discloses how Maximum Length Sequences (MLS) can be used to measure the impulse response of a linear system.  
         [0019]     This system may work well even with poor SNR, but it is limited to MLS noise as the source. Since the MLS sequence must be compared with the received response, it may be necessary to provide enough memory to store the entire MLS sequence for comparison purposes. The MLS signal may then be generated and a record of the signal stored in memory. The received signal may then be compared to this stored signal in order to determine the delay time and other acoustical characteristics of the speaker system. The memory and processing requirements can be prohibitive when measuring long distances. For example, a 1024-sample cross-correlation may be required to measure distances as large as 20 feet at a 48 kHz sampling frequency.  
         [0020]     One seemingly minor problem with MLS techniques when used for audio calibration, is that the MLS signal, when played over a speaker system, produces an unpleasant audio sound which consumers may find annoying. When using an MLS signal to calibrate a home theater system, for example, a loud static-like noise may be produced from each speaker during the calibration process. Bystanders and even users may find this noise unpleasant and even uncomfortable. It would be advantageous to provide a system where more pleasing test signals could be used. It would also be desirable to provide a system whereby a characteristic signal could be used that may be indicative of manufacturer source (e.g., as a Trademark) or could produce musical or vocal renderings that could be pleasant to hear or even instructive (e.g., as part of the calibration process).  
         [0021]     The general concept of Adaptive Filtering is known in the art.  FIG. 2  is a block diagram illustrating the basic concepts of a single input, single output adaptive filter of the Prior Art. Adaptive Filtering is described in detail by Widrow and Stearns. “Adaptive Signal Processing” (1985, Prentice-Hall, Englewood Cliffs, N.J., ISBN 0-13-004029-0) incorporated herein by reference. The adaptive filtering system of  FIG. 2  includes an input signal X k    400  driving both the unknown plant  410  and the adaptive model  430 . Output d k    460  from the unknown plant  420 , is fed back and combined in subtractor  440  with the output Y k  of the adaptive model to produce an error signal, which in turn adjusts the adaptive model  430  in a feedback loop.  
         [0022]     In many practical cases, the unknown plant to be modeled  420  is noisy, that is, has internal random disturbing, forces. In  FIG. 2 , this noise is represented by plant noise  410  combined with the output of unknown plant  420  in adder  450 . When the adaptive model  430  has enough flexibility to match the dynamic response of the unknown plant, its output will perfectly match that of the unknown plant  420  except for plant noise n k .  
         [0023]     Internal plant noise  410  appears at plant output  460  and is commonly represented there as an additive noise. This noise is generally uncorrelated with the plant input  400 . If adaptive model  430  is an adaptive linear combiner having weights adjusted to minimize mean-square error, the least-squares solution will be unaffected by the presence of the plant noise  410 . The convergence of the adaptive process will be unaffected by plant noise  410 , and the expected weight vector of the adaptive model  430  after convergence will be unaffected. The least-squares solution will be determined primarily by the impulse response of the plant to be modeled. The impulse response of the plant to be modeled may also be significantly affected by the statistical or spectral character of the plant input signal  400 .  
         [0024]     Thus, it remains a requirement in the art to provide a technique for measuring distance (delay) for a speaker whereby a noise source other than a MLS may be used to measure impulse response. It remains a further requirement in the art to provide a technique for measuring home theater system response while reducing memory requirements and/or increasing the available range of measurement for a given memory (MIPS) requirement.  
       SUMMARY OF The INVENTION  
       [0025]     The present invention uses a downsampled adaptive filter to find the impulse response. Downsampling yields higher maximum measurable distance for given filter length (e.g., down sample by two, and the maximum distance measurable doubles). By using a Least-Mean-Square (LMS) adaptive filter, almost anything can be used as the source noise. Downsampling does decrease the resolution of the distance measurement, but at 48 kHz downsampling by four changes, the resulting resolution may change, for example, from approximately 0.25 inch to approximately one inch, which for home theater calibration is not a significant change.  
         [0026]     Maximum Length Sequence (MLS) and Adaptive Filtering are two separate techniques. In the preferred embodiment of the present invention, the particular form of Adaptive Filtering is a least-mean-square algorithm, or LMS. Adaptive Filtering allows a much broader range of test signals, as opposed to MLS where the test signal defines the technique (a pseudo-random Maximum Length Sequence.) With Adaptive Filtering, theoretically, almost any kind of test signal (music, speech, swept tones, noise, or the like), could be used. In one embodiment, a noise burst is used, but one that has a much more “pleasing” sound than the MLS techniques of the Prior Art.  
         [0027]     The system of the present invention may perform two downsampling operations. Adaptive Filtering works by allowing a filter to change its coefficients based on the error between the test signal and the received signal, so both need to be at the same sample rate. The test signal sent to a speaker may be downsampled by a predetermined factor. This signal is also sent to the adaptive filter. A Digital to Analog Converter (DAC) converts the signal to analog form, which in turn drives the speaker.  
         [0028]     A signal detector (e.g., microphone) receives the speaker signal, and an Analog to Digital Converter (ADC) converts the signal to digital form. The digital signal is also downsampled by the same predetermined factor and sent to the adaptive filter, so that both the input and output signals have the same sample rate.  
         [0029]     A separate downsampling step may not be required if Digital to Analog Converters (DACs) and Analog to Digital Converters (ADCs) are available which support lower sampling rates (e.g., 48 kHz/4). However, since such DACs and ADCs are not widely available, the downsampling may be performed in a separate operation. In an alternative embodiment, downsampling may be applied similarly to a system using a MLS signal.  
         [0030]     Adaptive filtering itself is known in the art and has been described previously with regard to  FIG. 2 . As noted above, adaptive filtering is a well-known concept in the digital filtering arts, and is basically a digital version of an analog feedback loop. Basically, the filter “adapts” to model the response of the “plant” (a system having unknown characteristics). From the adaptive model, it is possible to model the behavior of the system. However, it does not appear that the concept of the adaptive filter has been applied to the Home Theater calibration problem.  
         [0031]     In the present invention, once the system is modeled with the adaptive filter, the characteristics of the system are known, and from this, system parameters, such a speaker placement in the room (here the “plant” being the speaker/room/home theater combination) can be determined. As illustrated in  FIG. 2 , an adaptive filtering system is relatively immune from background noise (in this application, people talking in the room, random noises, etc.). Thus, using an adaptive filter in this application results in a better measurement that is less susceptible to noise.  
         [0032]     One additional bonus of using an adaptive filter for the home theater applications, is that the annoying MLS signals of the Prior Art, which sound like very loud static bursts, do not need to be used. In fact, almost any noise can be used from a soothing tone, to even a musical sound or verbal instruction.  
         [0033]     A further aspect of the present invention is that by downsampling by 4, a better range of measurement is obtained. As noted above, for a 20 foot range (speaker distance from measuring point) a certain signal length may be required (to be stored and compared), which in turn requires a certain amount of memory.  
         [0034]     By downsampling by four, the range is extended to 80 feet, which in the modern “mini-mansions” of today, is not an unheard of distance. The tradeoff is that the “granularity” or resolution of measurement goes from ¼ inch to 1 inch. However, since even minor movement of the user&#39;s head can exceed 1 inch, this tradeoff is considered well worthwhile. 
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0035]      FIG. 1  is a Prior Art diagram illustrating a configuration of a measurement system including a sound field measuring apparatus.  
         [0036]      FIG. 2  is a block diagram illustrating the basic concepts of adaptive filtering in the Prior Art.  
         [0037]      FIG. 3  is a block diagram of the apparatus of the present invention.  
         [0038]      FIG. 4  is a block diagram of the home theater surround sound speaker system. 
     
    
     DETAILED DESCRIPTION OF THE INVENTION  
       [0039]     In the system of  FIG. 2 , the unknown plant refers to a general electrical system or circuit. The plant noise refers to any general intervening noise of such a plant. Applying the system of  FIG. 2  to the problem of home theater calibration, the “unknown plant”  420  would, in this application, comprise the response of the room and speaker(s), and the “plant noise”  410  is any noise in the room (A/C, refrigerator, speech, and/or other background noise). The use of an adaptive filter for home theater calibration may make such calibration less sensitive to background interference and thus not require absolute quietness in the room for the consumer in order to perform the calibration.  
         [0040]      FIG. 3  is a block diagram of the apparatus of the present invention employing an adaptive filter for home theater calibration. For the sake of simplicity, many of the basic components in an auto-setup home theater system are not illustrated here. Referring to  FIG. 3 , a noise source  210  may be used to generate a sound pattern or series of impulses or the like. As discussed above, this sound source could comprise a number of sound patterns generated from a stored sound pattern or generated spontaneously. Since the choice of sound pattern is somewhat flexible (as opposed to MLS systems), a more pleasing sound to the consumer may be selected. A digital to analog converter (DAC) converts this digital sound pattern into an analog signal, which is then driven through speaker  220 .  
         [0041]     The digital signal from noise source  210  may also be downsampled (e.g., by a factor of  4 ) in downsampler  270 , and the resultant signal sent as an input to adaptive filter  280 . Downsampling, as discussed above, reduces the need to store large segments of data in adaptive filter  280  for signal comparison. To measure speaker distances between 0 and 20 feet may require 1024 or more digital samples to properly correlate the two signals (plant input and plant output) in order to effectively measure time delay, which in turns yields distance. The further apart the speakers are located, the more samples are generally required for comparison, as the correlation between the two responses may be further apart in time.  
         [0042]     At a 48 KHz sampling frequency, as used in the Prior Art, this resultant granular resolution of speaker distance measurement is about one quarter inch, much more than is required for home theater calibration. A user can move their head several inches even while sitting in one place, so it makes no sense to provide for such fine granularity in distance measuring. The present invention sacrifices this unnecessary calibration accuracy for increased distance measurement capability, without increasing the memory (sample) requirements of the adaptive filter.  
         [0043]     Thus, for example, by downsampling by a factor of four (4), the granularity may be increased to one inch (more than acceptable for home use) while the overall distance range is increased by a factor of four (e.g., to 80 feet), while using the same number of samples (1024). Given the cavernous nature of many new homes, such a distance range may be required for successful home theater calibration. Of course, other numbers of samples, granularities, downsampling rates, and distance ranges may be used within the spirit and scope of the present invention. The conversion from samples to distance is based on the speed of sound at sea level. Changes in altitude, temperature and humidity slightly affect the speed of sound, but only by hundredths of inches per sample at 48 kHz samplerate.  
         [0044]     Referring again to  FIG. 2 , microphone  240  may be located by the consumer at a preferred listening location (e.g., near the head of the consumer at a favorite chair or the like). Microphone  240  picks up noise or other sound from speaker  230 , which will be delayed by an amount of time equal to the speed of sound divided by the distance between microphone  240  and speaker  230 . Other internal delays may, of course, exist within the electronics of the system, but such delays are minor and uniform and can be easily compensated for and are not affected by speaker location.  
         [0045]     The output of microphone  240  may then be converted into a digital signal in analog to digital converter (ADC)  250 . Output of ADC  250  is downsampled by the same factor as downsampler  270  (e.g.,  4 ) and the output fed as the plant output to adaptive filter  280 . The output of adaptive filter  280  generates an impulse response  290 , which in turn provides a value indicative of the distance between speaker  230  and microphone  240 .  
         [0046]     In co-pending application Ser. No. 11/002,102, incorporated herein by reference, and from which the present application claims priority, calculation of speaker distance may be achieved by measuring the location of the impulse response peak as well as from the width of this peak at a given level, or by combining these two values using a polynomial equation, lookup table, or the like.  
         [0047]     With the apparatus of the present invention, almost any sound could be used as a noise source. Thus, a characteristic pleasing sound may be used, which may also be indicative of a product or system source, much as the THX sound is used in movie theaters to inform audience members of the sound system type. Alternately, a voice instruction may be used to help the consumer understand the process (e.g., “now calibrating, left speaker”). Similarly, a consumer provided sound source (e.g., CD or the like) may be used such that the system can be calibrated without having to interrupt the playback of a CD, DVD, or other audio source. By pressing a button on a remote, the system could calibrate each speaker (selectively) without having to interrupt the audio being played at the time.  
         [0048]     In an alternative embodiment, once the impulse response of the system has been measured (either directly, or by MLS or LMS), mathematical operations can be performed on the impulse response data using, e.g., a Fast Fourier Transform (FFT) to obtain the magnitude and phase response of the system (room plus speaker). Many Prior Art systems using MLS already do this; however, the low-frequency resolution of the resultant response is not favorable. The impulse response from LMS may also be converted to obtain the magnitude and phase response in a similar manner to Prior Art MLS systems. Downsampling, however, reduces the frequency range of the response. Thus, in such an embodiment, downsampling may be reduced or eliminated to limit this reduction.  
         [0049]     The term “phase” may be used in two slightly different ways. First, the “phase of the speaker” is used to refer to the polarity, that is, which way the two wires are connected. This version of “phase” may take one of two values, either “in-phase” or “out-of-phase”. Second, the “phase response” of the speaker in the room is a function of frequency, like the power spectrum. The power spectrum, or magnitude response (often inaccurately just called the “frequency response”) is the power level (Y-axis, usually in dB) plotted against frequency (X-axis in Hz). The phase response is also a function of frequency.  
         [0050]     Also, the magnitude and phase responses are really two halves of the overall “Response” of the system. Sometimes this term “Response” may be referred to by the phrase “the magnitude and phase response”, which is singular term instead of a plural term, as the response that contains both magnitude and phase information before they are separated into two responses. In some instances, the term “frequency response” may be used to indicate the total response (magnitude and phase).  
         [0051]     In the present invention, downsampling actually helps in finding the phase (polarity) of the speaker (the first definition of “phase” noted above). However, if one wanted to know the complete magnitude and phase response of the speaker, (the second definition of “phase” noted above) downsampling may reduce the upper ¾ of the response. Thus, for this type of Phase response measurement, it may be advisable to reduce or eliminate downsampling.  
         [0052]     While the preferred embodiment and various alternative embodiments of the invention have been disclosed and described in detail herein, it may be apparent to those skilled in the art that various changes in form and detail may be made therein without departing from the spirit and scope thereof.