EP0209336A2 - Digitaler Schallsynthesierer und Verfahren - Google Patents

Digitaler Schallsynthesierer und Verfahren Download PDF

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Publication number
EP0209336A2
EP0209336A2 EP86305358A EP86305358A EP0209336A2 EP 0209336 A2 EP0209336 A2 EP 0209336A2 EP 86305358 A EP86305358 A EP 86305358A EP 86305358 A EP86305358 A EP 86305358A EP 0209336 A2 EP0209336 A2 EP 0209336A2
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EP
European Patent Office
Prior art keywords
coefficients
signal
providing
sets
transient
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Granted
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EP86305358A
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English (en)
French (fr)
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EP0209336A3 (en
EP0209336B1 (de
Inventor
Michael A. Deaett
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Raytheon Co
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Raytheon Co
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K15/00—Acoustics not otherwise provided for
    • G10K15/02—Synthesis of acoustic waves

Definitions

  • This invention relates to the synthesis of audio signals from stored digital data and more particularly to a synthesizer in which the stored data are sets of the coefficients of a recursive filter and where each set is applied to the filter for a fixed time period, the periods totaling the duration of the synthesized audio signal.
  • Acoustic trainers are typically required to produce signatures characteristic of signals received from sources in a real ocean environment.
  • the broadband and harmonic spectral content of targets and the broadband content of background noise have been emphasized for replication.
  • active echoes and reverberation have been added to the trainer repertoire.
  • An additional component of the acoustic environment which is required for purposeful training is the set of transient signatures. These include occasional and also continuous biologic emissions, hatch openings and closings, ice fractures in the arctic environment, undersea seismic disturbances, and the noise of submerged wrecks moving with currents - just to name a few.
  • the synthesis of these transients has typically resided in an instructor controlled analog tape recorder.
  • the method of synthesis utilizes linear prediction coding techniques to derive time-varying-filter coefficients. These coefficients are stored in digital form and are used to program a recursive filter which is driven by white noise. The resulting signatures are then an inherent part of the trainer and are generated under complete computer control.
  • a system which approximates the desired transient signature by the storage of sets of coefficients of a recursive digital filter, which coefficients are updated periodically thereby resulting in an output from the recursive filter which is a close approximation of the actual transient signature. It is assumed that an autoregressive model will provide an adequate description of the desired transient signature.
  • the signature which is desired to be synthesized is most easily obtained from a recording of the signature which is later to be synthesized. Because the spectral content of the transient signature is time varying, the auto-regressive model is nonstationary and must be updated periodically.
  • the transient signature is synthesized by considering the signal to be comprised of a serial sequence of blocks of the signal. Each block of the signal has its amplitude sampled to provide 1024 samples of digital data.
  • the autocorrelation function of the 1024 amplitude samples provides the 12 most significant autocorrelation values and a gain value which is obtained through the normalization of the autocorrelation values.
  • a matrix equation is obtained relating the autocorrelation values of the actual signal to the unknown coefficients of a recursive filter.
  • the system of equation of the matrix is solved for each block of data and the filter coefficients are stored.
  • the coefficients are periodically updated in real-time by a control processor.
  • the coefficients are recovered from memory in real-time and provided to the recursive filter circuitry whose output is provided to a digital to analog converter to produce audible sound replicating the original transient signature.
  • the transient analog signal 10 of figure 1 which is to be simulated by the apparatus of this invention is operated upon by first partitioning the analog signal into a sequence of frames 11.
  • the time duration of each frame is determined by examining the power spectral characteristics of the signal over a multiple of frame durations and then choosing the maximum duration over which those spectral characteristics are essentially constant. The greater the frequency exteht of the power distribution, the shorter is the time period for that frame.
  • the signal within each frame is then periodically sampled at a rate Tc exceeding the Nyquist rate and stored in digital form.
  • the set of samples 12 which is stored for each signal frame is a block of digital data. One block of data results for each signal frame.
  • the number of sample points per block is determined by the frame duration and by the highest frequency contained in the data signal waveform which is to be synthesized. At least two and preferably four data points are obtained within a signal frame for the highest frequency component within that frame.
  • the sample data points contained within each block are autocorrelated and a selected number of the autocorrelation coefficients are determined.
  • the number of autocorrelations values which are used is equal to the least number required to reduce the autocorrelation coefficient recursive prediction residual to an acceptable fraction of the zero lag autocorrelation value. An acceptable fraction is commonly 0.01.
  • the autocorrelation values are normalized by a factor R which provide unity value of the autocorrelation coefficient at zero displacement.
  • the autocorrelation function coefficients have been designated by the letters a o , a l , ... a m with the subscript indicating the relative lag displacement of the autocorrelated data blocks.
  • the unity value coefficient a o is the normalized autocorrelation coefficient at zero relative displacement.
  • the covariance matrix is next employed to determine the values for the multiplication factors "b" applied to the output of each of the delay units of the recursive filter as shown in FIG. 2.
  • the covariance matrix is given below where "a” with subscripts are the normalized autocorrelation coefficient values. "b” with subscripts are the covariance matrix coefficients obtained from the covariance matrix and are the multiplying factors which are applied to the multipliers of the recursive filter in addition to the gain factor A.
  • the transient and biologic signatures are generated by feeding "white” (uncorrelated) noise samples S(n) through a time-varying recursive digital filter.
  • the transfer function of this filter, G(Z) is given as
  • the analog synthesizer 50 comprises a pseudo-random noise generator 51 which produces an analog output signal having a value between zero and one which changes with every clock pulse input, the clock pulses having a period T c which is the same period as that at which the original signal 10 was sampled.
  • the clock pulses are provided by clock pulse generator 52 which also provides the clock pulses to the counter 53 having a modulo F where F is the number of samples of the analog signal in one block time.
  • a pulse having period T f is applied to the memory 54 to produce a new set of analog numbers A 1 , -b 1 , ..., -b m .
  • the memory 54 is represented as a multi-pole switch having n + 1 poles with the switch arms 55 moving by one switch position in response to each energization of the switch coil 56 by the pulse T f .
  • Each set of coefficients appearsat a selected position of switch arms 25. As shown in FIG. 2, the initial position of the switch arm provides the set of coefficients A, -b l , ..., -b m .
  • the second switch arm position which would exist as a result of one pulse T f would provide a different set of coefficients for the second block time; namely, A', -b l ', ..., -b m '.
  • the last set of coefficients corresponding to the last block of the sampled input signal is provided by the memory 54 as A k , -b l k , ..., . -b m k , where k is the number of blocks.
  • the output of the pseudo-random noise generator 51 is provided to a multiplier 57 whose other input during a block time is the amplitude coefficient A.
  • the output of multiplier 57 is provided at one input of the summing circuit 58.
  • the output of the summing circuit 58 is provided as the input to a delay unit 59 1 whose output is provided to delay unit 59 2 and to multiplier 57 1 .
  • the other input to multiplier 57 1 is the coefficient -b l provided by the memory 54 during the first time block.
  • the output of multiplier 57 1 is provided at another input to the summing circuit 58.
  • the time delay provided by delay 59 1 is equal to the interpulse period T c of the clock pulses provided by generator 52.
  • the circuit 50 has a cascade of delay elements 59 2 , ..., 59 m connected serially to the delay 59 1 .
  • the output of the summer circuit is the desired simulated signal which corresponds to the original signal which is being simulated. This simulated signal is designated as y(n).
  • the output of each delay unit 59 1 , 59 2 , ..., 59 m is correspondingly y(n - 1), y(n - 2 ), ..., y(n - m).
  • the circuit 50 will, therefore, provide an output y(n) in accordance with the equation presented earlier which sounds like the original audio signal which was sampled to provide coefficients b in the manner described earlier and stored in memory 54.
  • the coefficients which have been computed in the manner detailed in the preceding paragraphs are stored in sequential addresses of a RAM or ROM coefficient memory 31.
  • a RAM or ROM coefficient memory 31 In the example of the embodiment of this invention, it will be assumed that seven coefficients bi through b 7 of FIG. 2 together with the gain factor A are adequate for the synthesis and are stored in the first eight addresses 0, ..., 7 of memory 31. Addresses 8, ..., 15 will contain the coefficients A', -b l ', ..., -b 7 l .
  • Successive groups of eight addresses of RAM 31 have successive groups of coefficients A, -b l through -b 7 , (one group for each block of signal data) with the total number of groups of coefficients equaling the number of blocks of the original audio signal which is to be simulated by the synthesizer 30 of FIG. 4.
  • Counter 34 has input clock pulses having a period T c obtained from the modulo eight output line 44 of counter 33.
  • Counter 34 is of modulo L, where L is the number of samples per block of input signal.
  • the output pulse of counter 34 at the count of L increments by one the block counter 341.
  • the output count of counter 341 is provided to the more significant bits (MSB) of buffer register 35 which provide the block address to the memory 31.
  • the output count on line 33' of counter 33 is provided as the least significant bits (LSB) of register 35.
  • the output address of address generator 32 on line 321 will initially produce (through adder 65 and register 66) the sequential addresses 0, ..., 7 to the memory 31 repetitively for the number of samples L in the block, followed by the addresses 8, ..., 15, repeated L times, etc. Therefore, the memory 31 output will be a group of sequential coefficients A, -b l , ..., -b m at a period T c/ 8 (for addresses 1 through 8) repeated L times because of the modulo L of counter 34.
  • Block counter 341 which is incremented by one changes the MSB of register 35 so that the addresses 8, ..., 15 of memory 31 provide the next group of coefficients A', -b' 1 , ..., -b' m repeated L times also. This process of providing successive groups of coefficients to synthesize blocks of a signal continues until the memory 31 addresses contain no coefficients.
  • the pseudo-random noise generator 36 produces a 16-bit word for every 16-bit coefficient provided by memory 31.
  • the word produced by noise generator 36 is stored in a 16-bit register 37.
  • the memory 31 also produces the coefficients as 16-bit digital words and stores the words in register 38.
  • Registers 37 and 38 provide digital inputs to multiplier 39 which provides a 32-bit output word to adder 40.
  • Adder 40 provides an input to accumulator register 41 whose output is provided as a second input to adder 40 and whose output is provided also as an input to switch 42.
  • Switch 42 is open except when closed in response to a pulse on line 44 provided by the moculo m output of counter 33 to the random access memory 45.
  • the counter 33 of modulo 8 provides clock pulses T c on line 44 as an input to counter 47 which increments a write address to memory 45 at a time such that the switch 42 provides the output y(n) as an input to memory 45.
  • Switch 46 is also responsive to clock pulses at the period Tprovided by counter 33 on line 44. Closing of switch 46 by a pulse on line 44 allows the 16-bit number from random number generator 36 to be provided to the register 37 at that time as stated earlier. Since pulses in lines 44 only occur during the eighth count of counter 33, during the remaining seven other outputs of counter 33, switch 46 has an input 461 connected to the output of memory 45.
  • the write address provided by counter 47 at output 48 is provided as one input to subtract circuit 49.
  • the other input to subtractor 49 is the output count from counter 33 on line 33 1 .
  • the read address is provided at the time that the switch 46 input line 461 is providing a signal corresponding to that address from memory 45 to the register 37.
  • the circuit of F I G. 4 provides a newly calculated value of y(n) at intervals corresponding to the original sampling period T c .
  • the RAM 31 generates an amplitude coefficient A which is stored in register 38 and multiplied in multiplier 39 by the output x(n) of the random noise generator 38 provided to register 37 through closed switch 46.
  • the product A ⁇ x(n) is stored in accumulation register 41.
  • Switch 42 is open and no output appears to be read into memory 45 through its input register 60.
  • the switch 46 connects register 37 with the output of memory 45.
  • Subtractor circuit 49 has an input address 48 and an input address 33' which causes the next read address presented to memory 45 to be the address next preceding that at which the output y(n) has been written in by write address counter 47. This address will cause the value y(n - 1) to be read out to the register 37.
  • the address generator 32 is indexed to the second address of memory 31 and the value -b l will be read out and provided to register 38.
  • the resulting product provided by multiplier 39, -b l y(n - 1), is added in adder 40 to the previously stored value Ax(n) in register 41 and the sum ( A x(n) - b l y(n - 1)) is then stored in accumulation register 41.
  • the next timing pulse T c/ m causes the read address provided to memory 45 to be decremented by one and provide the output y(n - 2) to the register 37 through switch 46.
  • the address provided to memory 31 is incremented by one to provide the coefficient -b 2 to register 38.
  • the contents of the registers 37, 38 are multiplied in multiplier 39 to provide -b 2 y(n - 2) which is added in adder 40 to the exiting contents ( A x(n) - b l y(n - 1)) of accumulation register 41 and the result (Ax(n) - b l y(n - 1) - b 2 y(n - 2)) is then stored in register 41.
  • Register 60 contains the output y(n + 1) (which becomes the new value of y(n)) which is written into the next sequential address of memory 45 inasmuch as the write address counter 47 responsive to a pulse at the rate 1/T c on line 44 from counter 33 has caused the write address on line 48 to be incremented by one.
  • switch 42 When the new value of y(n) appears at the output of register 41, switch 42 is caused to close by a pulse on line 44 to thereby provide a new y(n) output and to provide this new value as the input to the memory 45 at the incremented address.
  • the output y(n) of switch 42 is in digital form and is converted to an analog signal Y(n) in digital-to-analog converter 62.
  • Signal Y(n) is smoothed in filter 63 to remove the sampling frequency components, centered at frequencies 1/T c and multiples thereof, and to thereby provide the synthesized analog signal y(t) that is desired corresponding to the blocks of coefficients selected by the initial address provided by start address register 64 which is added in adder 65 to the output of buffer register 35 and stored in register 66 before being provided to coefficient memory 31.
  • the computer 67 is programmed to provide one or a series of start addresses at predetermined time intervals to register 64 in response to a START command to thereby produce one or a series of timed, different synthesized audio output signals y(t), each corresponding to a different start address.

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  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • General Health & Medical Sciences (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Complex Calculations (AREA)
  • Electrophonic Musical Instruments (AREA)
  • Filters That Use Time-Delay Elements (AREA)
EP86305358A 1985-07-18 1986-07-11 Digitaler Schallsynthesierer und Verfahren Expired - Lifetime EP0209336B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US75622085A 1985-07-18 1985-07-18
US756220 1985-07-18

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EP0209336A2 true EP0209336A2 (de) 1987-01-21
EP0209336A3 EP0209336A3 (en) 1987-05-20
EP0209336B1 EP0209336B1 (de) 1991-03-13

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EP86305358A Expired - Lifetime EP0209336B1 (de) 1985-07-18 1986-07-11 Digitaler Schallsynthesierer und Verfahren

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EP (1) EP0209336B1 (de)
AU (1) AU588334B2 (de)
DE (1) DE3678054D1 (de)
ES (1) ES2000520A6 (de)
IL (1) IL79268A0 (de)

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5005204A (en) * 1985-07-18 1991-04-02 Raytheon Company Digital sound synthesizer and method
EP0436397A2 (de) * 1990-01-02 1991-07-10 Raytheon Company Schallsynthesizer
EP0535570A1 (de) * 1991-10-01 1993-04-07 Rockwell International Corporation Verarbeitung transienter Detektion, insbesondere Erkennung akustischer Signale unter Wasser
EP0575033A1 (de) * 1992-06-17 1993-12-22 Advanced Micro Devices, Inc. Architektur zur Erzeugung einer kovarianten Matrize

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4344148A (en) * 1977-06-17 1982-08-10 Texas Instruments Incorporated System using digital filter for waveform or speech synthesis
FR2510288A1 (fr) * 1981-07-24 1983-01-28 Labo Cent Telecommunicat Procede et dispositif de generation de bruits sous-marins, en particulier pour la simulation de bruits sonar
GB2103908A (en) * 1981-07-31 1983-02-23 Gen Electric Co Plc Linear predictive coder

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB1603993A (en) * 1977-06-17 1981-12-02 Texas Instruments Inc Lattice filter for waveform or speech synthesis circuits using digital logic

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4344148A (en) * 1977-06-17 1982-08-10 Texas Instruments Incorporated System using digital filter for waveform or speech synthesis
FR2510288A1 (fr) * 1981-07-24 1983-01-28 Labo Cent Telecommunicat Procede et dispositif de generation de bruits sous-marins, en particulier pour la simulation de bruits sonar
GB2103908A (en) * 1981-07-31 1983-02-23 Gen Electric Co Plc Linear predictive coder

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
Makhoul, John, Proc IEEE vol 63, no 4, April 75, pages 561-580; Schmid, Charles, Int Conf on Acoustic Speech, Signal Process IEEE Boston 1983; Chapter 7 and 10 of "Principles of Underwater Sound for Engineers" 2nd Edition MacGraw Hill, New York, 1975 *
NHK LABORATORIES NOTE, no. 209, February 1977, pages 2-12, NHK, Tokyo, JP; K. OZEKI: "A speech analysis and synthesis system by linear prediction for pitch controlled speech compilation experiments" *

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5005204A (en) * 1985-07-18 1991-04-02 Raytheon Company Digital sound synthesizer and method
EP0436397A2 (de) * 1990-01-02 1991-07-10 Raytheon Company Schallsynthesizer
EP0436397A3 (en) * 1990-01-02 1992-09-02 Raytheon Company Sound synthesizer
EP0535570A1 (de) * 1991-10-01 1993-04-07 Rockwell International Corporation Verarbeitung transienter Detektion, insbesondere Erkennung akustischer Signale unter Wasser
US5278774A (en) * 1991-10-01 1994-01-11 Rockwell International Corporation Alarm for transient underwater events
EP0575033A1 (de) * 1992-06-17 1993-12-22 Advanced Micro Devices, Inc. Architektur zur Erzeugung einer kovarianten Matrize

Also Published As

Publication number Publication date
IL79268A0 (en) 1986-09-30
EP0209336A3 (en) 1987-05-20
AU5923086A (en) 1987-01-22
ES2000520A6 (es) 1988-03-01
AU588334B2 (en) 1989-09-14
EP0209336B1 (de) 1991-03-13
DE3678054D1 (de) 1991-04-18

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