Title of Invention

"AUDIO SIGNAL PROCESSING"

Abstract

An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed and signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L - R signals.

Full Text

TITLE AUDIO SIGNAL PROCESSING BACKGROUND OF THE INVENTION The invention pertains to audio signal processing and more generally to methods for processing two channel audio signals to create more than two output channels SUMMARY OF THE INVENTION In one aspect of the invention, a method for processing two input audio channel signals to provide n output audio channel signals where n>2, includes dividing the first mput^hannel signal and the second input channel signal into a plurality of corresponding non-bass frequency bands, measuring the amplitude of the audio signal m the two input channels in one the frequency bands to provide a first channel first frequency band audio signal and a second channel first frequency band audio signal to provide a first channel first frequency band audio signal amplitude and a second channel first frequency band audio signal amplitude, determining the correlation between the first channel first frequency band audio signal and the second channel first frequency band audio signal to provide a first frequency band correlation, scaling the first channel first frequency band audio signal by a first factor (a(first)) related to the first frequency band correlation and further related to the first channel first frequency band audio signal amplitude and the second channel first frequency band audio signal amplitude, the scaling to provide a first scaled first output channel first frequency band audio signal first portion, scaling the second channel first frequency band audio signal by a second factor (a(second)) related to the first frequency band correlation and further related to the first channel first frequency band audio signal amplitude and the second channel first frequency band audio signal amplitude, the scaling to provide a first scaled first output channel first frequency band audio signal second portion, combining the fust scaled first channel first frequency band audio signal first portion and the first scaled first channel first frequency band audio signal first portion to provide a first frequency band portion of a center channel output audio signal The method may further include scaling the first channel first frequency band audio signal by a third factor, which may be =√1- a{first)' to provide a first frequency band portion of a left channel output signal The method may further include combining the first frequency band portion of the left channel output audio signal with a second frequency band portion of the first channel audio signal to provide a left non-bass audio signal The frequency bands may be time varying The first frequency band may be the speech band The two input audio channel signals comprise compressed audio signal data The compressed audio signals may be in a non-reconstructable data format, which may be the MP3 format In another aspect of the invention, a method for processing two input audio channel signals to provide n output audio channel signals wherein n > 3 and wherein the n output channel signals include surround channels includes separating the two input channels into a plurality of corresponding non-bass frequency bands, processing each of the plurality of input channel non-bass frequency bands to provide the corresponding frequency band of a center channel output signal and two non-sun ound non-centei output channel signals, processing at least one of the two non-center non-surround output channel signals to provide a surround output channel signal, wherein the processing the two non-center channel output signals does not include processing a signal representing the difference between the two input channels The processing the two non-center channel output signals comprises at least one of time delaying, attenuating, and phase shifting one of the two non-center input channel signals In another aspect of the invention, a method for processing two input audio channels to provide n output audio channels where n>2, includes dividing the first input channel signal and the second input channel signal into a plurality of corresponding non-bass frequency bands, processing according to a first process a first input channel first frequency band audio signal to provide a first portion of a first frequency band of a center output channel signal, processing according to a second process a input channel fust frequency band audio signal to provide a second portion of the first frequency band of the center output channel signal, processing according to a third process a first input channel second frequency band audio signal to provide a first portion of a second frequency band of the center output channel signal, and processing according to a fourth process a second input channel second frequency band audio signal to provide a second portion of the second frequency band of the center output channel signal, wherein the third process is different from the first process and the second process and wherein the fourth process is different from the first process and the second process The method may further include processing according to a fifth process the first input channel first frequency band audio signal to provide a first portion of a first frequency band of a non-center output channel signal, and processing according to a sixth process the first input channel second frequency band audio signal to provide a first portion of a second frequency band of the non-center output channel signal, wherein the fifth process is different from the sixth process The first process may include scaling the first input channel first frequency band audio signal by a factor a The fifth process comprises scaling the first input channel first frequency band audio signal by a factor √1-a The sixth process may include providing the unattenuated first input channel second frequency band audio signal so that the center output channel signal comprises the first input channel first frequency band audio signal scaled by a and so that the non-center output channel comprises the first input channel first frequency band signal scaled by √l-a and the unattenuated first input channel second frequency band signal The third process may include providing none of the first input channel second frequency band audio signal to provide a first portion of a second frequency band of the center output channel signal so that the center output channel signal comprises the first input channel first frequency band audio signal scaled by a and no portion of the first input channel second frequency band audio signal The sixth process may include providing the unattenuated first input channel first frequency band audio signal At least one of the first process, the second process, the thud process, or the fourth process may be time varying In still another aspect of the invention, a method for processing two input audio channel signals to provide n output audio channel signals wherein n>2 and wherein the two input audio channel signals comprise unreconstructable compressed audio signal data, the method includes separating the input audio channel signals into frequency bands, separately processing the frequency bands, and combining the separately processed frequency bands to provide the n output audio channels The separately processing the frequency may include scaling a first channel first frequency band signal, scaling a second channels first frequency band signal, and wherein the separately processing does not include processing a signal representing the difference between any portions of the first input audio channel signal and the second audio channel signal Other features, objects, and advantages will become apparent from the following detailed description, when read in connection with the following drawing, in which BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWING FIGS 1A and IB are block views of audio systems, FIG 2 is a block diagram of a decoding and playback system, FIG 3 is a block diagram of a filter network, FIG 4 is a block diagram of an audio system showing steering circuitry in greater detail, FIGS 5 A and 5B are block diagrams of audio systems showing implementations of the steering circuitry of FIG 4, FIGS 6A - 6C are plots showing the behavior of a first steering circuit, and FIGS 7A - 7C are plots showing the behavior of a second steering circuit DETAILED DESCRIPTION Though the elements of several views of the drawing are shown and described as discrete elements in a block diagram and are referred to as "circuitry", unless otherwise indicated, the elements may be implemented as one of, or a combination of, analog circuitry, digital circuitry, or one or more microprocessors executing software instructions The software instructions may include digital signal processing (DSP) instructions Unless otherwise indicated, signal lines may be implemented as discrete analog or digital signal lines, as a single discrete digital signal line with appropriate signal processing to process separate streams of audio signals, or as elements of a wireless communication system Some of the processing operations are expressed in terms of the calculation and application of coefficients The equivalent of calculating and applying coefficients can be performed by other signal processing techniques and are included within the scope of this patent application Unless otherwise indicated, audio signals may be encoded in either digital or analog form Referring to FIGS 1A and IB, there are shown two audio systems In FIG 1A, a stereo audio signal source 2A is coupled to an x or x 1 channel decoding and playback system 8 The decoding and playback system 8 has a plurality x of audio channels, including a center channel and at least one surround channel Typically x is 4 or 5, but may be more The decoding and playback system may also have a low frequency effects (LFE) channel, as indicated by the " 1" The decoding and playback system 8 receives stereo audio signals from the stereo audio signal source 2A and processes the stereo audio signals in a manner to be described below to provide the x channels Many decoding and playback systems that process stereo audio signals to provide additional channels introduce undesirable acoustic effects into one or more of the channels of the x or x 1 channel playback Some decoding and playback systems may separate and process an L - R signal to create the sunound channels An "L - R signal" lefers to a signal that is the difference between the L (left channel) signal and the corresponding R (right channel) signal In some instances, a difference between an L and an R signal, present in material created for stereo reproduction, may result from an acoustic effect desired by a content creatoi which was not intended to be radiated from surround speakers In some conventional sunound audio systems, L - R signals are interpreted as intended to be radiated by surround speakers If L - R signals of a conventionally created stereo recording are interpreted as intended to be radiated by surround speakers, sound that is intended to come from in front of the listener may appear to come from behind the listener If the L - R signal is used to create the surround speaker signals, vocal sounds may not be well anchored or spatial effects may be altered from what was intended by the content creator, or audible artifacts may appear In FIG IB, an audio signal data compressor 4 leceives audio signal data from an audio signal source 2B and compresses the audio signal data and stores the compressed audio signal data in a compressed audio signal data storage device 6 A decoding and playback system 8 decodes the compressed audio signals, processes the audio signals to provide the x channels, and transduces the decoded audio signals to acoustic energy The audio signal source 2A may be a conventional stereo device, such as a CD player or may also be stereo radio signals received by an AM or FM radio receiver, an 1BOC (in-band on channel) radio receiver, a satellite radio receiver, or an internet device The audio signal source 2B may likewise be a conventional stereo device such as a CD player, but may also be a multi-channel audio source The audio signal data compressor 4 may be one of many types of audio signal data compressors that (if necessary downmix the multi-channels to two channels and) compress audio signal data so that the audio signal data can be transmitted more quickly and with less bandwidth, or stored in significantly less memory, or both, than uncompressed audio signal data Some compressors compress the data in non-reconstructable or "lossy" manner, that is they compress the signals in a manner such that some information is discarded so that the original signal data cannot be exactly recreated by the decoding and playback system 8 One class of such devices uses the so-called MP3 compression algorithm Compressors using the MP3 algorithm typically store the audio signal on a storage device 6 such as a hard disk, the stored audio signal may then be copied to another storage device such as a hard disk on a portable MP3 player or may be decoded and transducer by a decoding and playback system 8 Since lossy compressors may discard data, the audio signal stored on the storage device may have undesirable artifacts that can be transducer into acoustic energy The compression algorithm may therefore be configured so that the artifacts are masked and are therefore substantially inaudible when played on a conventional stereo system Many algonthms, such as the MP3 algorithm, are designed to provide two channel (typically stereo L and R) audio signals to the storage device When the compressed audio signals are decoded and transduced by a stereo playback device, artifacts resulting from the discarding of data are substantially inaudible due to masking, as stated above Some playback systems, however, have more than two channels, for example in addition to the left and right channels, a center channel and one or more surround channels Some of these multichannel playback systems have signal processing circuitry that processes the two channels to provide additional channels, such as a center channel and one or more surround channels Sometimes however, the processing of the two channels to provide additional channels causes the artifacts created by the discarding of data to become unmasked so that they are audible and annoying One example of how the processing of the two channels to provide additional channels can cause the unmasking of artifacts is when a difference operation (i e generating an L - R signal) is used to create the additional channels In audio signals compressed by algonthms such as the MP3 algorithm, the difference signal of the de-compressed L and R signals (1 e signals that are the result of passing through a lossy compression and de-compression process) may not be representative of the difference between the uncompressed L and R input signals Instead, a significant portion of the difference between the de-compressed L and the R signals may be artifacts resulting from the discarding of data by the compression algorithm Some of the content that was common to the de-compiessed L and R signal may have been necessary to mask artifacts If this common content is removed by a difference operation (1 e creating a signal that is the difference of the de-compressed L and R signals), the artifacts may become unmasked and therefore audible Stated differently, the de-compressed L and R signals each contain artifacts, but the signal to artifact ratio (analogous to a signal to noise latio) is sufficiently high that the artifacts are not audible Extracting the common content by performing a difference operation on the de-compressed signals may remove significant signal content, so the signal to artifact ratio is significantly lower and the artifacts are audible Referring to FIG 2, there is shown a decoding and playback system 8 The decoding and playback system 8 includes two input terminals 1OL and 1 OR, each communicatingly coupled to a filter network 12L and 12R, respectively The filter networks 12L and 12R are coupled to steering circuitry 40 by n signal lines designated LI - Ln and Rl - Rn, respectively Steering circuitry 40 is coupled to loudspeakers 20L (left), 20LS (left sunound), 20C, (center), 20R (light) and 20RS (right surround) Loudspeakers 20L, 20LS, 20C, 20R, and 20RS collectively may be referred to as loudspeakers 20 below The filter networks 12L and 12R may also be coupled to bass processing circuitry 42, which may be coupled to bass loudspeaker 44 Some elements, such as amplifiers and digital to analog converters, that are typically present in audio systems, are not shown in this view In operation, a channel (such as a left channel) of an audio signal stream (which may be a stream of compressed audio signals, a stream of broadcast audio signal, a stream of conventional stereo signals, etc ) is received at terminal 10L and split by filter network 12L into n frequency bands The filter network 12L may also separate a bass frequency band A second channel (such as a right channel) of an audio signal is received at terminal 10R and split by filter network 12R into n frequency bands The filter network 12R may also separate a bass frequency band Steering circuitry 40 processes the several frequency bands of the left and right signals and re-combines the frequency bands to form output multi-channel audio signals, which are transmitted to loudspeakers 20 for transduction into acoustic energy The multiple channels may include surround channels For simplicity, the audio signal formed by the steering circuitry to be transmitted to the left speaker will be hereinafter refened to as the "left speaker signal " Similarly, the signal to be transmitted to the center speaker will be referred to as the "center speaker signal", the signal to be transmitted to the right speaker will be referred to as the "light speakei signal", the signal to be transmitted to the left sunound speaker will be referred to as the "left surround speaker signal" and the signal to be transmitted to the right surround speaker will be referred to as the "light surround speakei signal " Steering circuitry 40 may operate on each frequency band by scaling a signal by a scaling factor and routing the scaled signal to an output channel, in some embodiments through a summer that sums signals from several frequency bands to form an output channel signal The scaling factor may have a range of values, such as between zero (indicating complete attenuation) and one (unity gain) as in one of the examples below Alternatively, the scaling factor may have a range other than zero to one or may be expressed m dB Conventional audio systems may also provide a user with balance or fade controls to allow a user to control the amount of amplification of the signals in individual speakers or in groups of speakers More specific descnptions of the operation of the steering ciicuitry 40 will be explained below Referring now to FIG 3, there is shown a circuit suitable for filter network 12L or 12R of FIG 2 Input teimmal 10L is coupled in parallel to low pass filter 25, band pass filters 27A and 27B, and high pass filter 28 The output signal of low pass filter 25 is frequency band LI, the output signal of band pass filter 27A is frequency band L2, the output signal of band pass filter 27B is L3, and the output signal of high pass filter 28 is frequency band L4 The filter networks of FIG 3 is exemplary only Many other types of digital or analog filter networks can be employed The behavior of the steering circuitry 40 of FIG 2 can be determined and implemented in a number of ways The desired behavioi can be determined subjectively, for example by listening tests, or objectively for example by a predetermined measurable response to test audio signals, or by a combination of subjective and objective methods The desired behavior may be implemented by some sort of algebraic equation or set of equations, a look-up table, oi by some sort of rules based logic, or by some combination of algebraic equations, look-up table, and rules based logic The algebraic equation or set of rules may be simple or may be complex, for example the behavior of the steering circuitry applied to one spectral band could be affected by conditions in an adjacent band Each of spectral bands (for example band Ll/Rl, band L2/R2, band L3/R3 etc of FIG 2) can be treated differently, and each band can have a different behavior applied to it by the steering circuitry The behavior of each band can vary over time The behavior can be expressed in an algebraic equation, where the values of the vanables (such as a correlation coefficient, described below) for each frequency band can lesult in the same algebraic equation resultmg in different behavior in different frequency bands The values of the variables may be time varying, resulting in changing behavior for each band over time and in the behavior of one frequency band differing from the behavior of another frequency band Additionally, different equations may be used to control the behavior in different bands The behavior applied by the steering circuitry can include making no modification at all to one or more of the bands, which can be indicated by a scaling factor of one, the behavior can also include significantly attenuating the signal for one oi more of the bands, which could be indicated by a scaling factor of zero Referring now to FIG 4, there is shown a decoding and playback system 8, with steering circuitry 40 shown in more detail The LI output terminal of filter network 12L and the Rl output of filter network 12R are coupled to band 1 steering logic block 46-1 The L2 output terminal of filter network 12L and the R2 output of filter network 12R are coupled to band 2 steenng logic block 46-2 Similarly, each of the output terminals of filter network 12L and a corresponding output terminal of filter network 12R are coupled to a steering logic block For clarity, only steering logic 46-1 and 46-2 are shown in this view Each of the steering logic blocks, such as 46-1 and 46-2 are coupled to one or more summers 18LS, 18L, 18C, 18R, and 18RS For clarity, only signal lines from band 1 and band 2 steenng logic blocks 46-1 and 46-2 and signal line to summer 18C are shown Output signal lines to summers 18LS, 18L, 18C, 18R, and 18RS are shown, howevei, depending on the steering logic, signal lines to one or more of the summers may be omitted Input lines from center summer 18C shows inputs from all frequency bands, depending on the steering logic, signal lines form one or more of the steenng logic blocks may be omitted Summers 18LS, 18L, 18C, 18R, and 18RS are coupled to speakers 20LS, 20L, 20C, 20R, and 20RS, respectively If there is only one signal line to one of the summers, the summei can be omitted and the signal line can couple directly to the speaker In operation, a steenng logic block such as 46-1 or 46-2 for a frequency band applies logic to the left and right frequency band audio signals The logic applied by a steenng logic block such as 46-1 may differ from the logic applied by steenng logic block 46-2 and from the steering logic blocks associated with the othei frequency bands The logic may be in the form of an equation that yields different results for each channel portion of each frequency band, or may be in the form of different equations for each frequency band Each logic block outputs processed audio signals to one or more of the summers 18LS, 18L, 18C, 18R, and 18RS The summei s 18LS, 18L, 18C, 18R, and 18RS sum the signals from the frequency bands and output audio signals to an associated speaker for transduction to acoustic energy The audio system may have circuitry for processing bass range frequencies, and may have a separate speaker for bass range frequencies One example of circuitry for processing bass range frequencies is described in U S Patent App 09/735,123 Referring now to FIG 5 A, there is shown an implementation of the audio signal processing system of FIG 4 In the implementation of FIG 5 A, the filter network has four output terminals for each of four spectral bands (LI, L2, L3, and L4, and Rl, R2, R3, and R4, of the left and right channels, respectively) Each logic block includes an amplitude detector 24-1, a correlation detector 26-1, a scaling operator such as 14L-1 coupling an output terminal such as LI to left summer 18L, a scaling operator such as 16L-1 coupling an output terminal such as LI to center summer 18C, a scaling operator such as 14R-1 coupling an output tenninal such as Rl to light summei 18R, and a scaling operator such as 16R-1 coupling an output tenninal such as Rl to centei summer 18C Logic blocks for the other frequency bands have similar components, not shown in this view Left summer 18L is communicatingly coupled to left speaker 20L and is communicatingly coupled through transfer function block 22LS to left surround speaker 20LS Right summer 18R is communicatingly coupled to nght speakei 20R and is communicatingly coupled through transfer function block 22RS to right suiround speaker 20RS In operation, a left channel signal is received at input terminal 10L and split into frequency bands LI, L2, L3, and L4 and optionally a bass frequency band A nght channel signal is received at input terminal 10R and split into frequency bands Rl, R2, R3, and R4 and optionally a bass frequency band Each of left channel frequency bands LI, L2, L3, and L4 is processed with a corresponding right channel frequency band Rl, R2, R3, and R4 respectively, by a correlation detector 24-1 and an amplitude detector 26-1 Amplitude detector 26-1 measures the amplitude of the left LI band signal and the right Rl band signal, and provides information to scaling operators such as 14L-1 and 16L-1 as will be desci ibed later Similar amplitude detectors not shown measure the amplitude of the conesponding L and R signal lines, such as L2/R2, L3/R3, and L4/R4 The conelation detector 24-1 compares the signals on signal lines LI and Rl and provides correlation coefficient c/ Similar correlation detectors compare the signals on signals lines L2/R2, L3/R3, and L4/R4 and provide correlation coefficients c2, c3 and c4 "Correlation" refers to the tendency of the signals to vary together over time Conelation can be determined in a number of different ways For example, in a simple fonn, two signals can be compared over a coincident period of time Correlation could be the tendency of the two signals to vary together over that period of time A typical interval of the coincident period of time is a few milliseconds In a more sophisticated form of con elation detection the data may be smoothed to prevent aberrant conditions from unduly influencing the correlation calculation, or the tendency of the two signals to vary together may be measured over similar but non-concurrent intervals of time So, for example, two signals that vary m the same way over time, but phase shifted or time delayed could be considered cor related The amplitude and polanty of the signals may or may not be considered in determining correlation The simpler forms of determining correlation require less computational power than other forms, and for many situations produces results that are not audibly different than other fomis The degree of correlation is typically defined by a correlation coefficient c calculated according to a formula Typically if the correlation coefficient calculation formula yields a result of zero or near zero, the signals are said to be uncorrected If the correlation coefficient calculation formula yields a result of one or near one, the signals are said to be correlated Some correlation coefficient formula calculations may allow the correlation coefficient to have a negative value, so that a correlation coefficient of minus one indicates two signals that are correlated but out of phase (or in other words, tend to vary inversely to each other) Scaling operator 16L-1 scales the left lower frequency band signal by a factor related to the correlation coefficient c/ and to the relative amplitudes of the signals on signal lines LI and Rl The resultant signal is transmitted to summer 18C Scaling operator 14-1 scales the LI signal by a factor related to the coefficient ci and to the relative amplitudes of the signals in signal lines LI and Rl and transmits the scaled signal to summer 18L The Rl signal is scaled at scaling operator 16R-1 by a factor related to the correlation coefficient c/ and to the relative amplitudes of the signals on LI and Rl and transmitted to summer 18C Scaling operator 14R-1 scales the Rl signal by a factor I elated to the coefficient c\ and to the relative amplitudes of the signals in signal lines LI and Rl and transmits the scaled signal to summer 18R Specific examples of determination of scaling factors will be described below Summers 18L, 18C, and 18R sum the signals that are transmitted to them and transmit the combined signal to speakers 20L, 20C, and 20R, respectively The signal from summeis 18L and 18R may also be processed by a transfer function and transmitted to speakers LS and RS, respectively The values of the coefficients are calculated on a band by band basis, so that the values of coefficients may be different for frequency bands Ll/Rl, L2/R2, L3/R3, and L4/R4 Additionally the LI coefficient may be different than the Rl coefficient, the L2 coefficient may be different than the R2 coefficient, and so on The values of the coefficients may vary over time The values of the break frequencies of the filters of the frequency bands may be fixed, or may be time varying based on some factor, such as correlation The equations used to calculate the scaling factors may differ in different bands In one embodiment, speakers 20L, 20R, 20C, 20LS, and 20RS are satellite speakers in a subwoofer-satelhte type audio system The transfer functions 22LS and 22RS may include time delays, phase shifts, and attenuations In other embodiments, transfer functions 22LS and 22RS may be time delays of different length, phase shifts, or amplifications/attenuations, or some combination of time delay, phase shift, and amplification, in either analog or digital form In addition, other signal processing operations to simulate other acoustic room effects can be performed on the signals to speakers 20L, 20R, 20C, 20LS, and 20RS Referring now to FIG 5B, there is shown an example of another audio system embodying elements of the audio system of FIG 4 Left signal input terminal 10L is coupled to filter netwoik 12L Filter network 12L outputs three frequency bands a bass frequency band, and two non-bass frequency bands, one of which is higher than the other and is referred to as a "higher" frequency band and correspondingly, one of which is lower than the other and is referred to as a "lower" frequency band For example, the lower" band could be from the speech band (for example 20 Hz to 4 kHz) and the higher" band could be frequencies above the speech band The output terminal for the bass frequency band is coupled to bass processing circuitiy The lowei non-bass frequency terminals of filter network 12L is coupled to scaling operators 14L-1 and 16L-1 The output terminal of scaling operator 16L-1 is coupled to summer 18C The output terminal of scaling operator 14L-1 is coupled to summer 18L The higher non-bass frequency output terminal of filter network 12L is coupled to summei 18L The output terminal of summer 18L is coupled to speakei 20L and through transfer function 22LS, which in this case is a time delay of 8 ms and a 3 dB attenuation, to speaker 20LS Right signal input terminal 1 OR is coupled to filter network 12R Filter network 12R outputs three fi equency bands similar to the frequency bands output by filter network 12L The output terminal for the bass frequency band is coupled to bass processing circuitry The lower non-bass frequency terminals of filter network 12R is coupled to scaling operators 14R-1 and 16R-1 The output terminal of scaling operator 16R-1 is coupled to summer 18C The output terminal of scaling operatoi 14R-] is coupled to summer 18R The higher non-bass frequency output terminal of filter network 12R is coupled to summer 18R The output terminal of summer 18R is coupled to speaker 20R and through transfer function 22RS, which in this case is a time delay of 8 ms and a 3 dB attenuation, to speaker 20RS Amplitude detector 26-1 and conelation detector 24-1 are coupled to the left lower frequency band filter network output terminal and the right lower frequency band filter output terminal so that they can measure and compare the amplitudes and determine correlation of the left lower signal and the light lower signal as to provide information to the scaling operators to for the calculation of scaling factors The use of rms values for taking into account the relative amplitudes of the signals is convenient, but other amplitude measures, such as peak or average values can be used In one implementation, amplitude detector 26-1 measures the amplitude of the signal of the left lower frequency band signal and the amplitude of the signal of the right lower frequency band signal and provides amplitude information to the scaling operators associated with the frequency band, in this case scaling operators 14L-1, 16L-1,14R-1, and 16R-1 The correlation detector 24-1 compares the signals in the left and right lower frequency band and provides a correlation coefficient c, = (Formula Removed) where Li and R1 are the rms values of L and R of the lower frequency band over a time period, and x is the greater of the mis values of (L + R) oi (L - R) over a period of time Conelation coefficient o can have a value of 0 to 1, with 0 indicating perfectly uncon elated and 1 indicating coi related, m this implementation, phase is not considered in calculating the correlation coefficient The "L" subscript indicates that the correlation coefficient is for the lower non-bass frequency band Scaling operator 16L-1 scales the left lower frequency band signal by a factor a(left), =(Formula Removed) where LPRi is the rms value of (L + R) or (L - R) over a period of time, and Y is the greater of LPRi and LMRi, where LMRL is the rms value of (L - R) over a period of time Scaling operator 14L-1 scales the left lower frequency band signal by a factor √1 - a(left), Scaling operator 16R-1 scales the right lower frequency band signal by a factor a(right), =), (Formula Removed) which may be different than a(i ight)i Scaling operator 14R-1 scales the left lower frequency band signal by a factor√l-a(right), The left higher frequency band output is coupled directly to summer 18L so that the audio signal to speaker 20L consists of the left higher frequency band output from filter network 12L and the output from scaling operator 14L-1 The right higher frequency band output is coupled directly to summer 18R so that the audio signal to speaker 20R consists of the right higher frequency band output from filter network 12R and the output from scaling operator 14R-1 Scaling the portion of the L and R signals contributed to the centei channel by a factor a and scaling the portion of the L and R signals that remains in the L and R channels, respectively, by a factor √1-a results essentially in a conservation of energy routed to the center speaker and the left and right speakers If the scaling results in a very strong center speaker signal, the L and R signals will be correspondingly significantly less strong If the L and R signals (and not an L - R signal) are processed to provide the left sunound speaker and the right surround speaker signals, respectively, then the left surround speaker signal and the right surround speaker signal will be less stiong than the centei speaker signal This relationship results in a centei acoustic image that remains firmly anchored in the center and in the front If the scaling results in a weak center speaker signal, the L and R signals will be correspondingly significantly strongei If the L and R signals (and not an L - R signal) are processed to provide the left surround speaker and the right surround speaker signals, respectively, then the left surround speaker signal and the right surround speaker signal will be stronger than the center speaker signal This relationship results m a spacious acoustical image when there is no strong central acoustic image Referring now to FIG 6, there are shown plots of the behavior of the lower non-bass frequency band according to the exemplary steering circuitry 40 described in FIG 5B for various combinations of correlation and relative amplitudes The left side of each plot represents the steering behavior of the exemplary steering circuit for one or more spectral bands if the amplitude of the signal in the right channel (for example channel Rl of FIG 2) is significantly lower ( for example - 20dB) relative to the signal in the left channel (for example channel LI of FIG 2), or m other words if the amplitude of the signal in the left channel is significantly greater than the amplitude of the signal in the right channel (a condition hereinafter referred to as "left weighted") The right side of each plot represents the steering behavioi of the exemplary steering circuit foi one or more spectral bands if the amplitude of the signal in the right channel (for example channel Rl of FIG 2) is significantly greater (for example, + 20dB) relative to the signal in the left channel (for example channel LI of FIG 2), a condition hereinafter referred to as ' right weighted" The middle portion of each plot is the behavioi of the exemplary steering cncuit if the amplitudes of the left channel and the right channel are substantially equal The behavior of the steering circuitry is expressed in terms of the scaling factor applied to the various signals The behavior of the exemplary steering circuitry is shown for three conditions FIG 6A shows the effect of the steering circuitry when the signals in the left and right channels are correlated and in phase (typically indicated by a correlation coefficient c of 1) FIG 6B shows the effect of the steering circuitiy when the signals in the left and right channels are uncorrelated (typically indicated by a correlation coefficient c of 0 or if the signals in the left and right channels are in phase quadrature In othei examples of steering circuitry, the behavior in uncorrelated and phase quadrature conditions could be different FIG 6C shows the effect of the exemplary steering circuit if the signals in the left and right channels are correlated and out of phase (I e vary inversely with each other) The plots are intended to illustrate geneial behavior and are not intended to be used for providing precise data FIGS 6 and 7 show the behavior of the steering circuit for cardinal values of the correlation coefficient c For other values of c, the curves will differ from FIGS 6 and 7 It can be seen in FIG 6A if the signals in the left and right channels are correlated (c = 1), and if the signals are left weighted, the right speaker signal and the light surround speaker signal, are scaled by a factor near zeio The left speaker signal is scaled by a factor about 1 0 The left surround speaker is scaled by a factor of about 0 5 Similarly, if the amplitudes of the signals are right weighted, the left speaker signal and the left surround speaker signal are scaled by a factor near zero The right speaker signal is scaled value of about 1 0 The right surround speaker signal is scaled by a factor of about 0 5 For situations in which the amplitudes of the signals m the left and right channels are approximately equal, the center speaker signal is scaled by a factor of about 1 0 and the signals to the other speakers are scaled by a factor of near zero Looking at the curves corresponding to the individual speakers in FIG 6A, for left and right weighted conditions, the center speaker signal is scaled by a factor of approximately 0 3 As the amplitudes become less left or right weighted, the scaling factoi increases so that when the amplitudes of the signals in the left and light input channels are equal, the scaling factor of center speaker signal is about 1 0 For a left weighted condition, the scaling factor of the left speaker signal is about 0 9 As the amplitude becomes less left weighted, the scaling factor of the left speaker signal decreases, until it becomes approximately 0 when the amplitudes of the signals in the left and right channels are equal, and remains approximately zero for all values in which the signal m the right input channel is greater than the signal in the left input channel Foi a left weighted condition, the scaling factor of the left surround speaker signal is approximately 0 6 As the amplitudes becomes less left weighted, the scaling factoi of the left surround speakei signal decreases, until it becomes approximately zero when the amplitudes of the signals in the left and right channels are equal, and lemains appioximately zero foi all values in which the signal m the right input channel is gi eater than the signal in the left input channel The effect of the exemplary steering circuitry of FIG 6A on the right and right sunound channels is substantially a mirror image of the effect on the left and left surround channels It can be seen in FIG 6B (c = 0) that if the signals in the two channels are uncorrelated or in phase quadrature, for a left weighted condition, the left surround speaker signal has the highest scaling factor and the left surround speaker signal has the next highest weighted value The light, right surround and center speaker signals have a lelatively low scaling factor For a nght weighted condition, the signals show a substantially mirror image relationship For situations in which the amplitudes of the signals in the left and right channels are substantially equal, the scaling factors to all five speakers are in a relatively narrow band, with the left/right speaker signals having a slightly larger scaling factor than the center speaker signal, and the centei speaker signal having a slightly higher value that the left surround speaker signal and right surround speaker signal The plot of FIG 6C, in which the L and R signals are correlated (c = 1) and out of phase, shows the behavior of the steering circuitiy relative to the left, left surround, right, and right sunound speakers is similar to the behavior shown in FIG 6B However, in the curve of FIG 6C, the center speaker signal has a low scaling factor undei all conditions, and decreases to substantially zero if the signals m the input channels have the same amplitude FIG 7 discloses the behavior of another exemplary steering circuitry The behavior shown in FIG 7A (c = 1) is similar to the behavioi shown in FIG 6A for the left, right, and center speaker signals The scaling factor for the left surround and right surround speaker signals is substantially zero for all amplitude relationships of the input signals, indicating that the scaling factors are substantially independent of the amplitude lelationships of the input channels The behavior shown in FIG 6A and FIG 7A is substantially the same foi situations m which the amplitude of the signals in the two input channels is the same, which is consistent with an assumption that when signals are correlated, in phase, and of equal amplitude, the source of the sound is desired by the creator of the audio source matenal to be localized between the left and right speakeis A difference between the behavior shown in FIG 7B (c = 0) and the behavior shown in FIG 6B is that at certain amplitude relationships, in this example when the amplitudes of the signals in the two channels differ by less than 10 dB, in FIG 7B the scaling factors of the surround speaker signals are greater than the scaling factors of the left and right speaker signals Unlike the behavior of FIG 6B, the behavior shown m FIG 7B provides for a situation (uncorrelated, amplitudes relatively equal) in which the surround speaker scaling factors are larger than the left and right speaker scaling factors, therefore causing the audio image to move toward the rear A difference between the behavior shown in FIG 7C (c = 1, out of phase) and the behavioi shown in FIG 6C is that at most points on the plot, the scaling factor applied to the surround speaker signals (for example, the left surround speaker) is significantly greater than the scaling factor applied to the corresponding front speakei (for example the left speaker) This is consistent with audio encoding systems in which surround information is encoded as out of phase correlated audio signals Audio systems of the type shown m FIG 1A using steering circuitry 40 of the type disclosed in FIG 4 are advantageous over conventional audio systems that process stereo channel signals to provide x channel signals Conventional audio systems that process an L - R signal to provide surround channels from conventionally create stereo matenal may result in undesirable audible effects For example, a stereo recording of a sound source located equidistant from two stereo microphones may include direct ladiation from the source that is highly correlated, but reverberant radiation that is not highly correlated because of acoustical asymmetries in the environment in which the recoiding was made The uncorrelated reverberations may contribute to an L - R signal A conventional audio system that generates an L - R signal to use as a sunound signal may then cause the reverberations to be reproduced in a manner that sounds unnatural relative to the direct ladiation Audio systems of the type shown in FIG 1A using the steeimg circuitry 40 of the type disclosed in FIG 4 are also advantageous over audio systems that do not process signal in multiple frequency bands because they do not acoustic events in one frequency band to unnaturally affect acoustic events in other frequency bands Foi example, if an acoustic source in the vocal range is intended to be in the center, and instrumental acoustic sources outside the vocal range are intended to be on the sides, the vocal range acoustic source does not cause the instrumental range acoustic source to tend to appear to come from the center, and the instrumental range acoustic source does not cause the vocal range acoustic source to tend to appear to come from the sides Audio systems of the type shown in FIG IB using steering circuitry 40 of the type disclosed in FIG 4 are advantageous over conventional audio systems that decompress two channel compressed audio signal data because they do not fonn a difference signal of the de-compressed L and R signals Therefore systems using the circuitry 40 of FIG 4 unmask artifacts or misinterpret differences between de-compressed L and R channel signals to a much lesser extent than do conventional audio systems that generate and process the L - R signal to provide additional channels If the uncompressed audio signals are conventionally created stereo signals, audio systems of the type shown m FIG IB are also advantageous for the reasons stated in connection with the audio systems of the type shown in FIG 1A Those skilled in the art may now make numerous uses of and departures from the specific apparatus and techniques disclosed herein without departing from the inventive concepts Consequently, the invention is to be construed as embracing each and every novel feature and novel combination of features disclosed herein and limited only by the spirit and scope of the appended claims What is claimed is WE CLAIMS 1. A method for processing two input audio channel signals to provide n output audio channel signals where n>2, comprising: dividing the first input channel signal and the second input channel signal into a plurality of corresponding non-bass frequency bands; measuring the amplitude of the audio signal in the two input channels in one the frequency bands to provide a first channel first frequency band audio signal and a second channel first frequency band audio signal to provide a first channel first frequency band audio signal amplitude and a second channel first frequency band audio signal amplitude; determining the correlation between the first channel first frequency band audio signal and the second channel first frequency band audio signal to provide a first frequency band correlation; scaling the first channel first frequency band audio signal by a first factor (a(first)) related to the first frequency band con-elation and further related to the first channel first frequency band audio signal amplitude and the second channel first frequency band audio signal amplitude, the scaling to provide a first scaled first output channel first frequency band audio signal first portion; scaling the second channel first frequency band audio signal by a second factor (u(second)) related to the first frequency band correlation and further related to the first channel first frequency band audio signal amplitude and the second channel first frequency band audio signal amplitude, the scaling to provide a first scaled first output channel first frequency band audio signal second portion; combining the first scaled first channel first frequency band audio signal first portion and the first scaled first channel first frequency band audio signal second portion to provide a first frequency band portion of a center channel output audio signal. 2. A method for processing two input audio channel signals in accordance with claim 1, further comprising: scaling the first channel first frequency band audio signal by a third factor (a(third)) to provide a first frequency band portion of a left channel output audio signal. 3. A method for processing two input audio channel signals in accordance with claim 2, wherein a(third) = √ 1- a(first) 4. A method for processing two input audio channel signals in accordance with claim 2, further comprising: combining the first frequency band portion of the left channel output audio signal with a second frequency band portion of the first channel audio signal to provide a left non-bass audio signal. 5. A method for processing two input audio channel signals in accordance with claim 1, where the frequency bands are time varying. 6. A method for processing two input audio channel signals in accordance with claim 1, where the first frequency bands is the speech band. 7. A method for processing two input audio channel signals in accordance with claim 1, wherein the two input audio channel signals comprise compressed audio signal data. 8. A method for processing two input audio channels signals in accordance with claim 7, wherein the compressed audio signals are in a non-reconstructable data format. 9. A method for processing two input audio channel signals in accordance with claim 1, wherein the input signals are compressed according to the MP3 format. 10. A method for processing two input audio channel signals to provide n output audio channel signals wherein n > 3 and wherein the n output channel signals include surround channels, comprising: separating the two input channels into a plurality of corresponding non-bass frequency bands; processing each of the plurality of input channel non-bass frequency bands to provide the corresponding frequency band of a center channel output signal and two non-surround non-center output channel signals; processing at least one of the two non-center non-surround output channel signals to provide a surround output channel signal, wherein the processing the two non-center channel output signals does not include processing a signal representing the difference between the two input channels. 11. A method for processing two input audio channels to provide n output audio channels in accordance with claim 10, wherein the processing the two non-center channel output signals comprises at least one of time delaying, attenuating, and phase shifting one of the two non-center input channel signals. 12. A method for processing two input audio channels to provide n output audio channels where n>2, comprising: dividing the first input channel signal and the second input channel signal into a plurality of corresponding non-bass frequency bands; processing according to a first process a first input channel first frequency band audio signal to provide a first portion of a first frequency band of a center output channel signal; processing according to a second process a input channel first frequency band audio signal to provide a second portion of the first frequency band of the center output channel signal; processing according to a third process a first input channel second frequency band audio signal to provide a first portion of a second frequency band of the center output channel signal; and processing according to a fourth process a second input channel second frequency band audio signal to provide a second portion of the second frequency band of the center output channel signal; wherein the third process is different from the first process and the second process and wherein the fourth process is different from the first process and the second process. 13 A method for processing two input audio channels in accordance with claim 12, further comprising: processing according to a fifth process the first input channel first frequency band audio signal to provide a first portion of a first frequency band of a non-center output channel signal; and processing according to a sixth process the first input channel second frequency band audio signal to provide a first portion of a second frequency band of the non-center output channel signal; wherein the fifth process is different from the sixth process 14. A method for processing two input audio channels in accordance with claim 13, wherein the first process comprises scaling the first input channel first frequency band audio signal by a factor a. 15. A method for processing the two input audio channels in accordance with claim 14, wherein the fifth process comprises scaling the first input channel first frequency band audio signal by a factor √ l - a2 . 16. A method for processing two input audio channels in accordance with claim 15, wherein the sixth process comprises providing the unattenuated first input channel second frequency band audio signal so that the center output channel signal comprises the first input channel first frequency band audio signal scaled by a and so that the non-center output channel comprises the first input channel first frequency band signal scaled by √1 - a2 and the unattenuated first input channel second frequency band signal. 17 A method for processing the two input audio channels in accordance with claim 14, wherein the third process comprises providing none of the first input channel second frequency band audio signal to provide a first portion of a second frequency band of the center output channel signal so that the center output channel signal comprises the first input channel first frequency band audio signal scaled by a and no portion of the first input channel second frequency band audio signal. 18. A method for processing two input audio channels in accordance with claim 13, wherein the sixth process comprises providing the unattenuated first input channel first frequency band audio signal. 19. A method for processing two input audio channels in accordance with claim 12, wherein at least one of the first process, the second process, the third process, and the fourth process are time varying. 20. A method for processing two input audio channel signals to provide n output audio channel signals wherein n>2 and wherein the two input audio channel signals comprise unreconstructable compressed audio signal data, the method comprising: separating the input audio channel signals into frequency bands; separately processing the frequency bands; and combining the separately processed frequency bands to provide the n output audio channels. 21. A method for processing two input audio channel signals in accordance with claim 20, wherein the separately processing the frequency bands includes scaling a first channel first frequency band signal, scaling a second channels first frequency band signal, and wherein the separately processing does not include processing a signal representing the difference between any portions of the first input audio channel signal and the second audio channel signal.

Documents:

1168-del-2005-Abstract-(09-10-2012).pdf

1168-del-2005-abstract.pdf

1168-del-2005-assignment.pdf

1168-del-2005-Claims-(09-10-2012).pdf

1168-DEL-2005-Claims-121214.pdf

1168-del-2005-claims.pdf

1168-del-2005-Correspondence Others-(03-06-2011).pdf

1168-DEL-2005-Correspondence Others-(10-01-2012).pdf

1168-del-2005-Correspondence Others-(16-04-2012).pdf

1168-del-2005-Correspondence Others-(22-08-2012).pdf

1168-DEL-2005-Correspondence-121214.pdf

1168-del-2005-Correspondence-Others-(09-10-2012).pdf

1168-del-2005-correspondence-others.pdf

1168-del-2005-description (complete).pdf

1168-del-2005-Drawings-(09-10-2012).pdf

1168-DEL-2005-Form 1-121214.pdf

1168-DEL-2005-Form 5-121214.pdf

1168-del-2005-Form-1-(22-08-2012).pdf

1168-del-2005-form-1.pdf

1168-del-2005-Form-13-(09-10-2012).pdf

1168-del-2005-form-18.pdf

1168-del-2005-form-2.pdf

1168-DEL-2005-Form-3-(10-01-2012).pdf

1168-del-2005-form-3.pdf

1168-del-2005-form-5.pdf

1168-del-2005-GPA-(22-08-2012).pdf

1168-del-2005-gpa.pdf

1168-DEL-2005-OTHERS-121214.pdf

1168-DEL-2005-Petition-137-(10-01-2012).pdf