EP0706745A1 - Procede de generation de son tridimensionnel - Google Patents

Procede de generation de son tridimensionnel

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Publication number
EP0706745A1
EP0706745A1 EP92924182A EP92924182A EP0706745A1 EP 0706745 A1 EP0706745 A1 EP 0706745A1 EP 92924182 A EP92924182 A EP 92924182A EP 92924182 A EP92924182 A EP 92924182A EP 0706745 A1 EP0706745 A1 EP 0706745A1
Authority
EP
European Patent Office
Prior art keywords
sound
digital
ratio
sample
generating
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.)
Withdrawn
Application number
EP92924182A
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German (de)
English (en)
Other versions
EP0706745A4 (fr
Inventor
David Chandler Platt
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.)
3DO Co
Original Assignee
3DO Co
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Filing date
Publication date
Application filed by 3DO Co filed Critical 3DO Co
Publication of EP0706745A4 publication Critical patent/EP0706745A4/fr
Publication of EP0706745A1 publication Critical patent/EP0706745A1/fr
Withdrawn legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/007—Two-channel systems in which the audio signals are in digital form

Definitions

  • PCT Patent Application Serial No. PCT/US92/09350 entitled METHOD FOR CONTROLLING A SPRYTE RENDERING PROCESSOR, by inventors Mical et al., filed concurrently herewith, Attorney Docket No. MDIO3040, and also to U.S. Patent Application Serial No. 07/970,278, bearing the same title, same inventors and also filed concurrently herewith;
  • PCT Patent Application Serial No. PCT/US92/09462 entitled SPRYTE RENDERING SYSTEM WITH IMPROVED CORNER CALCULATING ENGINE AND IMPROVED POLYGON-PAINT ENGINE, by inventors Needle et al., filed concurrently herewith, Attorney Docket No. MDI04232, and also to U.S. Patent Application Serial No. 07/970,289, bearing the same title, same inventors and also filed concurrently herewith;
  • PCT Patent Application Serial No. PCT/US92/09460 entitled METHOD AND APPARATUS FOR UPDATING A CLUT DURING HORIZONTAL BLANKING, by inventors Mical et al., filed concurrently herewith, Attorney Docket No. MDIO4250, and also to U.S. Patent Application Serial No. 07/969,994, bearing the same title, same inventors and also filed concurrently herewith;
  • PCT Patent Application Serial No. PCT/US92/G9461 entitled IMPROVED METHOD AND APPARATUS FOR PROCESSING IMAGE DATA, by inventors Mical et al., filed concurrently herewith, Attorney Docket No. MDIO4230, and also to U.S. Patent Application Serial No. 07/970,083, bearing the same title, same inventors and also filed concurrently herewith; and PCT Patent Application Serial No. PCT/US92/09384, entitled PLAYER BUS APPARATUS AND METHOD, by inventors Needle et al., filed concurrently herewith, Attorney Docket No. MDIO4270, and also to U.S. Patent Application Serial No. 07/970,151, bearing the same title, same inventors and also filed concurrently herewith.
  • the invention relates to a method for generating three dimensional binaural sound from monaural digital
  • the sound associated with the train not only give the queues as to the location of the train as it moves between the two locations, but to also include the doppler shift associated with the movement of the train as it moves toward or away from the player.
  • the listener has the means of controlling his relative position with regard to what is being observed on the screen, the sound being generated must reflect the relative movement made by the listener. In that the relative position of the sound source, i.e. object on the screen, and the listener is no longer fixed, a method must be used to produce the sound being generated on a real time basis.
  • the method to include the doppler shift associated with the relative movement of the object and the listener.
  • the preferred embodiment of the invention is to be implemented by a computer program.
  • the method calls for input data indicative of the location (X and Y coordinates) and the time associated with a start point and end point of travel of the object.
  • Inputs to the method can also comprise a descriptor of the amount of reverberation in the environment in which the action is taking place and, secondly, the relative loudness of the sound associated with the object.
  • One set of input data is called a segment.
  • the user continuously processes segments to define the relative movement of the object and the listener. The segments must be short enough in duration to allow the method to produce the proper sound as the player interacts with the system.
  • the method first determines if the input segment to the system meets the segment requirements of the system. If the segment is too large the segment is broken into subsegments until all subsegments meet the criteria of the method. The subsegments are ordered sequentially so they define the initial segment to the system. Each subsegment is then processed sequentially. From the input data associated with the segment or subsegment being processed, ratios are formed for both ears as well as the value for various multipliers used in the reverberation, frequency shaping, and amplitude control portion of the method. The method uses monaural digital sound samples stored in the memory. These monaural sound samples have been sampled at the compact disc (CD) audio rate of 44.1 kHz.
  • CD compact disc
  • the method will generate digital output sound samples at the same rate, i.e. 44.1 kHz.
  • a tick is the period of the frequency 44.1 kHz and is used as a basic time unit in the method.
  • the method uses the ratio for each ear to control the rate at which monaural sound samples for each ear are taken from memory.
  • the source sound samples are taken consecutively from the memory.
  • the sound represented by the source sound samples can be compressed or elongated in time to provide the effect of the doppler shift caused by the object moving towards or away from the listener.
  • each tick one digital output sound sample is generated for each ear.
  • the generated sound samples for each ear are processed separately.
  • the generated sound samples for each ear are processed for reverberation and passed through a combined notch and low pass filter for frequency shaping.
  • the samples are then processed for amplitude adjustment which is a function of the distance between the listener and the object and the relative loudness of the sound.
  • the processed digital output sound sample for each ear for each tick is stored in memory. Samples for each ear are taken from memory at the rate of 44.1 kHz and passed through a digital to analog converter.
  • the resulting analog signal for each ear is fed to respective sides of a set of earphones.
  • FIGURE 1 is a diagram indicating the movement of an object from point P ⁇ to P 2 and depicts the parameters and orientation used in the invention.
  • FIGURE 2 is a logic diagram representative of the method for generating a digital sound sample as a function of the ratio.
  • FIGURE 3 is a logic diagram representative of the method used for interpolation.
  • FIGURE 4 is a graph exemplifying 22 monaural sound samples used by the method.
  • FIGURE 5 is a graph showing the sound samples generated by the method using the ratio of 1.25 with reference to the center of the head.
  • FIGURE 6 is a graph showing the sound samples generated by the method using the ratio of 0.9 with reference to the center of the head.
  • FIGURE 7 is a graph showing the sound samples generated by the method for the off ear using the ratio of 1.23.
  • FIGURE 8 is a graph showing the sound samples generated by the method for the near ear using the ratio of 1.27.
  • FIGURE 9 is a logic diagram depicting the method practiced by the invention for introducing reverberation for both ears.
  • FIGURE 10 is a logic drawing depicting the combination notch filter and low pass filter used for providing waveshaping.
  • FIGURE 11 is a logic diagram depicting the function of volume adjustment as a function of distance, the relative loudness of the sound to be generated and the storage of the final digital sound sample in memory which is connected to a digital to analog converter to provide an analog output for each ear.
  • FIGURE 12 is a graph depicting alpha and beta values for the left ear as a function of the angle of the object at an average distance of 500 units.
  • FIGURE 13 is a graph depicting alpha and beta values for the right ear as a function of the angle of the object at an average distance of 500 units.
  • FIGURE 14 is a graph depicting the conversion tables for alpha and beta from decibels to units.
  • FIGURE 15 is a graph depicting the relationship of the volume adjust multiplier as a function of average distance and the relationship of the left and right ear reverberation multipliers as a function of distance in accordance with the methods described.
  • the method of this invention is used to generate the sound that an object associated with that sound would make as the object travels through three dimensional space.
  • the method generates the sound on a real time basis thereby allowing the sound generated to be responsive to the interaction between the listener and the system the listener is using.
  • One use for this method is in computer games which allows the viewer to interact with the computer system the viewer is using to play the game.
  • the use of this method is not limited simply to computer games and may be used wherever virtual sound is desired.
  • the method is carried out by a computer program stored within a processor that has a computer having the capacity and speed to perform the method included within the program.
  • This method is to be used with other programs which will provide segment data to the method containing necessary parameters to generate the given sound.
  • Table 1 lists the segment data provided to the method and the constants used by the method.
  • Table 2 lists the parameters that will be calculated in order to practice the method. All of the parameters to be calculated as show in Table 2 are well known in the art and can be found in any physics text.
  • the basic unit of distance is 0.1 meters.
  • the basic unit of time is derived from the CD recording rate of 44.1 kHz.
  • the basic unit of time is one cycle of that frequency and is referred to as a tick. Therefore there are 44,100 ticks per second.
  • the time between ticks, 226 sec is the amount of time that the processor has to perform the method practiced by this invention.
  • the method has five major parts.
  • the first part is segment determination for determining if the segment given to the system meets the requirements and criteria of the method.
  • the second part is the generation of a digital sound sample for each ear as a function of the position of the object and the listener for each tick of the segment. This portion also adjusts for the doppler effect caused by the object moving relatively toward or away from the listener.
  • the third portion is the addition of reverberation to the generated sound sample.
  • the user of the method can define the reverberation characteristics of the environment in which the object exists.
  • the fourth portion is frequency shaping for the purpose of inserting queues to the listener for positioning the object in three dimensional space.
  • the fifth portion is volume adjusting to account for the decrease in the loudness of the sound as the sound travels the distance between the listener and the object and for the relative initial loudness of the sound generated by the object. For example, a jet engine at 1,000 yards will be heard very clearly while a human voice at that same distance will not.
  • the method defines and accounts for the variable in initial loudness or power of the sound which is to be defined by the user.
  • Figure 1 illustrates the listener 10 observing an object going from point P ⁇ to point P 2 and the various parameters used by the method. It is understood that the location of the object is purely for exemplary purposes and the method operates for an object moving between any two points in three dimensional space around the listener. In the method the listener is always at the center of the coordinate system and the object moves relative to the listener. Where the listener can interact with the systems such that the listener can change the relative motion, such a change in relative motion is dealt with by rotating the axes of the system in an appropriate manner. The rotation of the coordinate system is not included within this method. Such a motion change would be reflected in the next segment of data sent to the method for sound generation.
  • a segment is defined as a start point P ⁇ and an end point P 2 . Both points are defined by X,Y coordinates in units of distance of the system.
  • the time, T- ⁇ and T 2 , at which the object is at P and P2 are also provided.
  • the method will first determine if the segment defined meets the criteria of the method.
  • the criteria are:
  • angle B ⁇ is less than 5°, and if angle B 2 is less than 5°, and if the difference between distance d ⁇ and dm is less 5% of dm, and if the difference between d 2 and d m is less 5% of d m , then use the segment as presented otherwise divide the segment into subsegments.
  • the segment is divided at the midpoint generating two new segments.
  • the first portion of the first subsegment would have a start point Pi, a stop point P m , and a start time T* ⁇ , and an end time T m .
  • the second segment would have as its parameters the
  • the method next generates sound samples at the rate of 44.1 kHz that will correspond to the sound generated by the object as the object moves between P and P2.
  • the time that sound would be generated by the object is ⁇ T, the difference between time Ti and T 2 • Since the object is moving away from the listener in Figure 1, the time t2 for sound to reach the listener from the end point P 2 will be greater than the time ti for sound to reach the listener from the start point P*j_. Therefore the listener will hear sound for a longer period of time than the time sound was actually generated by the object. Conversely, the listener will hear sound for a shorter period of time than the time sound was generated by the object as it moves from Pi to P 2 . This gives rise to a change in pitch of the sound which is commonly referred to as the doppler effect.
  • the method further takes into account the difference in time for the sound to be heard by the right and left ear of the listener as a function of the location of the sound in the coordinate system as shown in Figure 1.
  • the listener's right ear will receive sound waves earlier than the left ear.
  • the method generates for each ear a ratio between the length of time that the sound would have been generated by the object as it moved from Pi to P2, to the length of time that the listener hears the sound that would have been generated as the object moved from Pi to P2•
  • the ratio includes a correction factor to adjust for the location of the object with reference to the listener's ears.
  • the ratio is first derived for the center of the head as follows:
  • d2/s are in units of decimeters/sec.
  • Ratio (right ear) is:
  • Ratio (left ear) is:
  • Equations (10) and (11) can be used when the criteria for segment determination has been used to limit the size of the segment. If the criteria for segment determination has not been used or made less stringent, then equations (8) and (9) should be used for the ratios.
  • the ratios for the right and left ear are generated once for each segment and is used throughout the segment for the purpose of generating sound samples.
  • a sound sample is generated for each ear for each tick in the segment. It is envisioned that a segment will be changed once to twice a second. Therefore a segment will have 22,000 to 44,100 ticks and thus 22,000 to 44,100 sound samples will be generated for each ear for each segment.
  • the method at this stage is divided for the right and left ear. The description hereinafter will be with regard to the right ear, but it should be understood that the same processing is done for the left ear.
  • Ratio for the right ear, RR will be composed of two parts, an integer part and a fraction part. Where the object is going away from the listener, the integer part will be 0 and if the object is coming towards the listener, then the integer portion of the ratio will be 1. For each cycle of operation, i.e. a tick, the ratio RR is added to the fractional portion of the summation ratio of the previous cycle. The results of this addition leads to a summation ratio for the right ear. Thus, by adding a fractional portion of a previous number to a number having an integer and fractional portion, the resulting summation ratio can have the integer portion to be equal to 1 if originally 0, and 2 if originally 1.
  • the integer portion is then used to select the samples from the monaural digital samples for the sound that has been stored in memory for use with this process.
  • the user of this method can store any monaural digital sampled sound in the memory. It is further well understood in the art that the time necessary for the monaural digital sound samples to describe the actual sound may be short. Therefore the monaural digital sound samples associated with the sound are looped so as to give a continuous source of digital sound samples of the sound to be generated. In the event there is a requirement that the monaural digital sound samples are greater in number than the allocated memory space, it is well within the art for the user to update the memory with new monaural digital sound samples as they are needed.
  • the integer portion of the summation ratio is used to control the number of monaural sound samples withdrawn from memory.
  • the last two monaural sound samples that have been retrieved from the memory are used to generate the digital sound sample for the present tick.
  • Interpolation is done between the two values of the sound sample using the fractional portion of the summation ratio. The interpolated value then becomes the generated digital sound sample for that tick for further processing during the tick.
  • Figure 2 is a logic diagram which logically depicts this portion of the method.
  • Memory 21 stores the fractional portion D of the summation that results from adder 22.
  • Adder 22 has as its inputs the ratio for the right ear RR and the fractional portion D of the previous summation from the previous cycle stored in memory 21. After a new summation is done by adder 22, the new fractional portion D is stored in memory 21 to be used during the next cycle or tick.
  • the integer portion I is sent to a comparator 23 to be tested. If the integer I is equal to 1 then the next monaural sound sample value is taken from memory 24 and stored in a two-stage FIFO 25. If the integer is equal to 2 then the next two monaural sound samples are taken from memory 24 and transferred to FIFO 25.
  • FIFO 25 If the tick has an integer value equal to 0 then the previous two fetched monaural sample values will still exist in FIFO 25. If one sample was fetched then FIFO 25 would have the previous sample that was fetched from memory 25 plus the present sample that has been fetched during this cycle. If two new samples were fetched from the memory 24, FIFO 25 will have the two samples stored this cycle. Interpolation is then done by interpolator 26 between the values of the two samples in FIFO 25 using the fractional portion D of the summation ratio. The interpolator 26 generates a digital sound sample for further processing by the remaining portion of the method.
  • Figure 3 is a logic diagram of the interpolator 26 of Figure 2.
  • the interpolation is straightforward.
  • the digital values stored in the second stage of FIFO 25 are subtracted from the digital values stored in the first stage of FIFO 25 yielding the difference between those two values.
  • This difference (A - B) is then multiplied by multiplier 32 by the fractional portion D of the summation ratio to yield D(A - B) .
  • the output of multiplier 32 is then added together with the digital value of the first stage of FIFO 25 by adder 33 yielding the interpolated value for the sound sample for that tick.
  • Figure 4 depicts 22 monaural sound samples stored in memory 24. In this example it is assumed that the samples of Figure 4 are the middle of the segment being processed. The numbers assigned to the sample numbers are for convenience.
  • Table 3 demonstrates the generation of output value of the sound samples where the ratio to the center of the head is equal to 1.25.
  • the summation ratio (SUM RATIO) illustrates the addition of the ratio 1.25 being added to the fractional portion of the preceding summation ratio. It is assumed that the value of the fractional portion of the preceding summation ratio prior to the start of this example was 0.
  • the table further shows the monaural sample values used and the output value for the sound sample after interpolation.
  • Figure 5 is a graph of the output values of Table 3. It should be understood that at the start of this example FIFO 25 would have included samples 1 and 2 within that FIFO and, therefore, in Figure 4, samples 1 and 2 have already been read from memory 24. The next value that would be read from memory 24 would be sample 3. At the end of the example, sample 22 has been read from the memory store and is stored in FIFO 25.
  • Table 4 is another example for the generation of sound samples in accordance with the method.
  • Table 4 assumes the ratio R R to be 0.9, that is the object is moving away from the listener rather than toward the listener.
  • Figure 6 illustrates the sound samples generated by the method from the monaural sound samples of Figure 4. Again, sound samples 1 and 2 of Figure 4 were stored in FIFO 25 at the start of the example and it was assumed that the fractional portion D of the resulting summation ratio was equal to 0 at the start of the example. Table 4 indicates that for the first cycle of operation the integer portion of the summation ratio is 0, thus no new samples will be extracted from memory 24 and the existing values of samples 1 and 2 in FIFO 25 will be used.
  • Figures 7 and 8 can be compared with Figure 5.
  • the number of ticks generated are the same but the magnitude of each of theticks are different.
  • the off ear Figure 8) one additional tick was necessary for the method than for the near ear. Again, each of the ticks are of a different magnitude than that for the center of the head. Thus, the near and the off ear will have a different set of generated sound samples for the same segment.
  • Figure 9 logically depicts the method for introducing reverberation.
  • the generated sound sample for the right ear is first multiplied by multiplier 91 and then added together with the output of multiplier 93.
  • the values of the multiplication factors for multipliers 91 and 93 when added together are equal to 1.
  • the input to multiplier 93 is the output of the reverberation buffer 94.
  • the reverberation buffers 93 and 94 are analogous to a FIFO buffer where the oldest sample stored in the buffer is the sample that will next be multiplied by multiplier 93 and added by adder 92 to the output of multiplier 91.
  • reverberation buffer 94 The input of reverberation buffer 94 is the output of adder 98 which is the adder associated for the left ear. Thus there is cross- coupling between the two ears in the reverberation section of the method. It has been found that reverberation buffers 94 and 95 should be of different lengths. In the present embodiment reverberation buffer 94 is 2,039 ticks and reverberation buffer 95 is 1,777. This delay equates to a 46 and 40 millisecond delay respectively for buffers 94 and 95.
  • the upper limit of the multiplication factor for the multiplier 96 and 93 is 0.5.
  • the maximum reverberation level is set to 100 units on a scale of 256 units, or a decimal value of 0.391. It has further been found that it is desirable to not only have the delay for each ear different but also the amount of reverberation should be different for each ear. To this end, the method calls for a 5% reduction in the reverberation level between the two ears.
  • Reverberation is a function of distance from the listener to the object that would be generating the sound. The greater the distance the greater the reverberation for a given set of reverberation characteristics. It is desirable to have some reverberation in all cases and, therefore, a minimum reverberation level is set to 20 on a scale of 256 or 0.078.
  • Figure 15 is a graph which shows the reverberation levels settings for the two ears as a function of distance. It has been found convenient in this . method to use a scaling factor of 256 for calculating the various values of multiplied functions used throughout the method.
  • multiplier number 93 0.195. This would cause the setting of multiplier 91 to be equal to 0.805 in that the multiplication factors for the multiplication steps illustrated by multipliers 90 and 91 must equal 1.
  • the multiplier factors for the left ear would be set slightly different in that multiplier factor associated with multiplier 96 is set at a value of 95% of the multiplication factor represented by multiplier 93.
  • the resulting value of the multiplication factor associated with the multiplier 96 would be 0.185, which in turn would cause the multiplication factor associated with multiplier 97 to be 0.815. Again, the summation of the multiplication factors associated with multipliers 96 and 97 must equal 1.
  • the method defines the reverberation level to equal the MINIMUM REVERBERATION plus the sum of the average distance d m minus the NEARBY value divided by the REVERBERATION STEPS.
  • the MINIMUM REVERBERATION is 20, NEARBY is 100 units of distance, REVERBERATION STEP is 10 and MAXIMUM REVERBERATION is 100. Therefore the settings of the reverberation multiplication factor is a function of the distance between the listener and the object, taking into account maximum and minimum reverberation values allowed.
  • the user of the system is given the prerogative of setting the values for MAXIMUM REVERBERATION level, MINIMUM REVERBERATION, NEARBY and REVERBERATION STEP.
  • the user has the freedom to control the reverberation characteristics which the user wishes to have associated with the sound being generated.
  • the user can, by adjusting these values, have the reverberation sound like the object and listener are in a tunnel or in a deadened sound-proof room.
  • a sound generated in three dimensional space has its frequency components filtered by the media through which it is travelling such that the sound heard by the listener has the frequency components of the original sound substantially altered.
  • the first phenomenon is the greater the distance sound has to travel, the greater the high frequency components of the sound are attenuated.
  • a second phenomenon is that of head shadowing of the sound where low frequencies easily go around the head while the high frequencies are blocked or attenuated.
  • Another phenomenon is the sensitivity peak for the normal ear in the range of 7 - 8 kHz. Since it is envisioned that the listener will be wearing earphones, all sound will be subject to this phenomenon. It has been understood that the effects of this phenomenon is a function of the location of the object making the sound with respect to the listener such that it is maximum when the object making the sound is perpendicular to an ear of the listener and minimum when the object making the sound is in front or behind the listener. Since earphones are used any sound queue with regard to this phenomenon that would have been attainable are destroyed because the earphones are directly perpendicular to the listener's ear.
  • the method adjusts for this phenomenon by having a notch filter at 7 - 8 kHz where the depth of the notch is a function of the object's location relative to the listener.
  • the notch of the notch filter is approximately 0, leaving the phenomenon to exist in its natural state.
  • the depth of the notch of 7 - 8 kHz is increased to a maximum level of 5db when the object is either directly in front of or to the rear of the listener.
  • FIG 10 is a logic illustration of the combined digital notch and low pass filters used for waveshaping for location queueing.
  • the notch filter is comprised of a three sample delay 101, two multipliers, 102 and 103, and adder 104.
  • the low pass filter is shown as a one sample delay 106 and multiplier 105.
  • the output of multiplier 105 is added to the output of multipliers 103 and 102 by adder 104.
  • methodology has been established to emulate the frequency shaping that is provided in the natural environment for sound generated in three dimensional space.
  • beta is calculated for the right and for the left ear as follows:
  • Beta left ear ((d m - NEARBY) 100) + l ⁇
  • Beta right ear ((d m - NEARBY) 100 + l ⁇
  • Beta is used to account for distance roll-off, rear head and side head shadowing.
  • a second value (alpha) is calculated for each ear as follows:
  • Alpha left ear Beta left ear 2 + 5
  • Alpha right ear Beta right ear 2 + 5
  • Alpha values depend on the beta values thereby allowing flatter and lower knee of the high frequency roll-off characteristics of the filters.
  • controls the notch of the notch filter as a function of the position of the object and limits the notch to be 5 decibels. Most earphones have designed compensation for the frequency at 7 - 8 kHz.
  • alpha and beta are in decibels and it is necessary to convert those values into multiplication factors for the various multiplication functions to be performed within the digital filters.
  • a scale of 256 has again been used and by experimentation the following tables have been generated. As can be seen from Table 6, the maximum decibel level that would be allowed in the alpha table is 20db and, therefore, if the value of alpha should be greater than 20, it would be limited to the value for 20db.
  • Table 7 shows that if the value of beta should be greater than 9db, the value will be limited to 9db.
  • alpha and beta is obtained by interpolation.
  • the results of the interpolation are always rounded to the nearest whole number. Because the filters used were combined the alpha value must be adjusted.
  • the alpha value is mapped into the remaining scale units not used by beta such that alpha will have the same percentage of the remaining units that it had in the original scale.
  • the value of alpha is obtained as follows:
  • Figure 14 is a plot of the conversion scale of Tables 6 and 7 for alpha and beta.
  • Figure 12 is a graph showing the values for alpha and beta in decibels as a function of the average angle ⁇ m for a constant average distance d m , of 500 units for the left ear.
  • Figure 13 shows the value of the alpha and beta in decibels as a function of the average angle ⁇ m for a constant average distance d m of 500 units for the right ear.
  • the resulting values for the alpha and beta for the right and left ear will be different for the same average distance dm for the sum average angle ⁇ m .
  • Each of the generated samples after being altered for reverberation, are processed through the digital filters using the values for the multiplication functions as herein described.
  • the output of the filters is a filtered sound sample.
  • FIG. 11 is a logic diagram illustrating the method of the invention.
  • the digital sound sample is first multiplied by multiplier 111.
  • Multiplier Ill's multiplication factor is determined as a function of the distance from the listener to the object. Once again a scale from 0 - 256 is used.
  • FIG. 15 is a graph which depicts the volume setting as a function of distance in units. For example, if the average distance d m was 1,000 units (100 meters), the resulting volume adjust level would be 25.6. When converted to a decimal value the resulting multiplication factor would be 0.1.
  • multiplier 111 The output of multiplier 111 is then multiplied by multiplier 112.
  • the multiplier factor associated with multiplier 112 is set by the user and determines the relative loudness or strength of the sound for the various sounds being generated by the method.
  • the output of the multiplier 112 or the multiplication function associated therewith is then stored in a memory 113.
  • the digital sound samples are taken from memory 113 at the rate of 44.1 kHz.
  • the output of the memory 113 is in turn sent to a digital- to-analog converter 114.
  • the output of digital analog converter 114 will be an analog signal which when processed by earphones will generate sound where the sound will be representative of the sound that would have been generated by the object as that object moves from Pi to P2- Again, it is understood that there is a channel for generating and processing sound samples for each ear and that the resulting analog signal for each ear will be different.
  • a fully computerize implementation of the invention as described heretofore uses known digital software implementations.
  • a program written in programming language C is provided in Appendix A. This program practices the method of the invention consisting of the five portions as set forth above.
  • segment determination portion reverberation portion
  • frequency shaping portion reverberation portion
  • volume adjust portion may be deleted. The omission of one or more of these portions will effectively lose some sound queues as to the location of the object and will decrease the quality of the sound produced.
  • R4-R7 R8-R11 104-107 200-203
  • General input/control 10C-10F 208-20B Left ear, dopplcr/plcker 114-117 210-213 Left ear, filter l lC-U F 218-21B Right ear, doppler/picker 124 -127 220-223 Right ear, filter 12C-12F 228-22B Final volume/mixing for both ears
  • NOP ' keep following RBASE from executing
  • RO reverb level Rl nonreverb level (1.0 - reverb level)
  • R2 Doppler pick rate integer portion
  • R3 Doppler pick rate fractional portion
  • ALU add 0x8000 to accume t calc F - 1.0
  • MAXREVERBTIME must be a power of 2, minus 1' */
  • SAMPLESPERUNIT defines the number of sound samples per measuring unit (currently 14 samples per decimeter unit... this misestimates the speed of sound by about 4%) .
  • a sound source located within NEARBY units of the observer will be heard at full volume. If further away than NEARBY, the volume will be reduced in proportion to the distance. Minimum volume is set by VERYQUIET.
  • a sound source located within NEARBY units of the observer will have a reverberation level of MINREVERB. If further away than NEARBY, the reverb level will be increased to a maximum of MAXREVERBLEVEL, with an increase of one reverb unit per REVERBSTEP units of distance Reverb time constants are set by REVERBTCl and REVERBTC2... prime numbers which correspond roughly to 50 and 40 milliseconds.
  • I* intBits and fractBits define the portions of a 32-bit word to be used for a fixed-point number.
  • the integer portion must be long enough to hold a signed number twice the magnitude of MAXDELTA.
  • fractMask must correspond to the rightmost fractBits in the word. /*
  • Alpha cuts are available from 0 dB (flat) to 20 dB.
  • */ 0 int alphaTable.NUMALPHAVALUES] ⁇ 0, 15, 27, 38, 48, 56, 64, 72, 77, 82, 88, 92, 96,
  • Beta cuts are available from 0 dB (flat) to 9 dB.
  • V mt betaTable[NUMBETAVALUES] (0, 15, 30, 46, 59, 73, 87, 102, 112, 127); 5 /-
  • leftEar->hfRolloff • rearRolloff + distanceRolloff
  • rightEar->hfRolloff 10 * sin(-avgAngle) + rearRolloff + distanceRolloff ;

Landscapes

  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)

Abstract

Un procédé permet de générer des sons tridimensionnels associés à un objet qui se déplace d'une première position (P1) vers une deuxième position (P2) par rapport à l'auditeur (10). Le procédé intègre les effets du déplacement Doppler, de l'occultation par la tête et de la distance sur les composantes de fréquence du son ainsi que sur le volume du son, et la sensibilité naturelle de l'oreille humaine dans la plage entre 7 et 8 kHz. Le procédé crée une séquence d'échantillons audio-numériques qui lorsqu'ils sont convertis en formes d'ondes analogiques pour produire des signaux audio produisent des signaux audio qui forment des files de sons perçues par l'auditeur de manière à situer le son dans un espace tridimensionnel.
EP92924182A 1992-11-02 1992-11-02 Procede de generation de son tridimensionnel Withdrawn EP0706745A1 (fr)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
PCT/US1992/009348 WO1994010815A1 (fr) 1992-11-02 1992-11-02 Procede de generation de son tridimensionnel

Publications (2)

Publication Number Publication Date
EP0706745A4 EP0706745A4 (fr) 1995-11-20
EP0706745A1 true EP0706745A1 (fr) 1996-04-17

Family

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EP92924182A Withdrawn EP0706745A1 (fr) 1992-11-02 1992-11-02 Procede de generation de son tridimensionnel

Country Status (4)

Country Link
EP (1) EP0706745A1 (fr)
JP (1) JPH08502636A (fr)
AU (1) AU3058792A (fr)
WO (1) WO1994010815A1 (fr)

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0036337A2 (fr) * 1980-03-19 1981-09-23 Matsushita Electric Industrial Co., Ltd. Système de reproduction sonore comportant des circuits de localisation d'image sonore
GB2238936A (en) * 1989-12-07 1991-06-12 Q Sound Ltd Sound imaging system for video game

Family Cites Families (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS53137101A (en) * 1977-05-06 1978-11-30 Victor Co Of Japan Ltd Signal converter
JPS5868400A (ja) * 1981-10-19 1983-04-23 Matsushita Electric Ind Co Ltd 音像定位制御方式
US4731848A (en) * 1984-10-22 1988-03-15 Northwestern University Spatial reverberator
JPS62140600A (ja) * 1985-12-13 1987-06-24 Matsushita Electric Ind Co Ltd 音響効果装置
US4817149A (en) * 1987-01-22 1989-03-28 American Natural Sound Company Three-dimensional auditory display apparatus and method utilizing enhanced bionic emulation of human binaural sound localization
US5046097A (en) * 1988-09-02 1991-09-03 Qsound Ltd. Sound imaging process

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0036337A2 (fr) * 1980-03-19 1981-09-23 Matsushita Electric Industrial Co., Ltd. Système de reproduction sonore comportant des circuits de localisation d'image sonore
GB2238936A (en) * 1989-12-07 1991-06-12 Q Sound Ltd Sound imaging system for video game

Non-Patent Citations (4)

* Cited by examiner, † Cited by third party
Title
AES, vol. 39, no. 11, November 1991 NEW YORK, pages 864-870, XP 000436498 D.R.BEGAULT 'CHALLENGES TO THE SUCCESSFUL IMPLEMENTION OF 3-D SOUND.' *
AES, vol. 39, no. 9, September 1991 NEW YORK, pages 604-622, XP 000226141 E.A.MACPHERSON 'A COMPUTER MODEL OF BINAURAL LOCALIZATION FOR STEREO IMAGING MEASUREMENT' *
ELECTRONICS & COMMUNICATIONS IN JAPAN, vol. 68, no. 4, April 1985 SILVER SPRING,MARYLAND, pages 54-63, KUROZUMI AND AL. 'METHOD OF CONTROLLING SOUND IMAGE DISTANCE BY VARYING THE CROSS-CORRELATION COEFFICIENT BETWEEN TWO-CHANNEL ACOUSTIC SIGNALS.' *
See also references of WO9410815A1 *

Also Published As

Publication number Publication date
JPH08502636A (ja) 1996-03-19
AU3058792A (en) 1994-05-24
WO1994010815A1 (fr) 1994-05-11
EP0706745A4 (fr) 1995-11-20

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