EP0641143A2 - Procédé pour simuler un effet spatial et/ou sonore - Google Patents

Procédé pour simuler un effet spatial et/ou sonore Download PDF

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Publication number
EP0641143A2
EP0641143A2 EP94112549A EP94112549A EP0641143A2 EP 0641143 A2 EP0641143 A2 EP 0641143A2 EP 94112549 A EP94112549 A EP 94112549A EP 94112549 A EP94112549 A EP 94112549A EP 0641143 A2 EP0641143 A2 EP 0641143A2
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EP
European Patent Office
Prior art keywords
impulse response
room
room impulse
determined
threshold value
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
EP94112549A
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German (de)
English (en)
Other versions
EP0641143B1 (fr
EP0641143A3 (fr
Inventor
Martin Dipl.-Ing.Dr.Techn. Opitz
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
AKG Acoustics GmbH
Original Assignee
AKG Akustische und Kino Geraete GmbH
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Publication date
Application filed by AKG Akustische und Kino Geraete GmbH filed Critical AKG Akustische und Kino Geraete GmbH
Publication of EP0641143A2 publication Critical patent/EP0641143A2/fr
Publication of EP0641143A3 publication Critical patent/EP0641143A3/fr
Application granted granted Critical
Publication of EP0641143B1 publication Critical patent/EP0641143B1/fr
Anticipated expiration legal-status Critical
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/002—Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/002—Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution
    • H04S1/005—For headphones

Definitions

  • the invention relates to a method with the necessary electroacoustic device for generating a spatial and / or sound impression of an actually existing or also calculated space, with any monophonic, stereophonic or multi-channel audio program being usable as the hearing program.
  • the playback is preferably binaural via headphones, but can also be carried out via loudspeakers.
  • Each audio program produced generally contains the room acoustics that were present when the recording was made, but the fine structure of the previously known stereophonic reproduction methods could never be reproduced in a completely recognizable manner. More than that the recording was made in a room with a certain reverberation, the listener could not ascertain during playback. Only additional measures with corresponding electro-acoustic devices could create better listening conditions, which then also allow the listener to recognize the space in which the program was recorded.
  • a simulation of room acoustics that is true to the original can be carried out, for example, by convoluting any audio program with the binaural room impulse response measured at a specific reception location in a room.
  • a binaural spatial impulse response is understood to mean two impulse responses, one impulse response being assigned to one ear and the other impulse response to the other ear.
  • the room together with the reception characteristics of the human ear, forms a linear causal transmission system that is described in the time domain by the room impulse responses.
  • the respective room impulse response is approximately the system response to a sound impulse, the duration of which is a period of twice the upper limit frequency of the audio signal.
  • Such a simulation process which simulates the listener the unmistakably precise temporal, spectral, spatial and dynamic sound field structures that actually exist at the original listening location, is extremely complex, especially with regard to the technical equipment required for the simulation.
  • the convolution is carried out in such a way that the audio signal and the room impulse responses are digitized, the folded signal is calculated in a computer and converted back into the analog signal. The number of calculation steps depends on the length of the impulse responses.
  • An electroacoustic arrangement for the simulation of a listening situation present at a certain listening position, which is almost true to the original, is described for the reproduction of stereophonic binaural audio programs by means of headphones in AT PS 394 650.
  • Adherence to the auditory fidelity to the original and the correct localization of certain sound sources distributed in the room is therefore out of the question, since a sound recording available for stereophonic loudspeaker reproduction is then correctly presented for headphone reproduction that is almost true to the original if, in addition to the directly arriving audio signals of the two channels, left and on the right also the room reflections of the listening room, but evaluated with the direction-dependent outer ear transmission functions, are reproduced.
  • the simulation of room acoustic events can be carried out very generally using a method which is known, for example, from EP-A-0 505 949.
  • a transfer function is simulated using a transfer function simulator.
  • This transmission function simulator is equipped with sound sources, sound receiving devices and devices for measuring the acoustic transmission function arranged in an acoustic system. To measure the acoustic transmission function, the large number of possible different positions between any two points in the acoustic system can be taken into account.
  • the simulator itself is characterized in that means are provided for estimating the poles present in the existing transfer function, the AR eigen coefficients, which correspond to physical poles of the acoustic system, being estimated from the multiplicity of measured transfer functions, and ARMA filters, which are composed of AR filters and MA filters, simulate what corresponds to the acoustic system from the large number of measured acoustic transmission functions.
  • This extremely complicated process serves to simulate such an acoustic transmission function, which is required for echo blocking devices, anti-hall devices, for active noise compensation and also for sound image localization.
  • a signal processor simulates the transmission characteristics. In the simulation process itself, the transfer function is simulated with little computing effort and consequently the shortest possible computing time.
  • This simulation method just mentioned could also be used to implement the lifelike one Use reproduction of room acoustic events after a modification has been made. From a technical point of view, however, it would be extremely complex and too specific for there to be a particular interest in the sensible and economical application of this method for the entire purpose.
  • the object for the present invention is to create a simulation method with the required electroacoustic device, which is simplified, whereby its implementation is technically and economically justifiable.
  • the new simulation method has the advantage that there is no deterioration in simulation quality with a greatly reduced effort for the method.
  • Simplified FIR filter structures can also be used for the convolution.
  • the folding process itself takes place in real time without any noticeable time delay.
  • the essence of the invention lies in the fact that a lifelike simulation associated with success with very specific ones Parts of the room impulse responses from the acoustic events can be carried out. All that is required is knowledge of those parts of the room impulse responses that, after a critical selection, are essential for the auditory impression. The path to knowledge about the respective room impulse responses leads to real or virtual room acoustic measurements. The decision as to which parts of the spatial replies are left out is based on the principles of hearing psychology.
  • An essential embodiment of the method is that the values of the room impulse response are compared with a time-dependent threshold value and only those values of the room impulse responses are used that exceed the threshold value.
  • the threshold value is time-dependent in relation to the room impulse response, insofar as it has its greatest amount in the area of the beginning of the room impulse response and decays towards the end of the room impulse response. As a result, large areas of the room impulse responses become zero.
  • the advantage of such a division lies in the greatly reduced computing effort for the simulation processor.
  • the area of the room impulse response that records the direct sound must be composed with the area containing the reverberation in such a way that the original quality is retained in the simulation.
  • the above-mentioned method with the necessary electroacoustic device can also be designed in such a way that the critical selection of essential parts for obtaining the lifelike simulation takes place by taking into account the psychoacoustic pre- and post-masking phenomena in the room impulse response.
  • the occlusion phenomena known in hearing acoustics mean that if sound is present, a further, second sound can only be heard if its excitation in the human ear exceeds that of the first. This results in a shift in the audibility threshold, which is simulated by the above-mentioned time-dependent threshold value, as a result of which sound is not perceived below this threshold.
  • the application of the simulation method according to the invention will lie in particular in the hi-fi and recording studio area, because that is where the advantages of binaural licensing lie both for headphone reproduction and for loudspeaker reproduction.
  • the device according to the invention creates that level of good and true-to-original room acoustics that eliminates the known disadvantages of hearing in anechoic rooms, but does not interfere with the acoustics given by the recording.
  • the simulation of, for example, a specific loudspeaker arrangement in a specific room by means of headphone reproduction is an essential application of the simulation method including the electro-acoustic device required for this.
  • FIG. 1a A possible method for determining the room impulse response is shown in FIG. 1a.
  • a measurement signal is emitted at the location of the sound source, which is recorded at the listening position with a measurement microphone.
  • the room impulse response is obtained from the received signal. If a pulse is used as the measurement signal, the duration of which is equal to a period of twice the frequency of the upper frequency limit of the audio signal range, the received signal is equal to the room impulse response h (t). Since the signal-to-noise ratio is low with this method, a longer measurement signal is preferred in practice and the room impulse response is calculated.
  • the binaural impulse response that is used for playback Is needed via headphones is obtained in that the measuring microphones are located in the ear canals of a test person for whom the rum impulse response is to be determined.
  • the impulse response for the loudspeaker-room-ear path and then the impulse response for the headphone-ear system are then measured.
  • the impulse responses obtained are transformed into the frequency domain, the transformed functions are divided and the quotient is transformed back into the time domain. If this process is carried out for both ears, a binaural room impulse response is obtained, which is composed of a right and a left room impulse response.
  • FIG. 1b shows the scheme for the process sequence for one of the two room impulse responses as determined above.
  • the room impulse response h (t) is fed to the divider 1 in order to divide it into the direct sound component d (t) and the reverberation component r (t).
  • the reverberation component r (t) also contains all of the individual reflections of the measurement signal originating from the room walls.
  • the corresponding time-dependent amplitude profiles are shown schematically in FIGS. 4a to 4c.
  • T N ⁇ the direct sound has arrived at the listening position, according to which only those portions are to be expected which result from reflections or from the reverberation.
  • the spatial impulse response outlined here is also determined in the area of direct sound by the transfer function from the sound source to the ear canal entrance and is extended to a few milliseconds, for example due to the reflections on the head and body.
  • the determined room impulse response divided into the two sound components d (n) and r (n) is now fed to that electronic device 2, which extracts from the determined room impulse response the components that assign the characteristic values of the listening room acoustics, the sound field present in the listening room and those that can be assigned to the listener contain left and right outer ear transfer function, which after the folding process with any audio program guarantee the lifelike simulation of the entire room acoustics.
  • the extraction takes place according to criteria which are described further below.
  • the extracted or reduced room impulse response h '(n) is folded in a processor 3 with the signal s (n) of an arbitrarily selected audio program, whereby the signal is formed. With correct sound reproduction on the two ears of the hearing person, the hearing result desired according to the invention is achieved, namely the lifelike simulation of a listening position in a specific listening room.
  • the extractor circuit 2 for selecting the essential components from the determined room impulse response is explained in more detail by the diagram in FIG. 2.
  • the room impulse response present at an input E and divided into the direct sound and reverberation components is divided into individual sections or portions with the length T i in a function block 4.
  • Figures 5a to 5e show how the determined room impulse response by means of the function block 4 is divided into individual blocks or portions T i with the sound components d (n), r2 (n), r3 (n) ... r i (n).
  • the division into direct sound and reverberation components is carried out because the direct component of the determined room impulse response should remain unchanged at least in studio use and only the reverberation component is reduced as described. However, applications are also conceivable in which both parts of the determined room impulse response are reduced.
  • the remaining portions of the room impulse response which are below a defined threshold value according to one of the criteria described below are set to zero by means of a comparator 5.
  • the number of samples in the remaining signal components of the reduced spatial impulse response are counted in a coefficient counter 6.
  • the counter value obtained is compared in a setpoint comparator 7 with a limit value which is determined by the permissible computational effort. If the limit has not yet been exceeded, further blocks of the determined room impulse response are requested according to FIGS. 5a-5e. In this way, the computing capacity is fully used in a later convolution with the reduced room impulse response. If the specified target is reached, the reduced room impulse response that is now present is sent to an output A.
  • the arrangement shown in FIG. 3 is required for this.
  • a dynamic threshold value adaptation which consists of a comparator 9 and a threshold value generator 10.
  • the comparator 9 the instantaneous value of the determined room impulse response is compared with the instantaneous threshold value, the size of the threshold value depending on the previous values of the determined room impulse response according to the concealment phenomenon.
  • the dynamic adaptation to the predetermined psychoacoustic criteria is implemented in accordance with the concealment phenomenon, for example according to Zwicker.
  • the critical selection of the signal components of the determined room impulse response that are essential for the simulation can be made in that all components of the determined room impulse response that are below a fixed fixed threshold value A are set to zero so that they are suitable for later use Convolution process are not taken into account, while the signal components exceeding the threshold value or the associated sample values are adopted with unchanged amplitude in the reduced spatial impulse response. Since there is a direct connection between the strength of the sound reflections and the values of the determined room impulse response that can be assigned to these reflections, the threshold value criterion offers significant help for extracting the values of the determined room impulse response that are essential for simulation.
  • FIGS. 7a and 7b show, the critical selection is also possible according to criteria according to the occlusion phenomena. Accordingly, portions from the determined room impulse response that are not perceptible when listening need not be taken into account. According to the information available, the hidden parts are to be excluded from the folding that occurs later. In this case, it is no longer necessary to differentiate between direct sound and reverberation, but instead the total determined room impulse response can be reduced from the beginning as described.
  • T V here denotes the areas of pre-masking and T N that of subsequent masking. These are the periods in which signals below a level limit, as outlined in FIG. 7a, are no longer perceptible to a main signal. As can be seen from the standard literature on this subject, these masking effects are dependent on the time interval, the level ratio and the frequency interval of the masked and masking signal. As a result, this cannot be fully illustrated. With the room impulse response, above all the time and level relationships are influenced. In any case, somewhat broader value ranges of the determined room impulse response must be used than would result directly from the boundary line criterion. Furthermore, the value ranges have to be extrapolated into the actually masked range in order not to obtain undesired filter effects in the frequency range.
  • 8a and 8b show how the threshold value is scaled down in steps and the signal components for the simulation are taken accordingly.
  • FIG. 9 shows the manner in which, for example, the architecture of a conventional FIR filter can be implemented.
  • a signal value is extracted from each connection in each sampling period and multiplied by the filter coefficient assigned to this location; the result is added in an adder with all other results and fed to the output and thus represents the direct implementation of the convolution on a processor.
  • this convolution can of course also be carried out in other conjugate structures, as a result of which Saving computing power. In principle, however, this always involves an optimal chronological sequence of additions and multiplications, so that at best a factor of two to three can be gained in computing power.
  • Fig. 10 illustrates how the architecture of the FIR filter is modified when the convolution is performed with the extracted spatial impulse response.
  • the successive samples of the remaining signal components of the spatial impulse response form the filter coefficients d j , r 1k , r 2l , r 3m , r in . These are those which are of essential importance for the lifelike simulation in accordance with the designations from the example of FIG. 5.
  • the number of all filter coefficients is one to two orders of magnitude less than the number of buffers. Since the filter coefficients no longer occur equidistantly in time, the filter processor is simultaneously informed of the delay time or the sample number with a filter coefficient.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Reverberation, Karaoke And Other Acoustics (AREA)
  • Stereophonic System (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Diaphragms For Electromechanical Transducers (AREA)
  • Stringed Musical Instruments (AREA)
  • Vehicle Interior And Exterior Ornaments, Soundproofing, And Insulation (AREA)
EP94112549A 1993-08-26 1994-08-11 Procédé pour simuler un effet spatial et/ou sonore Expired - Lifetime EP0641143B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE4328620 1993-08-26
DE4328620A DE4328620C1 (de) 1993-08-26 1993-08-26 Verfahren zur Simulation eines Raum- und/oder Klangeindrucks

Publications (3)

Publication Number Publication Date
EP0641143A2 true EP0641143A2 (fr) 1995-03-01
EP0641143A3 EP0641143A3 (fr) 1999-05-19
EP0641143B1 EP0641143B1 (fr) 2001-12-05

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EP94112549A Expired - Lifetime EP0641143B1 (fr) 1993-08-26 1994-08-11 Procédé pour simuler un effet spatial et/ou sonore

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US (1) US5544249A (fr)
EP (1) EP0641143B1 (fr)
JP (1) JP3565908B2 (fr)
AT (1) ATE210362T1 (fr)
DE (2) DE4328620C1 (fr)
DK (1) DK0641143T3 (fr)

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Also Published As

Publication number Publication date
JPH0787589A (ja) 1995-03-31
DE59409989D1 (de) 2002-01-17
DK0641143T3 (da) 2002-04-02
ATE210362T1 (de) 2001-12-15
US5544249A (en) 1996-08-06
JP3565908B2 (ja) 2004-09-15
EP0641143B1 (fr) 2001-12-05
DE4328620C1 (de) 1995-01-19
EP0641143A3 (fr) 1999-05-19

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