EP0989543B1 - Schalleffekt Addiergerät - Google Patents

Schalleffekt Addiergerät Download PDF

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Publication number
EP0989543B1
EP0989543B1 EP19990307560 EP99307560A EP0989543B1 EP 0989543 B1 EP0989543 B1 EP 0989543B1 EP 19990307560 EP19990307560 EP 19990307560 EP 99307560 A EP99307560 A EP 99307560A EP 0989543 B1 EP0989543 B1 EP 0989543B1
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EP
European Patent Office
Prior art keywords
impulse response
reverberation
data
sound
input
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EP19990307560
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English (en)
French (fr)
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EP0989543A2 (de
EP0989543A3 (de
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Shigetaka I.P. Dept. Sony Corporation Nagatani
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Sony Corp
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Sony Corp
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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
    • G10K15/08—Arrangements for producing a reverberation or echo sound

Definitions

  • the present invention relates to a sound effect adding apparatus such as may, for example, be used with a reverberation adding apparatus that adds a reverberation to an original audio signal.
  • a reverberator As an apparatus that adds a sound effect to an audio signal, a reverberator is known.
  • the reverberator is used to add reverberation to an audio signal in for example a recording studio so that listeners can have a spatial impression and a deep impression.
  • a sound effect performed in a hall and a special effect can be added to the audio signal.
  • a recursive digital filter When reverberation is added to an audio signal by a digital process, a recursive digital filter is, for example, used. With the recursive digital filter, an input digital audio signal is attenuated and recurred. Thus, reverberation is generated. The generated reverberation is mixed with the original digital audio signal. In reality, initial reflection sound is added at a position delayed by a predetermined time period against direct sound. After a predetermined time period, reverberation is added. The delay time period of the reverberation against the direct sound is referred to as pre-delay. By adjusting the reverberation time, adding sub-reverberation, and finely adjusting the level of reverberation, a variety of types of sound can be generated.
  • a mechanical reverberator such as a steel-plate echo apparatus or a spring echo apparatus may be used.
  • a mechanical reverberator such as a steel-plate echo apparatus or a spring echo apparatus may be used.
  • problems of aged tolerance and necessity of maintenance These problems become critical for an apparatus that cannot be obtained due to out-of-fabrication.
  • such apparatuses are adversely affected by vibration and external noise.
  • the reverberation time cannot be freely adjusted.
  • such apparatuses do not have good reproducibility.
  • the weight and size of these apparatuses are large and S/N ratio of obtained sound is not high.
  • the reverberation time cannot be varied.
  • the reverberation contains a direct sound or an initially reflected sound, the reverberation is similar to a simulated sound of a hall.
  • the level of a direct sound is too high, when it is mixed with an input sound, the reverberation becomes an unnatural sound.
  • the delay after an audio signal is input until a direct sound is generated cannot be decreased to a predetermined value.
  • an initially reflected sound cannot be adjusted.
  • an object of the present invention is to provide a reverberation adding apparatus that alleviates the problems that take place in the case that collected impulse responses are used as they are.
  • United States Patent No. US-A-5544 249 discloses a sound effect adding apparatus generally according to the pre-characterising part of claim 1 hereof.
  • the non-prepublished European patent application EP 0 989 540 A1 describes an impulse response collecting method, sound effect adding apparatus, and recording medium.
  • the present invention provides a sound effect adding apparatus as set forth in claim 1 hereof.
  • a direct sound and an initially reflected sound can be removed from a reverberation.
  • the reverberation time can be adjusted.
  • the delay until an initially reflected sound is generated can be adjusted.
  • Fig. 1 shows an example of a structure for performing a convolution calculation for an impulse response in time axis direction using an FIR (Finite Impulse Response) filter. Coefficients of an impulse response are required corresponding to samples of an input digital audio signal.
  • FIR Finite Impulse Response
  • a digital audio signal is supplied from a terminal 310.
  • the number of quantizing bits of the digital audio signal is for example 24.
  • the sampling frequency of the digital audio signal is 48 kHz.
  • the input signal is supplied to 512 k delaying circuits 311 connected in series. Each of the 512k delaying circuits 311 has a delay of one sample.
  • Output signals of the individual delaying circuits 311 are supplied to respective coefficient multiplying devices 312.
  • Impulse response data of the first point to 512 k-th point is supplied to the delaying circuits 311 with 24 quantizing bits.
  • the coefficient multiplying devices 312 multiply respective output signals of the delaying circuits 311 by respective impulse response data.
  • the multiplied results are added by an adding device 313. The added result is supplied as reverberation data against the input data to a terminal 314.
  • Fig. 2 shows another example of the structure of the reverberation adding apparatus.
  • an input digital audio signal and impulse response data are converted into frequency element data corresponding to Fourier transform method and then a convolution calculation process is performed for the frequency element data.
  • an input digital audio signal is supplied from a terminal 320.
  • Data for samples corresponding to a required reverberation time (namely, data for 512 k points) is stored in a memory 321.
  • Data stored in the memory 321 is supplied to an FFT (Fast Fourier Transform) circuit 322.
  • the FFT circuit 322 performs fast Fourrier transform for the data received from the memory 321 and outputs frequency element data.
  • impulse response data is supplied from a terminal 323.
  • the impulse response data is stored in a memory 324.
  • the impulse response data is supplied to an FFT circuit 325.
  • the FFT circuit 325 performs fast Fourier transform for the impulse response data received from the memory 324 and outputs frequency element data. Since the impulse response data is known, the FFT 325 and the memory 324 may be composed of a ROM 326.
  • Output data of the FFT circuits 322 and 325 is supplied to a multiplying device 327.
  • the multiplying device 327 multiplies the output data of the FFT circuit 322 by the output data of the FFT circuit 325 in such a manner that the frequency components thereof match.
  • the multiplied result is supplied to an IFFT circuit 328.
  • the IFFT circuit 328 performs inversely fast Fourrier transform for the data received from the multiplying device 327 and outputs the resultant data as time axis data to a terminal 329.
  • the preferred embodiments of the present invention can be applied to a reverberation adding apparatus corresponding to any method.
  • the hardware scale for the convolution calculation process is small.
  • the delay time is small.
  • An impulse response is divided on time base.
  • a convolution calculation process for each divided impulse response and audio data is processed in parallel.
  • Figs. 3A and 3B show the relation between reverberation according to the invention and reverberation of a conventional recursive filter.
  • Fig. 3A shows reverberation of the conventional recursive filter.
  • the reverberation shown in Fig. 3A is generated in the following manner. Direct sound is delayed by a predetermined time period and thereby initial reflection sound is generated. The initial reflection sound is further delayed by a predetermined time period. Reverberation generated by the filter is added. The generated reverberation attenuates corresponding to a simple attenuation curve.
  • the reverberation since reverberation is generated with an impulse response corresponding to really recorded data, the reverberation corresponds to acoustic characteristics of a real hall or the like (namely, it does not correspond to a simple attenuation curve).
  • the embodiment of the present invention as mentioned in Fig. 3B, more natural and high quality reverberation can be obtained.
  • an impulse response collecting method for obtaining natural reverberation is accomplished.
  • Fig. 4 shows an example of the structure of an impulse response collecting apparatus 97 according to an embodiment of the present invention.
  • the impulse response collecting apparatus 97 measures an impulse response of a steel-plate echo apparatus 92.
  • the impulse response collecting apparatus 97 can be composed of for example a personal computer.
  • the apparatus 97 generates a signal for measuring an impulse response and outputs the signal to a measurement object.
  • the apparatus 97 collects measured results and converts them into impulse response data.
  • the impulse response data is stored as for example a file.
  • a measurement signal generating portion 90 generates a TSP (Time Stretch Pulse) signal for measuring an impulse response.
  • the TSP signal is a kind of a sweep signal.
  • an impulse signal is obtained.
  • To measure the impulse response it is preferred to directly generate the impulse signal.
  • generating impulse signal is convoluted by inverse function to TSP signal
  • the TSP signal generated by the measurement signal generating portion 90 is supplied to a D/A converter 91.
  • the D/A converter 91 converts the TSP signal as a digital signal into an analog signal.
  • the resultant analog signal is supplied to a steel-plate echo apparatus 92.
  • the steel-plate echo apparatus 92 generates reverberation with the input TSP signal.
  • the reverberation is output as analog audio signals on left (L) and right (R) channels.
  • the analog audio signals on L and R channels are supplied to an A/D converter 93.
  • the A/D converter 93 converts the analog audio signals on L and R channels into respective digital audio signals.
  • the A/D converter 93 samples the digital audio signals at a sampling frequency of 48 kHz or 96 kHz with 24 quantizing bits.
  • Output signals on L and R channels of the A/D converter 93 are supplied to an impulse response collecting apparatus 97.
  • the input signals of the impulse response collecting apparatus 97 are stored to a hard disk unit or a memory (not shown).
  • the measurement signal generating portion 90 generates the TSP signal N times.
  • a synchronously adding portion 94 synchronously adds N output signals of the measurement signal generating portion 90.
  • the synchronously adding process is performed in such a manner that the output signals of the synchronously adding portion 94 are synchronized corresponding to the generation timing of the TSP signal.
  • N signals By synchronously adding N signals, only reproducible signals are added.
  • the S/N ratio of the resultant signal can be improved.
  • the S/N ratio of the resultant signal is improved by (10 log N) dB.
  • the synchronously added signals on L and R channels are supplied to an impulse response converting portion 95.
  • a convolution calculation with the TSP siginal is performed to a supplied digital audio signal.
  • the TSP signal is converted into an impulse signal.
  • the measured result is converted into an impulse response corresponding to reverberation generated with the impulse signal.
  • the impulse response data has peak values obtained at intervals corresponding to the sampling frequency. After the A/D converter 93 samples a signal with 24 quantizing bits, the number of quantizing bits becomes 32.
  • Impulse response data 96L on L channel and impulse response data 96R on R channel that are supplied from the impulse response converting portion 95 are stored to a predetermined record medium such as a CD-ROM or an MD.
  • the impulse response collecting apparatus 97 may be provided with an interface such as Ethernet so as to supply the impulse response data to an external apparatus.
  • Fig. 5 shows an example of the case that an impulse response is collected in a hall.
  • a hall 101 has a stage portion 101A and a guest seat portion 101B.
  • a sound source 102 is disposed at a particular position of the stage portion 101A.
  • the sound source 102 is a dodecahedron speaker of which 12 speakers are disposed in 12 directions on a sphere.
  • a microphone 103L on L channel and a microphone 103R on R channel are disposed in the guest seat portion 101B.
  • a TSP signal is supplied from an impulse response collecting apparatus 97 to a D/A converter 91.
  • the D/A converter 91 converts the TSP signal as a digital signal into an analog signal.
  • the analog signal is supplied to an amplifier 100.
  • the amplifier 100 amplifies the analog signal.
  • the amplified signal is supplied to the sound source 102.
  • the sound source 102 reproduces the amplified signal as sound.
  • the reproduced sound is collected by the microphones 103L and 103R.
  • - Output signals of the microphones 103L and 103R are supplied to an A/D converter 93.
  • the A/D converter 93 samples the output signals of the microphones 103L and 103R at a predetermined sampling frequency and with a predetermined number of quantizing bits.
  • the resultant signals are supplied as digital audio signals on L and R channels to the impulse response collecting apparatus 97.
  • the process of the impulse response collecting apparatus 97 is the same as the process of the above-described steel-plate echo apparatus
  • the position of the sound source 102 is varied and impulse response data corresponding to varied positions is collected.
  • the brands of the speakers used as the sound source 102 are also changed and impulse response data corresponding to the individual brands of the speakers is collected.
  • the positions and brands of the microphones 103L and 103R are changed and impulse response data corresponding thereto is collected. In such a manner, a plurality of types of data in the hall 101 are collected. When reverberation is added, one of these types can be selected as variations of reverberation.
  • FIG. 6 shows a flow of an editing process of impulse response data.
  • impulse response data 110 is supplied to an editing process portion 111.
  • Figs. 7A, 7B, and 7C show examples of the editing process 111.
  • a system delay takes place in data due to propagation of sound (a system delay portion is denoted by "A" in Fig. 7A).
  • the editing process portion 111 sets the value of the system delay portion to "0" so as to remove noise therefrom.
  • a fade-out process is performed so as to converge the last end of the data at [0].
  • noise of a low level portion of the second half of the signal is removed.
  • Figs. 7B and 7C show examples of the fade-out process.
  • Fig. 7B shows an example of which the fade-out process is performed corresponding to an attenuation exponential function.
  • the original impulse response is denoted by h(n) and the fade-out function is denoted by F 0 (n) (where n represents a point of impulse response data).
  • F 0 (n) (where n represents a point of impulse response data).
  • a point of impulse response data corresponds to a sampling point of a digital audio signal.
  • F 0 (n) 1
  • F 0 (n) represents an attenuation exponential function as shown in Fig. 7B.
  • Output data x(n) is represented by the following expression (1).
  • x n x n • F o n - a
  • a is the number of samples corresponding to the position of direct sound in the original impulse response.
  • the fade-out function is not limited to an attenuation exponential function.
  • the fade-out function may be a function having a linear attenuation characteristic.
  • the number of points of the impulse response data can be adjusted corresponding to the process capability of the reverberator that adds reverberator to an audio signal with such data of the fade-out process.
  • a level adjusting process may be performed.
  • the edited impulse response data is recorded as an FIR filter coefficient 112 for a convolution calculation process of the FIR filter to for example a CD-ROM 45.
  • data that represents a feature of the impulse response data is stored as additional data to the CD-ROM 45.
  • the additional data contains for example data that represents a reverberation time of an impulse response, start/end point values of a direct sound and an initially reflected sound (for example, the value of an input point is "1"), and a start point value of a reverberation.
  • a plurality of types of impulse responses collected by a steel-plate echo apparatus, one type of impulse response of a hall, and additional data corresponding to each impulse response are recorded on the CD-ROM 45.
  • the reverberation time can be varied, a plurality of types of impulse responses corresponding to a plurality of reverberation times are collected.
  • Fig. 8 shows an example of the structure of a reverberator that performs a convolution calculation process with impulse response data generated in the above-described manner.
  • a digital audio signal is input from an input terminal 120.
  • the input signal is supplied to a multiplying device 126.
  • the input signal is supplied to a pre-delaying portion 121.
  • the pre-delaying portion 121 delays the input data.
  • Output data of the pre-delaying portion 121 is supplied to a convolution calculation process portion 122.
  • the convolution calculation process portion 122 is composed of FIR filters on L and R channels (namely, a filter 122L and a filter 122R).
  • the impulse response data 96L and 96R generated by the impulse response collecting apparatus 97 are supplied as FIR filter coefficients on L and R channels from terminals 123L and 123R, respectively.
  • the impulse response data 96L and 96R are read from for example a CD-ROM (not shown).
  • the filters 122L and 122R perform convolution calculation processes for the input digital audio signals with the impulse response data 96L and 96R, respectively. Thus, reverberation corresponding to the impulse response data 96L and 96R is generated. Output signals of the filters 122L and 122R are supplied to multiplying devices 124L and 124R, respectively.
  • the multiplying devices 124L and 124R, the above-described multiplying devices 126, and the adding devices 128L and 128R compose a mixer of the original sound (a dry component) and reverberation (a wet component).
  • the multiplying device 126 and the multiplying devices 124L and 124R adjust the input digital audio signal and the output signal of the convolution calculation process portion 122.
  • the adding devices 128L and 128R add these signals.
  • the output signal on L channel and the output signal on R channel are supplied to output terminals 129L and 129R, respectively.
  • Fig. 9 shows a detailed example of the structure of the reverberator.
  • digital audio signals of two channels are input from a digital audio input terminal 10 corresponding to AES/EBU (Audio Engineering Society/European Broadcasting Union) standard.
  • the digital audio signals received from the input terminal 10 are supplied to an input switcher 12 through a digital inputting portion 11.
  • the sampling frequency and the number of quantizing bits of the input digital audio signals are 48 kHz and 24 bits, respectively.
  • the sampling frequency of digital audio signals handled by the reverberator 1 can be doubled (namely, 96 kHz).
  • digital audio signals at a sampling frequency of 44.1 kHz can be handled by the reverberator 1.
  • signals at a sampling frequency of 88.2 kHz can be handled by the reverberator 1.
  • analog audio input terminals 13L and 13R are used. Audio signals on L and R channels are input from the input terminals 13L and 13R, respectively.
  • An AID converter 14 samples the audio signals at a sampling frequency of for example 48 kHz with 24 quantizing bits so as to convert these signals into respective digital audio signals. Output signals of the A/D converter 14 are supplied to an input switcher 12.
  • the input switcher 12 switches a source of input audio signals under the control of a controller 40 (that will be described later) or with a manual switch. Output signals of the input switcher 12 are supplied to a DSP (Digital Signal Processor) 30 through a path 31.
  • DSP Digital Signal Processor
  • the input switcher 12 switches a source of input audio signals under the control of a controller 40 (that will be described later) or with a manual switch. Output signals of the input switcher 12 are supplied to a DSP (Digital Signal Processor) 30 through a path 31.
  • DSP Digital Signal Processor
  • the DSP 30 has a DRAM (Dynamic Random Access Memory) and performs various control processes for input/output digital audio signals corresponding to a program received from the controller 40.
  • the DSP 30 supplies input digital audio signals to DSPs 32A to 32K that perform convolution calculation processes for obtaining impulse response data corresponding to a predetermined process.
  • the DSP 30 generates initial reflection sound corresponding to the input signals.
  • the DSP 30 receives the result of the convolution calculation process for the impulse response data from a DSP 34 (that will be described later).
  • the DSPs 32A to 32K divide input digital audio signals into blocks with predetermined sizes and perform convolution calculation processes for the divided blocks with the pre-supplied impulse response data.
  • the DSPs 32A to 32K have respective DRAMs with relevant capacities corresponding to the number of samples to be processed.
  • each of the DSPs 32A to 32H has one DRAM.
  • the DSP 32I has two DRAMs.
  • Each of the DSPs 32J and 32K has one DRAM with a capacity of 16 Mbits.
  • the results of the convolution calculation processes for the impulse response data for individual blocks performed by the DSPs 32A to 32K are added by an adding device 33.
  • the added result is supplied from the adding device 33 to the DSP 30 through a DSP 34.
  • the DSP 34 detects an overflow in the added result, the DSP 34 sets the data of the overflow to a predetermined value.
  • the DSP 30 combines the input digital audio signals, the initial reflection sound, and the result of the convolution calculation process for the impulse response data received from the DSP 34 so as to add reverberation to the input digital audio signals.
  • Output data 35 of the DSP 30 is supplied to an output switcher 18.
  • the generated reverberation and non-processed input digital audio signals are referred to as "wet component” an “dry component”, respectively.
  • the DSP 30 can vary the mixing ratio of the wet component and the dry component on each of L and R channels. In addition, the DSP 30 adjusts the levels of the output signals.
  • a clock signal FS or 2FS with a frequency corresponding to the sampling frequency of the handled digital audio signals is supplied to the DSP 30.
  • the DSP 30 processes signals corresponding to the clock signal FS or 2FS.
  • the output switcher 18 selects an output destination of output signals under the control of the controller 40 or with a manual switch.
  • the output signals are digital audio signals or analog audio signals.
  • the output switcher 18 supplies digital audio signals of two channels to output terminal 20 corresponding to the AES/EBU standard through a digital outputting portion 19.
  • the digital audio signals that are output from the output switcher 18 are supplied to a D/A converter 21.
  • the D/A converter 21 converts the digital audio signals received from the output switcher 18 into analog audio signals.
  • the analog audio signals on L and R channels are supplied to analog output terminals 22L and 22R, respectively.
  • the input terminal 10, the input terminals 13L and 13R, the output terminal 20, and the output terminals 22L and 22R are of cannon type having three signal lines of hot, cold, and ground lines.
  • the output switcher 18 allows the reverberation adding process in the reverberator 1 for the input audio signals to be bypassed.
  • the input digital audio signals are directly supplied to the output switcher 18 through the input switcher 12 and a bypass path 17.
  • the controller 40 comprises for example a CPU (Central Processing Unit), a RAM (Random Access Memory), a ROM (Read Only Memory), and predetermined input/output interfaces.
  • the ROM stores a boot program for starting up the system and serial number.
  • the RAM is a work memory with which the CPU operates. An external program is loaded to the RAM.
  • the controller 40 is connected to a bus 41 with for example eight bits parallel.
  • the bus 41 is connected to the DSP 30, 32A to 32H, and 34.
  • the controller 40 communicates with each of the DSPs 30, 32A to 32H, and 34 through the bus 41.
  • the controller 40 supplies programs to the DSPs 30, 32A to 32H, and 34.
  • the controller 40 exchanges data and commands with the DSPs 30, 32A to 32H, and 34.
  • the input switcher 12 and the output switcher 18 are connected to for example the bus 41 (not shown) and controlled by the controller 40.
  • a display unit 42 that is a full-dot LCD (Liquid Crystal Display) is connected to the controller 40.
  • the display unit 42 displays data generated by the controller 40.
  • the inputting portion 43 has a plurality of inputting means (for example, a rotary encoder for inputting data corresponding to the rotation angle and a plurality of push switches). By operating these inputting means, relevant control signals are supplied from the inputting portion 43 to the controller 40. Corresponding to the control signals, the controller 40 supplies predetermined programs and parameters to the DSPs 30, 32A to 32H, and 34.
  • a plurality of inputting means for example, a rotary encoder for inputting data corresponding to the rotation angle and a plurality of push switches.
  • the reverberator 1 has a CD-ROM (Compact Disc-ROM) drive 44.
  • a CD-ROM 45 is loaded to the CD-ROM drive 44. Data and programs are read from the CD-ROM 45. The data and programs that have been read from the CD-ROM 45 are supplied to the controller 40.
  • impulse response data has been recorded on the CD-ROM 45.
  • the impulse response data is read from the CD-RM 45 and supplied to the controller 40.
  • the data is supplied from the controller 40 to the DSPs 32A to 32K.
  • the DSPs 32A to 32K perform convolution calculation processes for impulse response data corresponding to the received impulse response data.
  • the reverberator 1 has an external interface MIDI (Musical Instrument Digital Interface).
  • An MIDI signal is supplied from an MIDI input terminal 46 to the controller 40.
  • the controller 40 controls a relevant function of the reverberator 1.
  • the controller 40 generates and outputs the MIDI signal.
  • the controller 40 can edit the MIDI signal received from the MIDI input terminal 46 and outputs the resultant signal.
  • the MIDI signal is supplied from the controller 40 to an external apparatus through the MIDI output terminal 47.
  • An MIDI through-terminal 48 is used to directly output the MIDI signal received from the MIDI input terminal 46.
  • extended functions can be obtained.
  • two more digital audio signals at a sampling frequency of 48 kHz can be handled.
  • the digital audio signals of two channels (channels 3 and 4) are received from a terminal 15 through the option board 50.
  • the digital audio signals are supplied to the input switcher 12 through the digital inputting portion 16.
  • digital audio signals of two channels corresponding to a process of the option board 50 are output to a terminal 24 through the digital outputting portion 23.
  • the digital audio signals are output to an external apparatus from the terminal 24 through the option board 50.
  • audio signals of two channels can be handled at a sampling frequency of 96 kHz that is twice as high as the normal sampling frequency.
  • the option board 50 and the reverberator 1 are connected with terminals 51 to 56, 15, and 24.
  • Fig. 10 shows an example of the structure of the option board 50.
  • the option board 50 performs an extended convolution calculation process for impulse response data using the DSPs 32A to 32K and the adding device 33.
  • the option board 50 has DSPs 32L, 32M, and 60A to 60L, an adding device 61, and a DSP 62.
  • the DSPs 32L, 32M, and 60A to 60L correspond to the DSPs 32A to 32K shown in Fig. 9, respectively.
  • the DSP 62 corresponds to the DSP 34 shown in Fig. 9.
  • a bus 41' of the option board 50 is connected to the bus 41 of the reverberator 1 through a terminal 56.
  • the DSPs 32L, 32M, and 60A to 60L of the option board 50 can communicate with the controller 40 through the bus 41'.
  • the DSPs 32L and 32M have eight 16-Mbit DRAMs each and perform convolutional calculation processes along with the DSPs 32A to 32K.
  • Input digital audio signals are supplied from the DSP 30 to the DSPs 32L and 32M through the terminal 53.
  • the results of the convolution calculation processes of the DSPs 32L and 32M are supplied to the adding device 33 through the terminals 54 and 55, respectively.
  • the adding device 33 adds the results of the convolution calculation processes of the DSPs 32L and 32M along with the results of the convolution calculation processes of the other DSPs 32A to 32K.
  • the DSPs 60A to 60M perform the convolution calculation processes in parallel with those of the DSPs 32A to 32M shown in Fig. 9.
  • Input digital audio signals are supplied from the DSP 30 to the DSPs 60A to 60M through the terminal 51.
  • the DSPs 32A to 32M When digital audio signals of four channels (channels 1 to 4) are processed with the option board 50, the DSPs 32A to 32M perform convolution calculation processes for digital audio signals of channels 1 and 2, whereas the DSPs 60A to 60M perform convolution calculation processes for digital audio signals of channels 3 and 4.
  • pairs of DSPs for example a pair of the DSPs 32A and 60A, a pair of the DSPs 32B and 60B, ..., and a pair of the DSPs 32M and 60M) that receive blocks with respective samples can perform respective convolution calculation processes in parallel at double speed.
  • the results of the convolution calculation processes of the DSPs 60A to 60M are supplied to the adding device 61.
  • the added result of the adding device 61 is supplied to the DSP 62.
  • the DSP 62 performs an overflow process.
  • the resultant signals are supplied to the DSP 30 through the terminal 52.
  • the DSP 30 adjusts the ratio of a dry component and a wet component and the mixing ratio of signals of individual channels and supplies the resultant data to the output switcher 18.
  • the option board 50 also has a digital audio signal input terminal 63 and a digital audio signal output terminal 64 corresponding to AES/EBU standard. Signals of two channels (channel 3 and 4) are input to the input terminal 63. The input signals are supplied to the input switcher 12 through the terminal 15. Likewise, output signals of two channels (channels 3 and 4) are supplied from the output switcher 18 to the option board 50 through the terminal 24 and then output from the output terminal 64.
  • the terminals 63 and 64 are of cannon type.
  • Fig. 11 shows an example of a front panel 200 of the reverberator 1.
  • Four mounting holes are formed at four corners of the front panel 200. With the four mounting holes, the reverberator 1 can be mounted to a rack.
  • a power switch 201 is disposed on the left of the panel 200. Below the power switch 201, a CD-ROM loading portion 202 is disposed.
  • a CD-ROM 45 is loaded to a CD-ROM drive 44 through the CD-ROM loading portion 202.
  • the CD-ROM 45 is loaded and unloaded to/from the CD-ROM drive 44 through the CD-ROM loading portion 202.
  • a display portion 203 is disposed at a nearly center position of the panel 200.
  • the display portion 203 corresponds to the LCD 42 shown in Fig. 9.
  • a rotary encoder 204 is disposed on the right of the display portion 203.
  • function keys 206, 207. 208, and 209 are disposed below the display portion 203. With the rotary encoder 204 and the function keys 206 to 209, the user can select one of the functions of the reverberator 1 and input data thereto.
  • the display portion 203 displays various data corresponding to the selected function.
  • the display portion 203 displays parameters corresponding to a selected reverberation type.
  • the display portion 203 is largely separated into a display area 210 and a display area 211.
  • the display area 210 visually displays parameters corresponding to the selected reverberation type.
  • the display area 211 displays parameter names and parameter values.
  • Data displayed in the display area 211 corresponds to the function switches 206 to 209 disposed below the display portion 203.
  • a parameter displayed above the function switch that has been pressed is selected.
  • the value of the selected parameter is varied.
  • Another page can be displayed on the display portion 203.
  • the value of another parameter can be varied.
  • the display area 210 displays ripples corresponding to a parameter that is being currently set.
  • the user can visually know the effect of reverberation (spatial impression) corresponding to the parameter value.
  • Figs. 12A to 12H and Figs. 13A to 13H show examples of ripples displayed in the display area 210. As the reverberation time is prolonged, the number of wavers of ripples is increased in the order from Figs. 12A to 12H to Figs. 13A to 13H.
  • the ripples are displayed in 16 levels of the minimum value to the maximum value of the reverberation time.
  • the ripples in 16 levels are proportional to the reverberation time.
  • Ripple display data is stored in the CD-ROM 45.
  • ripple display data is read from the CD-ROM 45 and stored in the RAM of the controller 40.
  • the ripple display data may be prestored in the ROM of the controller 40.
  • ripples are displayed in the display area 210, the user can visually know the effect of the reverberation that has been set. In other words, the user can visually know the spatial impression of the reverberation with ripples displayed in the display area 210.
  • ripples are displayed in the upper right direction of the display area 210.
  • ripples may be displayed with a different pattern.
  • Figs. 18A, 18B, and 18C show other examples of ripples displayed in the display area 210.
  • the center point and spreading direction of ripples can be freely set.
  • the center position of ripples can be set at the left end of the display area 210 (see Fig. 18A).
  • the center position of ripples can be set at the center of the display area 210 (see Fig. 18B).
  • a section of ripples may be displayed in the display area 210 (see Fig. 18C).
  • the shape of ripples can be varied corresponding to the selected reverberation type.
  • ripples are displayed as a still pattern.
  • ripples can be displayed as an animated pattern.
  • Fig. 15 shows a process performed by each of the DSPs 32A to 32K.
  • Impulse response data is read from for example the CD-ROM 45 under the control of the controller 40 and supplied to the DSPs 32A to 32K.
  • the impulse response data that is read from the CD-ROM 45 is stored to the DRAMs of the DSPs 32A to 32K.
  • Each of the DSPs 32A to 32K divides impulse response data at predetermined intervals on time axis corresponding to the process block sizes assigned thereto.
  • the data unit of impulse response data processed by the DSP 32 is denoted by N.
  • the data unit N is 128.
  • one word corresponds to data of one sample of a digital audio signal.
  • one word has a time period of (1/sampling frequency).
  • the number of quantizing bits of digital data is 24 bits.
  • Input data supplied to the DSP 32 is divided as block data of N words.
  • the time period for the first N words is the time period for inputting the data.
  • the input data of N words is stored to the DRAM of the DSP 32.
  • a convolution calculation process is performed for impulse response data corresponding to input data of N words stored in the DRAM.
  • the result of the process for N words is output.
  • output data is delayed by 2N words to input data.
  • Fig. 16 shows the process of the DSP 32 in detail.
  • the DSP 32 performs a convolution calculation process for impulse response data by known recursive convolution overlap save method.
  • an n-th block 80B and an (n-1)-th block 80A that immediately precedes the block 80B are supplied every N words on time axis.
  • the n-th block 80B and the (n-1)-th block 80A are converted into frequency element data 81 composed of a real part 81A of (N+1) words and an imaginary part 81B of (N-1) words by DFT (Discrete Fourier Transform) method.
  • DFT Discrete Fourier Transform
  • real data 82A and zero data 82B of impulse response data 82 have been converted into frequency element data 83 composed of a real part 83A of (N+1) words and an imaginary part 83B of (N-1) words by DFT method.
  • frequency element data 81 of the input data and the real part and imaginary part of the frequency element data 83 of the impulse response are multiplied, respectively.
  • the multiplied results of the same frequency components are added. Namely, filter process (convolution calculation process) is performed.
  • frequency element data 84 composed of a real part 84A of (N+1) words and an imaginary part of (N-1) words is obtained.
  • IDFT process that is an inverse process of the DFT process is performed for the frequency element data 84.
  • data 86 of 2N words on time axis is obtained.
  • a convolution calculation process is performed for more impulse response data.
  • a longer reverberation time can be obtained.
  • an output block is delayed by two blocks against an input block. Consequently, when the size of each block is increased, the delay time of an output component of the reverberation process adversely becomes long.
  • a process for obtaining a desired reverberation time is performed in parallel for a plurality of blocks each of which is composed of a predetermined number of points (words).
  • Figs. 17 and 18 show a convolution calculation process according to the embodiment of the present invention.
  • a digital audio signal is divided to a plurality of blocks.
  • a convolution calculation process for 2 18 words (256 k words) is considered.
  • a convolution calculation process is performed for a digital audio signal with impulse response data of 256 k words (256 k points).
  • the sampling frequency is 48 kHz
  • a reverberation time of around 5.3 seconds is obtained.
  • the sampling frequency is 44.1 kHz
  • a reverberation time of around 5.9 seconds is obtained.
  • the impulse response data of 256 k words is divided into two portions.
  • the temporally earlier portion of the two portions is further divided into two portions.
  • the earlier portion on time axis is successively divided into two portions.
  • the later portion on time axis is successively divided into two portions.
  • Fig. 18 is an enlarged view showing a top portion A of 8 k words shown in Fig. 17.
  • the portion A is divided into two portions.
  • the first 256-words portion is divided into two blocks each of which has 128 words.
  • a convolution calculation process is performed for impulse response data of the two blocks.
  • the reverberation component is delayed by 256 words of the first portion.
  • the sampling frequency is 48 kHz
  • the delay is as small as 5 msec. Thus, it does not adversely affect reverberation.
  • Each of the DSPs 32A to 32K performs a convolution calculation process for the relevant pair with the same block size.
  • the DSPs 32A to 32K divide their input data as follows.
  • the DSP 32A divides the input data into blocks each of which is composed of 128 words.
  • the DSP 32B divides the input data into blocks each of which is composed of 256 words.
  • the DSP 32C divides the input data into blocks each of which is composed of 512 words.
  • the DSP 32D divides the input data into blocks each of which is composed of 1 k words.
  • the DSP 32E divides the input data into blocks each of which is composed of 2 k words.
  • the DSP 32F divides the input data into blocks each of which is composed of 4 k words.
  • the DSP 32G divides the input data into blocks each of which is composed of 8 k words.
  • the DSP 32H divides the input data into blocks each of which is composed of 16 k words.
  • the DSP 32I divides the input data into blocks each of which is composed of 32 k words.
  • Each of the DSPs 32J and 32K divides the input data into blocks each of which is composed of 64 k words.
  • each DSP performs the process for a pair of blocks with the same block size on time division basis.
  • each of the DSPs 32A to 32K performs a convolution calculation process for divided block data with relevant impulse response data.
  • the second pair member of each pair is delayed by one block against the first pair member.
  • each of the DSPs 32A to 32K successively outputs two blocks with the same block size.
  • the adding device 33 adds the output blocks of the DSPs 32A to 32K and generates reverberation data 88.
  • Fig. 19 shows an example of the structure of a convolution calculation filter 70 used in each of the DSPs 32A to 32K.
  • the convolution calculation filter 70 performs a convolution calculation process.
  • the convolution calculation filter 70 is accomplished by a predetermined program supplied from the controller 40 to the DSPs 32A to 32K.
  • a digital audio signal is input from a terminal 71.
  • the input digital audio signal is supplied to a DFT circuit 72.
  • the DFT circuit 72 converts the digital audio signal on time axis into frequency element data.
  • Output data of the DFT circuit 72 is supplied to a multiplying device 74 and a delaying circuit 73.
  • the delaying circuit 73 delays the input digital audio signal by N words.
  • Data delayed by the delaying circuit 73 is supplied to a multiplying device 76.
  • the multiplying device 74 receives a filter coefficient A from a terminal 75.
  • the filter coefficient A is impulse response data that has been processed by DFT method.
  • the multiplying device 74 multiplies the output data of the DFT circuit 72 by the relevant frequency element of the filter coefficient A.
  • the multiplying device 76 performs the same process as the multiplying device 74.
  • the multiplying device 76 receives a filter coefficient B from a terminal 77.
  • the filter coefficient B is impulse response data that has been processed by DFT method.
  • the multiplying device 76 multiplies the output data of the delaying circuit 73 by the relevant frequency element of the filter coefficient B.
  • the multiplied results of the multiplying devices 74 and 76 are added by an adding device 78.
  • the added result is supplied to an IDFT circuit 79.
  • the IDFT circuit 79 converts the frequency element data into data on time axis and outputs the resultant data from a terminal 80.
  • the convolution calculation filter 70 performs a convolution calculation process for two blocks of data of which one block is delayed by N words (namely, one block) against the other block and outputs data of two blocks. As was described with reference to Fig. 16, the first pair member of each pair is discarded.
  • Fig. 20 shows a process on time axis performed by the convolution calculation filter 70 shown in Fig. 19.
  • the left end and right end of Fig. 20 show input data and output data, respectively. It is assumed that in Fig. 20, time passes downwards.
  • a plurality of filters 70 are shown. However, in reality, these processes are performed by one filter 70 at different timings. Thus, the result of the DFT process at the preceding timing is delayed by the delaying circuit 73. The delayed result is used for the filter process at the next timing. Consequently, output data delayed by two blocks against input data is successively obtained.
  • Fig. 21 is a functional block diagram showing an outline of parallel processes of the DSPs 32A to 32K.
  • Input data is supplied to the DSPs 32A to 32K in parallel.
  • Each of the calculated results of the DSPs 32A to 32K is delayed by 2 N words. The delayed results are supplied to an adding device 22.
  • data of a plurality of impulse responses and additional data corresponding thereto are read from the CD-ROM 45 to the reverberation adding apparatus.
  • data of a plurality of impulse responses and additional data corresponding thereto are read from the CD-ROM 45 to the reverberation adding apparatus.
  • a plurality of types of impulse responses collected by a steel-plate echo apparatus, one type of impulse response collected from a hall, and additional data corresponding to each impulse response are read from the CD-ROM 45 to the reverberation adding apparatus.
  • the additional data data that represents a reverberation time of each impulse response, start/end point values of a direct sound and an initially reflected sound (for example, the value of an input point is "1"), and a start point value of each reverberation are used.
  • Impulse responses are edited and combined by a software process of the controller 40.
  • a process for removing a direct sound and an initially reflected sound will be described.
  • a direct sound and an initially reflected sound are contained in a reverberation, a process as a simulator of a hall is performed.
  • a direct sound and an initially reflected sound should be removed from a reverberation.
  • Fig. 22 shows original impulse response data of a hall. The original impulse response data is read from the CD-ROM 44. Referring to Fig. 22, the sampling frequency of the impulse response data is 96 kHz and 64 k points of 512 k points are shown.
  • Fig. 23 is an enlarged view showing an initial response portion (4 k points) of the impulse response data shown in Fig. 22.
  • Figs. 24 and 25 show impulse response data of which impulse response components corresponding to a direct sound and an initially reflected sound are removed from original impulse response data, respectively.
  • impulse response components corresponding to a direct sound and an initially reflected sound are set to "0"
  • the direct sound and the initially reflected sound can be removed from the original impulse response data.
  • Fig. 24 shows an impulse response component for 64 k points against the impulse response data shown in Fig. 22.
  • Fig. 25 is an enlarged view showing an impulse response component for 4 k points against the impulse response component shown in Fig. 24.
  • upper impulse response components and lower impulse response components correspond to L channel and R channel, respectively.
  • additional data corresponding to the impulse response data shown in Fig. 22 contains information of start/end points of impulse response components corresponding to a direct sound and an initially reflected sound
  • the positions of the impulse response components corresponding to the direct sound and the initially reflected sound can be specified.
  • the initially reflected sound can be removed.
  • a direct sound and an initially reflected sound can be removed without need to use the additional information.
  • a portion after the maximum value of a waveform of an envelope of an impulse response until the slope of the envelop is inverted to plus is estimated as impulse response components corresponding to a direct sound or an initially reflected sound. Since a direct sound largely affects an reverberation, only the direct sound may be removed.
  • Fig. 26A shows original impulse response data.
  • the start point of a direct sound and an initially reflected sound (denoted by N1)
  • the end point thereof (denoted by N2)
  • the start point of a reverberation (denoted by N3)
  • a reverberation time (denoted by To)
  • original impulse response data is separated into impulse response components corresponding to a direct sound and an initially reflected sound (see Fig. 26B) and an impulse response component corresponding to a reverberation (see Fig. 26C).
  • the impulse response components corresponding to the direct sound and the initially reflected sound are very large against the impulse response component corresponding to the reverberation, the levels of the impulse response components corresponding to the direct sound and the initially reflected sound are lowered.
  • the resultant impulse response components are combined with the impulse response component corresponding to the reverberation.
  • Fig. 26D shows the combined impulse response data.
  • the level of the impulse response component corresponding to the reverberation can be raised.
  • the balance of impulse response components may be adjusted using a cross fade method.
  • impulse response components corresponding to a direct sound and an initially reflected sound are faded out, whereas an impulse response component corresponding to a reverberation is faded in.
  • the fade-in/out curves may be straight lines or exponential curves.
  • a third example of the impulse response editing process will be described.
  • the reverberation time is adjusted.
  • the frequency characteristic of the reverberation adding apparatus is also varied.
  • Fig. 27 shows an example of measured results of frequency characteristics of different reverberation times at 0.2 sec steps (1.5 sec, 1.75 sec, 2.0 sec, ... 3.5 sec).
  • impulse responses corresponding to representative reverberation times is provided.
  • Impulse responses corresponding to other reverberation times are obtained by calculations.
  • the range of a reverberation time is divided into a plurality of time zones. In each time zone, an impulse response corresponding to a desired reverberation time is obtained with the representative impulse responses.
  • impulse responses corresponding to reverberation times 2.25 sec, 2.5 sec, and 2.75 sec are obtained.
  • impulse responses corresponding to nine types of reverberation times can be obtained.
  • an original impulse response is multiplied by an exponential function so as to shorten a reverberation time.
  • Trev To • Tx / ( To + Tx )
  • the resultant exponential function is multiplied by representative impulse response data.
  • the reverberation time can be adjusted to Tx.
  • Fig. 29A shows an impulse response of a steel-plate echo apparatus.
  • Fig. 29B shows an impulse response of a hall.
  • the point N1 of the impulse response corresponding to the direct sound is the same as that corresponding to the directly reflected sound.
  • the point N1 of the impulse response corresponding to the direct sound may be different from that corresponding to the directly reflected sound.
  • the impulse response components corresponding to the direct sound and the initially reflected sound are removed from the impulse response data of the steel-plate echo apparatus shown in Fig. 29A.
  • the impulse response component corresponding to the reverberation is separated as shown in Fig. 29C.
  • the impulse response component corresponding to the reverberation of the impulse response of the hall shown in Fig. 29B is removed.
  • the impulse response components corresponding to the direct sound and the initially reflected sound are separated as shown in Fig. 29D.
  • a combined impulse response as shown in Fig. 29E is generated.
  • two impulse response components may be combined by a cross fade method using a straight line or an exponential function.
  • a fifth example of the impulse response editing process will be described.
  • initially reflected sounds different from an original impulse response are combined.
  • Fig. 30, DL1, DL2,..., and DLn are delay devices with different delay amounts and connected in parallel.
  • An initially reflected sound x0 with an original impulse response shown in Fig. 31A is supplied to the delay devices DL1 to DLn.
  • Output signals of the delay devices DL1 to DLn are supplied to multiplying devices MP1 to MPn, respectively.
  • the multiplying devices MP1 to MPn multiply the output signals of the delay devices DS1 to DLn by coefficients k1 to kn, respectively.
  • the multiplying devices MP1 to MPn generate impulse response components x1 to xn corresponding to the initially reflected sound whose level has been adjusted at the positions corresponding to the delay amounts of the delay devices DL1 to DLn, respectively.
  • the impulse response x0 of the original initially reflected sound and the generated impulse response components x1 to xn corresponding to the initially reflected sound are added by an adding device AD.
  • the adding device AD outputs an impulse response component corresponding to an initially reflected sound that is different from the original impulse response.
  • the process of the block diagram shown in Fig. 30 is executed by software of the controller 40.
  • the value of a pre-delay that is a time period after an original audio signal is input until a reverberation is generated.
  • the value of a pre-delay may be set to 0 or a negative value.
  • the above-described impulse response editing/combining process can be performed by a switch operation of the reverberation adding apparatus or a rotating operation of a rotary encoder.
  • additional data may be displayed on an LCD display portion.
  • Parameters of reverberation time of an impulse response that has been edited may be displayed.
  • an impulse response may be displayed.
  • the impulse response collecting apparatus 97 is independent from the reverberator 1.
  • the reverberator 1 may have a measurement signal generating portion 90, a synchronously adding portion 94, and an impulse response converting portion 95.
  • the measurement signal generating portion 90 generates a TSP signal.
  • These portions can be composed of a CPU and several peripheral parts.
  • the DSP 30 and DSP 34 of the reverberator 1 may be used.
  • the convolution calculation process for impulse response data is performed by hardware such as DSPs 32A to 32K.
  • the convolution calculation process may be performed by software.
  • the processes of the DSPs 30 and 34 may be performed by software.
  • a sound effect is obtained by performing a convolution calculation process for a collected impulse response and audio data, a natural, high quality result is obtained.
  • an original impulse response is edited or combined with another one, various sound effects can be obtained as the user desires.
  • by editing an impulse response or combining impulse response components the following effects can be obtained.
  • a desired reverberation time can be obtained by an adjustment.
  • a desired reverberation time can be obtained by a calculation.
  • the data mount of an impulse response can be decreased.
  • impulse response components corresponding to a direct sound and initially reflected sounds can be removed. Their levels can be adjusted.
  • a desired initially reflected sound can be composed.
  • a new impulse response can be generated.
  • a pre-delay can be adjusted to a desired value. Since an impulse response can be edited and impulse responses can be combined, the operational characteristics of a conventional digital reverberation adding apparatus can be accomplished. In addition, a natural, high quality reverberation can be added to an original audio signal.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Reverberation, Karaoke And Other Acoustics (AREA)
  • Stereophonic System (AREA)

Claims (8)

  1. Klangeffekt-Addiergerät zum Durchführen eines Faltungsberechnungsprozesses für ein zugeführtes digitales Audiosignal und von Impulsantwortdaten, wodurch ein Klangeffekt dem zugeführten digitalen Signal hinzugefügt wird, wobei das Gerät umfasst:
    eine Impulsantwort-Editiereinrichtung (111; 40) zum Editieren einer ursprünglichen Impulsantwort; und
    eine Faltungseinrichtung (122; 32) zum Durchführen eines Faltungsberechnungsprozesses für die editierte Impulsantwort und ein digitales Audiosignal;
    gekennzeichnet dadurch, dass die Impulsantwort-Editiereinrichtung (111; 40) in der Lage ist, die ursprüngliche Impulsantwort zu editieren, indem eine Information verwendet wird, die die Eigenschaften der ursprünglichen Impulsantwort darstellt, wobei die Eigenschaften eine Nachhallzeit (To) der ursprünglichen Impulsantwort, Start- und Endpunktwerte (N1, N2) eines direkten Klanges und eines anfänglich reflektierten Klanges und einen Startpunktwert (N3) eines Nachhalls umfassen.
  2. Gerät nach Anspruch 1, das eine Einrichtung (44) zum Wiedergeben einer Mehrzahl von verschiedenen Impulsantworten von einem Aufzeichnungsmedium (45) umfasst.
  3. Gerät nach Anspruch 1, wobei die ursprüngliche Impulsantwort mit einer ersten Impulsantwort und einer zweiten Impulsantwort zusammengesetzt ist, wobei die erste Impulsantwort einem direkten Klang und einem anfänglichen reflektierten unmittelbar nach dem digitalen Audiosignal zugeführten Klang entspricht, und wobei die zweite Impulsantwort einem Nachhall nach der ersten Impulsantwort entspricht.
  4. Gerät nach Anspruch 3, wobei die Impulsantwort-Editiereinrichtung ausgebildet ist, um zumindest eine Impulsantwort-Komponente, die dem direkten Klang entspricht, von der ersten Impulsantwort zu entfernen.
  5. Gerät nach Anspruch 3, wobei die Impulsantwort-Editiereinrichtung ausgebildet ist, um den Wert einer Impulsantwort-Komponente, die zumindest dem direkten Klang der ersten Impulsantwort entspricht, anzupassen und diese mit der zweiten Impulsantwort nach der Anpassung zu kombinieren.
  6. Gerät nach Anspruch 3, wobei die Impulsantwort-Editiereinrichtung ausgebildet ist, um eine Impulsantwort, die einer gewünschten Nachhallzeit entspricht, zu erzeugen, indem eine Mehrzahl von repräsentativen Impulsantwortkomponenten verwendet werden, die verschiedenen durch Berechnung erhaltenen Nachhallzeiten der zweiten Impulsantwort entsprechen.
  7. Gerät nach Anspruch 3, wobei die Impulsantwort-Editiereinrichtung ausgebildet ist, um eine Mehrzahl von ursprünglichen Impulsantworten so zu kombinieren, um eine neue Impulsantwort zu erzeugen.
  8. Gerät nach Anspruch 3, wobei die Impulsantwort-Editiereinrichtung ausgebildet ist, um eine Vorverzögerungszeit anzupassen, die von einem Zeitpunkt nach dem Zuführen des direkten Klanges bis zu einem Zeitpunkt, zu dem ein Nachhall erzeugt wird, dauert.
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SE0202159D0 (sv) * 2001-07-10 2002-07-09 Coding Technologies Sweden Ab Efficientand scalable parametric stereo coding for low bitrate applications
EP1423847B1 (de) 2001-11-29 2005-02-02 Coding Technologies AB Wiederherstellung von hochfrequenzkomponenten
SE0202770D0 (sv) 2002-09-18 2002-09-18 Coding Technologies Sweden Ab Method for reduction of aliasing introduces by spectral envelope adjustment in real-valued filterbanks
MXPA05008317A (es) * 2003-02-06 2005-11-04 Dolby Lab Licensing Corp Audio con respaldo continuo.
JP4127094B2 (ja) 2003-03-26 2008-07-30 ヤマハ株式会社 残響音生成装置およびプログラム
KR100769990B1 (ko) 2004-07-20 2007-10-25 재단법인서울대학교산학협력재단 입체음향의 공간감 및 거리감 제어를 위한 공간 임펄스응답 제어 장치
US8180067B2 (en) 2006-04-28 2012-05-15 Harman International Industries, Incorporated System for selectively extracting components of an audio input signal
US8036767B2 (en) 2006-09-20 2011-10-11 Harman International Industries, Incorporated System for extracting and changing the reverberant content of an audio input signal
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