WO2007067320A2 - Decodeur audio matriciel de faible complexite - Google Patents

Decodeur audio matriciel de faible complexite Download PDF

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Publication number
WO2007067320A2
WO2007067320A2 PCT/US2006/044447 US2006044447W WO2007067320A2 WO 2007067320 A2 WO2007067320 A2 WO 2007067320A2 US 2006044447 W US2006044447 W US 2006044447W WO 2007067320 A2 WO2007067320 A2 WO 2007067320A2
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Prior art keywords
signals
directional
dominance
signal
audio
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PCT/US2006/044447
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English (en)
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WO2007067320A3 (fr
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Ching-Wei Chen
Christophe Chabanne
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Dolby Laboratories Licensing Corporation
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Priority to CN2006800519731A priority Critical patent/CN101336563B/zh
Priority to EP06837740A priority patent/EP1964443B1/fr
Publication of WO2007067320A2 publication Critical patent/WO2007067320A2/fr
Publication of WO2007067320A3 publication Critical patent/WO2007067320A3/fr
Priority to HK09101179.2A priority patent/HK1123663A1/xx

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other

Definitions

  • the invention relates to audio signal processing. More particularly, the invention relates to a low-complexity adaptive audio matrix decoder or decoding process usable for decoding both encoded and non-encoded input signals.
  • the decoder or decoding process may be advantageously used in combination with a "virtualizer” or “virtualization” process such that the decoder or decoding process provides multichannel inputs to the virtualizer or
  • the invention also relates to computer programs, stored on a computer-readable medium, for causing a computer to perform a decoding process or decoding and virtualization process according to aspects of the invention.
  • “Virtual headphone” and “virtual loudspeaker” audio processors typically encode multichannel audio signals, each associated with a direction, into two encoded channels so that, when the encoded channels are applied to a pair of transducers such as a pair of headphones or a pair of loudspeakers, a listener suitably located with respect to the transducers perceives the audio signals as coming from locations that may be different from the location of the transducers, desirably the directions associated with the directions of the multichannel audio signals.
  • Headphone virtualizers typically result in a listener perceiving that the sounds are "out- of-head” rather then inside the head.
  • Both virtual headphone and virtual loudspeaker processors involve the application of head-related-transfer- functions (HRTFs) to multichannel audio signals applied to them.
  • HRTFs head-related-transfer- functions
  • Virtual headphone and virtual loudspeaker processors are well known in the art and are similar to each other (a virtual loudspeaker processor may differ from a virtual headphone processor, for example, by including a "crosstalk canceller").
  • headphone and loudspeaker virtualizers examples include
  • Patents and an application relating to Dolby Headphone and Dolby Virtual Speaker include U.S. Patents 6,370,256; 6,574,649; and 6,741,706 and published International Application WO 99/14983.
  • Other "virtualizers” include, for example, those described in U.S. Patent 6,449,368 and published International Patent Application WO 2003/053099.
  • Dolby Headphone and Dolby Virtual Speaker provide, respectively, the impression of multichannel surround sound using a pair of standard headphones or a pair of standard loudspeakers.
  • Low-complexity versions of Dolby Headphone and Dolby Virtual Speaker were introduced that are useful, for example, in a wide variety of new, low-cost products, such as multimedia mobile phones, portable media players, portable game consoles, and low-cost television sets.
  • such low-cost products typically are two-channel stereophonic ("stereo") devices; whereas a virtualizer requires a multichannel surround sound input.
  • Dolby Pro Logic II and its predecessor Pro Logic are useful in matching the two-channel stereo audio output of low-cost devices to the multichannel surround sound input of a Dolby Headphone virtualizer
  • existing matrix decoders typically may be more complex and resource intensive than desirable for use with some low- cost devices.
  • Dolby Pro Logic and “Dolby Pro Logic II” are a trademarks of Dolby Laboratories Licensing Corporation. Aspects of Dolby Pro Logic II are set forth in U.S. Patents 6,920,223 and 6,970,567 and in published International Patent Application WO 2002/019768. Aspects of Dolby Pro Logic are set forth in U.S. Patents 4,799,260; 4,941,177; and 5,046,098.
  • such a new matrix decoder should minimize complexity in every stage of the process, while obtaining performance similar to a Dolby Pro Logic II decoder.
  • the invention relates to a method for processing audio signals by (1) deriving n audio output signals from m audio input signals, where m and n are positive whole integers and the n audio output signals are derived using an adaptive matrix or matrixing process responsive to one or more control signals, which matrix or matrixing process produces n audio signals in response to m audio signals, (2) deriving a plurality of time-varying control signals from the m audio input signals, wherein the control signals are derived from the m input audio signals using (a) a processor or process that produces a plurality of directional dominance signals in response to the m audio input signals, at least one directional dominance signal relating to a first directional axis and at least one other directional dominance signal relating to a second directional axis, and (b) a processor or process that produces the control signals in response to the directional dominance signals.
  • the adaptive matrix or matrixing process may include (1) a passive matrix or matrixing process that produces n audio signals in response to m audio signals, and (2) amplitude sealers or amplitude scaling processes, each of which amplitude scales one of the audio signals produced by the passive matrix or matrixing process in response to a time-varying amplitude-scale- factor control signal to produce the n audio output signals, wherein the plurality of time-varying control signals are n time-varying amplitude scale factor control signals, one for amplitude scaling each of the audio signals produced by the passive matrix or matrixing process.
  • the value m may be 2 and the value n may be 4 or 5.
  • the processor or process that produces directional dominance signals may use (1) a passive matrix or matrixing process that produces pairs of signals in response to the m audio input signals, a first pair of signals representing signal strength in opposing directions along a first directional axis and a second pair of signals representing signal strength in opposing directions along a second directional axis, and (2) a processor or process that produces, in response to the two pairs of signals, the plurality of directional dominance signals, at least one relating to each of the first and second directional axes.
  • the processor or process that produces a plurality of directional dominance signals may use linear amplitude domain subtractors or
  • subtraction processes that obtain a positive or negative difference between the magnitudes of each pair of signals, an amplifier or amplification process that amplifies each of the differences, a clipper or clipping process that limits each of the amplified differences substantially at a positive clipping level and a negative clipping level, and a smoother or smoothing process that time-averages each of the amplified and limited differences.
  • the processor or process that produces a plurality of directional dominance signals may use linear amplitude domain subtractors or
  • subtraction processes that obtain a positive or negative difference between the magnitudes of each pair of signals, a clipper or clipping process that limits each of the differences substantially at a positive clipping level and a negative clipping level, an amplifier or amplification process that amplifies each of the limited differences, and a smoother or smoothing process that time-averages each of the limited and amplified differences.
  • the relationship between the amplification factor of the amplifier or amplification process and the clipping level at which the clipper or clipping function limits the amplified difference may constitute a positive and negative threshold for magnitudes below which the limited and amplified difference may have an amplitude between zero and substantially the clipping level, and above which the limited and amplified difference may have an amplitude substantially at the clipping level.
  • the directional dominance signal may approximate a directional dominance signal based on a ratio of signal pairs comparison and for correlated audio input signals the directional dominance signal may tend toward the negative or positive clipping level.
  • the transfer function of the limited and amplified difference with respect to the difference may be substantially linear between the thresholds.
  • a difference above the positive threshold may indicate a positive dominance along a directional axis
  • a difference below the negative threshold may indicate a negative dominance along a directional axis
  • a difference between the positive and negative threshold may indicate non-dominance along a directional axis.
  • the processor or process that produces a plurality of directional dominance signals may also modify the amplified and limited difference signal prior to or after smoothing so that the derived directional dominance signal is biased along the axis to which the directional dominance signal relates.
  • the processor or process that produces a plurality of directional dominance signals may modify the amplified and limited difference signal differently when there is non-dominance along a directional axis than when there is positive or negative dominance.
  • the processor or process that produces the control signals in response to the plurality of directional dominance signals may apply at least one panning function to each of the plurality of directional dominance signals.
  • the invention may derive p audio signals from the n audio output signals, wherein p is two and the p audio signals are derived from the n audio signals using a virtualizer or virtualization process such that, when the p audio signals are applied to a pair of transducers, a listener suitably located with respect to the transducers perceives the n audio signals as coming from locations that may be different from the location of the transducers.
  • the virtualizer or virtualization process may include the application of one or more head-related-transfer-functions to ones of the n audio output signals.
  • the transducers may be a pair of headphones or a pair of loudspeakers.
  • aspects of the present invention are usable with other types of matrix decoders, in an exemplary embodiment a fixed-matrix-variable- gains approach is employed because of its low complexity compared to the variable-matrix approach.
  • the excellent isolation of single sound sources occurring with use of a variable gains decoder may be acceptable, if not preferable, for game audio where single audio events may be common.
  • optimizations may be made to deal with another known disadvantage of the variable-gains approach—the loss of non-dominant signals, resulting in a decoder with the best of both worlds.
  • the number of outputs may be restricted to four: Left, Right, Left Surround, and Right Surround.
  • the main goal of virtualizers is to convey a good sense directionality all around the listener; this may be achieved using only four channels, omitting the center channel, the inclusion of which would have significantly increased the processing execution time, while marginally enhancing the perception of directionality.
  • HRTFs Head Related Transfer Functions
  • FIG. 1 is a schematic functional block diagram showing an example of a processor or process according to aspects of the present invention for deriving pairs of intermediate control signals from a plurality of audio input signals, the pairs of intermediate controls signals representing signal strength in opposing directions along a directional axis.
  • the pairs of intermediate controls signals representing signal strength in opposing directions along a directional axis.
  • FIG. 2 is a schematic functional block diagram showing an example of a processor or process according to aspects of the present invention for deriving a plurality of directional dominance signals, at least one such signal for every pair of intermediate control signals.
  • Stage 2 there are two pairs of intermediate control signals, L-R and F-B and two directional dominance signals, LR and FB.
  • FIG. 3 shows an example of a notional or theoretical directional dominance vector in a two-dimensional plane based on orthogonal LR and FB axes.
  • FIG. 4 is an idealized plot of signal amplitude versus time showing the absolute values L and R, respectively, of a two-channel stereo signal in which the Left input channel (Lin) before taking its absolute value is a 50 Hz sine wave with a peak amplitude of 0.4, and the Right input channel (Rin) before taking its absolute value is a sine wave with frequency of (50 * V2) Hz and peak amplitude of 1.0.
  • the frequencies of the sine waves are uncorrelated, while the level of the Left channel is 0.4 times the level of the Right channel.
  • FIG. 5 is an idealized plot of signal amplitude versus time showing both the result of subtracting L from R and the result of multiplying the difference and then clipping at -1.0 and +1.0 to provide a quasi-rectangular wave.
  • FIG. 6 is an idealized plot of signal amplitude versus time showing a smoothed LR intermediate control signal resulting from feeding the quasi- rectangular wave of FIG. 5 through a smoother filter, illustrating that, for substantially non-correlated signal inputs, the directional dominance signal approaches a value close to a value that would result from a ratio-based comparison of signal strengths along the directional axis to which the LR intermediate control signal relates.
  • FIG. 7 is a schematic functional block diagram showing an example of a modification of the processor or process according to aspects of the present invention shown in FIG. 2. In this example, which may also be designated "Stage 2," the amplified and clipped FB difference is limited to values less than zero in order to bias the FB dominance signal towards the back.
  • FIG. 8 is an idealized plot of gain versus angle in radians showing a common pan-law between Left (L) and Right (R) audio channels, a
  • FIG. 9a is an idealized plot of gain versus directional dominant signal level for panL and panR when the same Sine/Cosine pan-law of FIG. 8 is applied to the LR axis, panL and panR representing the gain contribution, respectively, from Left and Right.
  • FIG. 9b is an idealized plot of gain versus directional dominant signal level for panB and panF when the same Sine/Cosine pan-law of FIG. 8 is applied to the FB axis, panB and panF representing the gain contribution, respectively, from Back and Front.
  • FIG. 10 is an idealized plot showing a quasi-3 -dimensional
  • FIG. 11 is an idealized plot showing a quasi-3 -dimensional
  • the lower curve is the approximation.
  • FIG. 13 is an idealized plot showing a quasi-3 -dimensional
  • FIG. 14 is a schematic functional block diagram showing an example of a processor or process according to aspects of the present invention for deriving a plurality of control signals from the plurality of directional dominance signals.
  • this example which may be designated "Stage 3”
  • four control signals LGain, RGain, LsGain and RsGain are derived from two directional dominance signals LR and FB.
  • FIG. 15 is a schematic functional block diagram showing an example of an adaptive matrix processor or process according to aspects of the present invention for deriving a plurality of audio output signals from the input audio signals and a plurality of control signals.
  • a pair of audio input signals Lin and Rin are applied to a passive matrix and the level of each matrix output is controlled by a respective one of the four control signals LGain, RGain, LsGain and RsGain to produce four audio output signals LOut, ROut, LsOut and RsOut.
  • FIG. 16 is a schematic functional block diagram showing an overview of all four Stages of the example, indicating their inter-relationships.
  • FIG. 16 The overall relationship of the four stages in the context of an adaptive matrix audio decoder or decoding process receiving m input audio signals, two signals, Lin and Rin in this example, and outputting n audio signals, four signals, LOut (left out), ROut (right out), LsOut (left-surround out), and RsOut (right-surround out), in this example, is shown in FIG. 16.
  • the decoder or decoding process has a control path that includes Stages 1, 2 and 3 and a signal path that includes an adaptive matrix or matrixing process in Stage 4.
  • a plurality of time-varying control signals, four control signals in this example, are generated by the control path and are applied to the adaptive matrix or matrixing process.
  • m audio input signals, Lin and Rin in this example are applied to a processor or process that derives pairs of signals in response to the m audio input signals, a first pair of signals, L and R in this example, representing signal strength in opposing directions along a first directional axis, an L-R or Left-Right axis in this example, and a second pair of signals, F and B in this example, representing signal strength in opposing directions along a second directional axis, an F-B or Front-Back axis in this example.
  • the processor or process of Stage 1 may be viewed as a passive matrix or matrixing process.
  • a simple passive matrix computes Left, Right, Sum and Difference signals, and their absolute values are used as intermediate control signals L, R, F, and B. More specifically, the passive matrix or passive matrixing process of this example may be characterized by the following equations:
  • Stage 2 the plurality of pairs of signals, each pair representing a signal strength in opposing directions along a directional axis, are applied to a processor or process that produces a plurality of directional dominance signals.
  • a processor or process that produces a plurality of directional dominance signals.
  • dominance signals , LR and FB are produced by Stage 2.
  • Stage 2 Before turning to details of a Stage 2 example, it is useful to explain the operational rationale of Stage 2.
  • a negative value along the LR axis may indicate dominance towards the Left, while a positive LR value may indicate dominance towards the Right.
  • a negative FB value may indicate dominance towards the Back, while a positive FB value may indicate dominance towards the Front.
  • the dominance in the LR direction is computed using the ratio of L and R
  • the dominance in the FB direction is computed using the ratio of F and B. Because a ratio is independent of the magnitude of the two signals being compared, it provides a steady dominant direction throughout the natural amplitude variations found in real audio signals.
  • DSP digital signal processor
  • aspects of the present invention retain much of the amplitude independence of the ratio-based comparison, but require much less computation.
  • the processor or process of Stage 2 produces a plurality of directional dominance signals using linear-amplitude-domain subtractors or subtraction processes that obtain a positive or negative difference between the
  • each subtraction is amplified by an amplifier or amplification process and the amplified difference is applied to a clipper or clipping process that limits each of the amplified differences substantially at a positive clipping level and a negative clipping level.
  • the order of the amplified differences is amplified by an amplifier or amplification process and the amplified difference is applied to a clipper or clipping process that limits each of the amplified differences substantially at a positive clipping level and a negative clipping level.
  • amplifier/amplification process and the clipper/clipping process may be reversed, using appropriate clipping levels in order to produce an equivalent result.
  • a smoother or smoothing process may time average each of the amplified and limited differences to provide a directional dominance signal.
  • the relationship between the amplification factor of the amplifier or amplification process and the clipping level at which the clipper or clipping function limits the amplified difference constitutes a positive and negative threshold for magnitudes below which the limited and amplified difference has an amplitude between zero and substantially the clipping level, and above which the limited and amplified difference has an amplitude substantially at the clipping level.
  • the particular transfer function is not critical and may take many forms, a transfer function in which the limited and amplified difference with respect to the difference is
  • substantially linear between the thresholds has very low computational requirements and is suitable.
  • the processor or process of Stage 2 may include modifications to an amplified and limited difference signal prior to or after smoothing during its processing so that the derived directional dominance signal is "biased" along the axis to which the directional dominance signal relates.
  • the bias may be fixed or adaptive.
  • a difference signal after amplification and clipping may be scaled in amplitude and/or shifted in amplitude ⁇ i.e., offset) and/or restricted in amplitude or sign in a fixed manner or, for example, as a function of the magnitude, sign, or magnitude and sign of the amplified and clipped difference signal.
  • the result for example, may include the application of less bias to non-dominant signals than to dominant signals (dominance and non-dominance are explained further below).
  • An example of applying "bias" to a directional dominance is described below in
  • two pair of signals, L-R and F-B 5 are applied in order to produce two directional dominance signals LR and FB.
  • the four intermediate directionality signals (L, R, F, B) as described above, one would like to derive two dominance signal components, LR and FB, by comparing the directionality along each axis. According to aspects of this invention, this is accomplished by subtracting R from L, and B from F (or vice-versa in each case), to provide a magnitude difference signal along each axis.
  • Heavy gain is applied to the difference signals, and the amplified difference is clipped (hard limited) to -1.0 and +1.0. The clipped difference signal is then applied to a time-smoothing filter.
  • any amount of dominance in a direction is treated as an absolute dominance in that direction.
  • the result of this operation is similar to a rectangular wave with varying frequency and duty-cycle.
  • the time- smoothing filter averages out the mostly rectangular wave to provide a continuous curve that approximates a ratio of the original directionality signals to one another.
  • the filter may be implemented efficiently, for example, as a first order digital IIR lowpass filter having a time constant of about 40 ms.
  • a general approach to doing this is to choose a threshold value, and assign differences with a magnitude greater than the threshold a value of -1.0 or 1.0 (depending on the sign of the difference), and assign differences with magnitudes smaller than this threshold some value in between the two extremes.
  • One possibility is to assign a value of 0.0 to all difference values below the threshold. To implement this in a program-controlled DSP would require some case statements and numerical comparisons. A better approach from the
  • both the gain and clipping stages may be implemented in a program-controlled DSP as an arithmetic left shift (for gains that are a power of 2) with the DSP's ''saturation logic" set (i.e., set a control register/bit in the DSP so that when the ALU overflows, the result is set to the maximum positive value or minimum negative value represented by the platform, depending on the sign).
  • Gains that are not a power of two may be
  • a three-regioned dominance signal (negative dominance, positive dominance, and non-dominance) permits distinguishing between dominance and non-dominance along a directional axis before smoothing.
  • Distinguishing dominance and non-dominance facilitates the adaptive application of "bias" to a directional dominance signal, as mentioned above and an example of which is given below in connection with FIG. 7.
  • musical material encoded with a Dolby Pro Logic II matrix encoder was decoded.
  • the average (F-B) difference signal was measured for a Left Surround- or Right Surround-steered input and this was used as an estimate of the maximum threshold (minimum gain) that would maintain a clear distinction between Left and Left Surround (or Right and Right Surround).
  • a gain factor of 1024 was used, equivalent to a threshold of approximately 0.001 for signals normalized to [-1 +1]. Thresholds smaller than 0.001 produce marginal audible improvement, while larger thresholds reduce the separation between the sides (Left and Right) and the surrounds (Left Surround and Right Surround) to unacceptable levels. In general, the threshold level is not critical.
  • the L and R intermediate signals are the magnitudes of the input signals Lin and Rin.
  • FIG. 5 shows the difference signal before and after clipping. Feeding the quasi-rectangular wave through a smoother filter that provides the LR directional dominance signal.
  • the directional dominance signal eventually reaches and oscillates around a value of 0.65, as shown in FIG. 6, close to the dominance value computed using a ratio-based comparison.
  • the smoothness of the oscillation is a function of the order and characteristics of the smoother filter.
  • This example is representative of audio material that has significant amounts of uncorrelated signals in each input, such as un-encoded two- channel stereo music, where the polarity of the clipped amplified difference signal is inverted very often. Under these input conditions, the
  • the clipped difference signal does not contain many zero crossings.
  • even the smoothed control signal tends to "lock” to one of the two extremes (i.e., +1.0 and -1.0), with a smoothed transition across to the other extreme if and when the polarity of the difference signal eventually inverts.
  • Such "locking" of one dominance component may be thought of as pulling a 2-dimensional dominance vector out along the edges of the LR/FB plane.
  • both components are "locked", the dominance vector is pulled to one of the four comers of the LR/FB plane.
  • such hard-panning improves the spatial imaging of matrix- encoded content, by providing a more discrete, single channel of input to a virtualizer.
  • variable gain approach A shortcoming of the variable gain approach is that non-dominant signals may be lost in the decoded output. This is apparent in musical sound sources, where there are a large number of sound sources mixed together with many different level and phase differences. Often, there are a few main instruments and vocals mixed equally in both Left and Right, while there are still many other less dominant, out-of-phase sounds that add to the overall space and ambience of the soundf ⁇ eld. Because the decoder uses only the direction of the most dominant sound component, a traditional variable gains approach on such material may result in almost no output of the out-of-phase material from the rear decoder outputs (the Left Surround and Right
  • this problem is mitigated by biasing the FB dominance signal towards the back, assuring that out-of-phase material is not completely removed from the surround outputs.
  • One way to accomplish this is to limit the FB signal to negative values before the smoother filter. This is shown in the example of FIG. 7. For a pure rectangular wave between -1.0 and 1.0, this is equivalent to scaling down by half the output of the smoother filter followed by a fixed offset of -0.5. Thus, such a modification may be imposed either before or after the smoother filter.
  • the clipped difference signal may not be a pure rectangular wave. Rather, it may contain in-between values when the difference signal falls below the threshold value, indicating non-dominance along a particular axis.
  • the processor or process of Stage 3 produces control signals for controlling the adaptive matrix or matrixing process in response to the plurality of directional dominance signals by applying one or more panning functions (a panning function is a transfer function representing an
  • One or more of the panning functions may implement one or more of:
  • a trigonometric transfer function (such as a sine or cosine transfer function)
  • the goal of Stage 3, in the example, is to take the LR and FB
  • the general approach for the matrix decoder or decoding process according to aspects of the present invention is this: having detected a certain dominant directionality in the input, emphasize the output channels closest to that dominant location, and de-emphasize the outputs furthest from the dominant location. Between the two outputs closest to the dominant location, the problem may be reduced to a pair- wise pan, which may be expressed as a panning function.
  • the gain for each decoder output channel must be expressed as a function of LR and FB:
  • LGain f L (LR, FB)
  • panL, panR, panB, and panF represent the gain contribution from respectively Left, Right, Back and Front.
  • panL cos ( (LR + I) / 2 * ⁇ /2)
  • panB cos ( (FB + 1) * ⁇ /2 )
  • LGain should be maximum only when both panL and panF are maximum, and should decrease as the dominance gets farther away on both, or either of the axis. This may be achieved by multiplying panL with panF.
  • the same principle may be applied to RGain, LsGain and RsGain, and the final equations for all gains become:
  • the use of a multiplication may also be seen as a mutual scaling of the two Sine/Cosine amplitude-panning functions, where the smallest value of the two components becomes the largest value that the overall gain can reach.
  • FIG. 10 shows the 3 -dimensional representation of the LGain equation
  • FIG. 11 the 3 -dimensional representation of all four gains superimposed.
  • the pan-law is composed of two curves: cos(x) and sin(x).
  • the sin function can be replaced by a cos function with the appropriate phase shift.
  • a second-order polynomial approximation of a cosine curve between 0 and ⁇ /2 may be used instead.
  • the anticipated audio input source is two-channel stereo, which is already mixed to pan naturally between L and R, it is an aspect of the present invention not to consider the LR panning component when calculating LGain and RGain.
  • the additional left-right panning in the variable gains would not significantly improve separation in this case, since L and R are already well separated.
  • it also allows a more stable soundfield in the front, by avoiding unnecessary gain riding. Removing the LR component, one arrives at these equations:
  • LGain l - FB 2
  • control signals LGain, RGain, LsGain, and RsGain are derived from the application of a panning function to a
  • the panning functions are panning functions that are not inherent in the n input audio signals.
  • one of the directional axes is a left/right axis and the panning functions are panning functions that do not include a left/right panning component.
  • the LR directional dominance signal is applied to a panL panning function and to a panR panning function.
  • the FB directional dominance signal (either without biasing as in FIG. 2 or with biasing as in FIG.
  • panF panning function is applied as both the LGain and as the RGain to the Stage 4 passive decoder or decoding process.
  • the result of applying the panB function to the FB dominance signal is multiplied by the result of applying the panL function to the LR dominance signal and is applied as the LsGain to the Stage 4 passive decoder or decoding process.
  • the result of applying the panR function to the LR dominance signals is multiplied by the result of applying the panB function to the FB dominance signal and is applied as the RsGain to the Stage 4 passive decoder or decoding process.
  • FIG. 15 shows a passive matrix or matrixing process that produces n audio signals in response to m audio signals, and amplitude sealers or amplitude scaling processes, each of which amplitude scales one of the audio signals produced by the passive matrix or matrixing process in response to a time-varying amplitude-scale-factor control signal to produce the n audio output signals, wherein the plurality of time-varying control signals are n time-varying amplitude scale factor control signals, one for amplitude scaling each of the audio signals produced by the passive matrix or matrixing process.
  • the plurality of time-varying control signals are n time-varying amplitude scale factor control signals, one for amplitude scaling each of the audio signals produced by the passive matrix or matrixing process.
  • the are two input audio signals, Lin and Rin, four audio output signals LOut, ROut, LsOut and RsOut, and four scale- factor control signals LGain, RGain, LsGain, and RsGain (from Stage 3).
  • four audio output signals may be provided.
  • a through h are matrix coefficients, as indicated in FIG. 15.
  • the coefficients a through h may be chosen to match those used in the Dolby Pro Logic II encode/decode system, where:
  • FIG. 16 shows an overview of all four Stages of the example, indicating their inter-relationships.
  • the invention may be implemented in hardware or software, or a combination of both ⁇ e.g., programmable logic arrays). Unless otherwise specified, the algorithms included as part of the invention are not inherently related to any particular computer or other apparatus. In particular, various general-purpose machines, such as digital signal processors, may be used with programs written in accordance with the teachings herein, or it may be more convenient to construct more specialized apparatus (e.g., integrated circuits) to perform the required method steps. Thus, the invention may be implemented in one or more computer programs executing on one or more programmable computer systems each comprising at least one processor, at least one data storage system (including volatile and non-volatile memory and/or storage elements), at least one input device or port, and at least one output device or port. Program code is applied to input data to perform the functions described herein and generate output information. The output information is applied to one or more output devices, in known fashion.
  • Program code is applied to input data to perform the functions described herein and generate output information.
  • the output information is applied to one or
  • Each such program may be implemented in any desired computer language (including machine, assembly, or high level procedural, logical, or object oriented programming languages) to communicate with a computer system.
  • the language may be a compiled or interpreted language.
  • Each such computer program is preferably stored on or downloaded to a storage media or device (e.g., solid state memory or media, or magnetic or optical media) readable by a general or special purpose programmable computer, for configuring and operating the computer when the storage media or device is read by the computer system to perform the procedures described herein.
  • the inventive system may also be considered to be implemented as a computer-readable storage medium, configured with a computer program, where the storage medium so configured causes a computer system to operate in a specific and predefined manner to perform the functions described herein.
  • a practical embodiment of the present invention embodied in a computer program suitable for controlling a digital signal processor has been implemented with under 30 lines of C code, running at an estimated 3 MIPS 5 and using virtually no memory.

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Abstract

Cette invention concerne un dispositif servant à extraire n signaux de sortie audio de m signaux d’entrée audio, où m et n sont des entiers positifs, les n signaux de sortie audio étant extraits à l’aide d’une matrice adaptative ou d’un procédé matriciel sensible à un ou plusieurs signaux de commande, cette matrice ou ce procédé produisant n signaux audio en réponse à m signaux audio, et (b) à extraire une pluralité de signaux de commande dynamiques des m signaux d’entrée audio, lesdits signaux de commande étant extraits à l’aide (i) d’un processeur ou d’un procédé qui produit une pluralité de signaux de dominance directionnelle en réponse aux m signaux d’entrée audio, au moins un signal de dominance directionnelle étant relatif à un premier axe directionnel et au moins un autre signal de dominance directionnelle étant relatif à un second axe directionnel, et (iÊ) d’un processeur ou d’un procédé qui produit les signaux de commande en réponse aux signaux de dominance directionnelle.
PCT/US2006/044447 2005-12-02 2006-11-16 Decodeur audio matriciel de faible complexite WO2007067320A2 (fr)

Priority Applications (3)

Application Number Priority Date Filing Date Title
CN2006800519731A CN101336563B (zh) 2005-12-02 2006-11-16 低复杂度的音频矩阵解码器
EP06837740A EP1964443B1 (fr) 2005-12-02 2006-11-16 Decodeur audio matriciel de faible complexite
HK09101179.2A HK1123663A1 (en) 2005-12-02 2009-02-10 Low-complexity audio matrix decoder

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US74156705P 2005-12-02 2005-12-02
US60/741,567 2005-12-02

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WO2007067320A2 true WO2007067320A2 (fr) 2007-06-14
WO2007067320A3 WO2007067320A3 (fr) 2007-11-01

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US9626975B2 (en) 2011-06-24 2017-04-18 Koninklijke Philips N.V. Audio signal processor for processing encoded multi-channel audio signals and method therefor

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CN102802112B (zh) * 2011-05-24 2014-08-13 鸿富锦精密工业（深圳）有限公司 具有音频文件格式转换功能的电子装置
CN106604199B (zh) * 2016-12-23 2018-09-18 湖南国科微电子股份有限公司 一种数字音频信号的矩阵处理方法及装置
US11095264B2 (en) * 2017-12-20 2021-08-17 Dolby Laboratories Licensing Corporation Configurable modal amplifier system
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US9626975B2 (en) 2011-06-24 2017-04-18 Koninklijke Philips N.V. Audio signal processor for processing encoded multi-channel audio signals and method therefor

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CN101336563B (zh) 2012-02-15
HK1123663A1 (en) 2009-06-19
TWI420918B (zh) 2013-12-21
EP1964443A2 (fr) 2008-09-03
EP1964443B1 (fr) 2011-09-21
TW200746872A (en) 2007-12-16
WO2007067320A3 (fr) 2007-11-01
CN101336563A (zh) 2008-12-31

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