Stereo audio signal processing system

Abstract

A system (100) for processing a stereo audio signal comprises a first input (I1) and a second input (I2), a first output (U1) and a second output (U2), a first summing block (S1) connected to the first input (I1) and the first output (U1), a second summing block (S2) connected to the second input (I2) and the second output (U2), a first delay block (D1) connected to the first input (I1) and the second summing block (S2), and a second delay block (D2) connected to the second input (I2) and the first summing block (S1).

Description

Technical Field

[0001] The present invention relates to a stereo audio signal processing system suitable for transmitting to acoustic headphones, such that the signal heard by a user wearing the headphones is as similar as possible to the virtual listening obtained with two or more speakers in an anechoic listening room.

[0002] As is known to all users accustomed to listening to audio signals (such as music works) using headphones, not limited to professional users, audio enthusiasts, sound operators, etc., the perception of music works using headphones is different from the perception obtained from two or more speakers in a listening room without headphones.

[0003] In the case of a basic stereo with two speakers, the main reason for the difference between the signal perceived by headphones and the signal perceived from two speakers is that the acoustic signal from the left speaker first reaches the left ear and then reaches the right ear after a time interval. Such a time interval is due to the propagation of sound in the space between the left and right ears.

[0004] The same phenomenon also occurs with the sound emitted from the right speaker, which first reaches the right ear and then the left ear.

[0005] Therefore, in the case of two speakers and one listener, the direct acoustic signal from the left speaker and the delayed acoustic signal from the right speaker reach the left ear. The direct acoustic signal from the right speaker and the delayed acoustic signal from the left speaker reach the right ear. In contrast, in the case of headphone listening, the sounds reproduced by the right and left earpieces are perceived by the right and left ears without any delay.

[0006] Such a phenomenon was analyzed as early as the 1960s (Robert Larson and John Eagle, Audio Magazine, November 1962 issue). A clearer electronic system that can improve headphone listening was devised by engineer Siegfried Linkwitz and published in "Improvement of Headphone Listening" in the December 1971 issue of Audio Magazine, and is still on the market today. The said system has been improved over the years, but is based on experimental and empirical principles and intuition.

[0007] The document by Siegfried Linkwitz describes a device suitable for placement between the sound source and the headphones. Such a device has two delay blocks that delay the sound signals going to the right and left earpieces of the headphones. The delay blocks are RC resonators, i.e., resonant circuits made of capacitance and resistance that generate a delay that changes with the change in the frequency of the signal. As a result, different delays occur for low-frequency signals and high-frequency signals, and high-fidelity sound reproduction cannot be obtained.

[0008] In "Improvement of Stereo Headphone Sound Image", Thomas M.V. describes a device suitable for placement between the sound source and the headphones. Such a device has two delay blocks that delay the sound signal going to the right earpiece of the headphones and the sound signal going to the left earpiece. The delay blocks are set to maintain a constant phase of the signal. To maintain a constant phase of the signal, it is necessary to change the delay with the change in the frequency of the signal. Figure 6 of the said document shows that the delay changes with the change in frequency and decreases at frequencies above 4 kHz. As a result, with such a device, high-fidelity sound reproduction is not guaranteed at high frequencies above 4 kHz.

[0009] "Improving Externalization and Frontal Recognition of Headphone Signals" by Vainrich, Soren Grett describes a device suitable for placement between a sound source and headphones. Such a device comprises two delay blocks that delay the sound signal directed towards the right earpiece of the headphones and the sound signal directed towards the left earpiece. This document does not disclose how the delay of each delay block is calculated, and it is certain that it does not disclose or suggest any relationship between the position of the speaker and the delay for a user wearing headphones. In this document, the delay of the block is stated to be between 50 μs and 2000 μs. Such a wide range deviates from the fact that the delay is calculated based on the inclination angle of the speaker with respect to the listener. In fact, assuming that the inclination angle of the speaker with respect to the listener is 90°, to give the maximum delay value, for a delay of 2000 μs, the distance between the earpieces would have to be approximately 68 cm, but since the distance between the two earpieces is generally in the range of 15 cm to 18 cm, this is clearly illogical.

[0010] There are other phenomena that can cause a difference between listening with headphones and listening without headphones, such as internal reflections in the listening room, high-power signals entering the nasal cavity, and low-frequency signals that vibrate the stomach and bones.

[0011] In music works from the 1960s when high-fidelity techniques (Hi-Fi techniques) in recording studios were not advanced, there was often a lack of low-frequency sounds. Figure 1a shows the distribution of the sound (S1) that can be produced from music works from the 1960s, accompanied by a typical lack of frequencies below 100 Hz.

[0012] To solve such drawbacks, a tone control known as "Baxandal type" is known, which is found in most audio amplifiers and enables the enhancement or attenuation of sound in two or more frequency bands of an acoustic signal.

[0013] The Baxandall - type tone control has an adjustable knob for selecting the frequency band to enhance the sound. Depending on the position of the knob, the Baxandall - type tone control has a response for enhancing low - frequency sounds of the type shown in FIG. 1b. FIG. 1b shows a family of 12 board diagrams as a function of the knob position. Specifically, the knob has 12 positions. The board diagrams are first - order (6 dB / octave) transfer functions (G1,...G12), characterized by a variable - frequency pole and a fixed zero at an average frequency, for example, 1 kHz.

[0014] When the knob is moved to enhance frequencies below 100 Hz, from the transfer function of the diagram in FIG. 1b, the transfer function G7 shown in FIG. 1c, which has a pole at 100 Hz, is selected. In other words, the transfer function at knob position 7, which has a pole at 100 Hz, a zero at 1 kHz, and as a result, a gain of 20 dB, is selected. Thus, a filter with the transfer function G7 is generated.

[0015] However, when multiplying the sound distribution (S1) in FIG. 1a by the G7 transfer function of the filter selected by the knob (thus adding the dB values of the two diagrams), as a result, a distribution (S2) shown in FIG. 1d, which has a zero at the origin, double poles at 100 Hz, and a zero at 1 kHz, is obtained. Therefore, in the resulting distribution (S2), the sound is enhanced even in the low - to - mid - frequency range (100 Hz - 1 kHz), and the final tone of the sound changes.

[0016] An object of the present invention is to solve the drawbacks of the prior art by providing a stereo audio signal processing system suitable for reproducing an acoustic signal with high fidelity and high realism similar to the perception of a user listening through a speaker.

[0017] Another object is to provide a reliable stereo audio signal processing system.

[0018] A further object is to provide a stereo audio signal processing system that is practical, highly versatile, and suitable for implementation in acoustic headphones.

[0019] An additional object is to provide a stereo audio signal processing system suitable for compensating for a lack of bass resulting from problems in a recording system or a sound diffusion system in sound.

[0020] These objects are achieved by the present invention having the features of the appended independent claims.

[0021] Advantageous effects of the present invention are manifested from the dependent claims.

Brief Description of the Drawings

[0022] Further features of the present invention will become more apparent from the following detailed description with reference to the embodiments shown in the accompanying drawings, which are purely illustrative and not limiting.

Figure 1a

Figure 1b

Figure 1c

Figure 1d

Figure 2

Figure 3

Figure 4

Figure 5

Figure 6a

Figure 6b

Figure 6c

Figure 6d

[0023] Figure 2 shows a typical listening situation using a speaker or loudspeaker in an anechoic chamber. The anechoic chamber has a left speaker (DL) placed on the left side of the listener (U) and a right speaker (DR) placed on the right side of the listener (U).

[0024] The listener (U) has a left ear (L) and a right ear (R). A segment (LR) connecting the left ear and the right ear is drawn on a plane (corresponding to the drawing sheet plane of Figure 2). The segment (LR) has a center (O). A first straight line (r1) connects the center of the left speaker (DL) to the center (O) of the segment (LR). A second straight line (r2) connects the center of the right speaker (DR) to the center (O) of the segment (LR). A central straight line (r) passes through the center (O) of the segment (LR) and is perpendicular to the segment (LR).

[0025] The first angle (A) between the central line (r) and the first line (r1) defines the inclination of the left speaker (DL). The second angle (B) between the central line (r) and the second line (r2) defines the inclination of the right speaker (DR). The first angle (A) and the second angle (B) can vary from 0° to 90° depending on the position of the speakers relative to the listener (U).

[0026] The segment (LR) has a length (D) equal to the distance between the listener's ears (L, R) and is typically in the range of 140 mm to 180 mm.

[0027] The left speaker (DL) emits sound that first reaches the left ear (L) and then reaches the right ear (R) which is clearly separated from the left speaker.

[0028] In fact, for the sound from the left speaker (DL) to reach the right ear after reaching the left ear once, the sound needs to cover the left-to-right movement (SL). Such left-to-right movement (SL) depends on the position of the left speaker (DL) and is given by the following formula. SL = D * sin(A)

[0029] In such a case, considering that the sound travels in the air, the average speed of sound (SS) can be assumed to be 343 m / s.

[0030] To cover the left-to-right movement (SL), the following left-to-right delay time (TL) given by the following formula is applied to the sound. TL = SL / SS

[0031] Similarly, the sound from the right speaker (DR) first reaches the right ear (D) and then reaches the left ear (L).

[0032] For the sound from the right speaker (DR) to reach the left ear after reaching the right ear once, the sound needs to cover the right-to-left movement (SR). Such right-to-left movement (SR) is given by the following formula. SR = D * sin(B)

[0033] To cover the right-to-left movement (SR), the following right-to-left delay time (TR) given by the following formula is applied to the sound. TR = SR / SS

[0034] The sound emitted from the left speaker and the sound emitted from the right speaker can be identified as acoustic signals.

[0035] EL represents the direct acoustic signal emitted from the left speaker (DL), which reaches the left ear (L) without delay.

[0036] EDL represents the delayed acoustic signal emitted from the left speaker (DL), which reaches the right ear (R) with a delay time (TL).

[0037] ER represents the direct acoustic signal emitted from the right speaker (DR), which reaches the right ear (R) without delay.

[0038] EDR represents the delayed acoustic signal emitted from the right speaker (DR), which reaches the left ear (L) with a delay time (TR).

[0039] Therefore, in the case of two speakers and one listener, the left ear (L) receives the direct acoustic signal (EL) from the left speaker and the delayed acoustic signal (EDR) from the right speaker. Conversely, the right ear (R) receives the direct acoustic signal (ER) from the right speaker and the delayed acoustic signal (EDL) from the left speaker.

[0040] The object of the present invention is to imitate the state of sound as described above in the right earpiece and left earpiece of the acoustic headphones.

[0041] Referring to FIG. 3, a system according to the present invention, comprehensively indicated by reference numeral 100, is shown.

[0042] This system (100) is used to process stereo audio signals so that a listener (U) wearing acoustic headphones (2) perceives sound as coming from a left virtual speaker (DL) located on the left side of the listener and from a right virtual speaker (DR) located on the right side of the listener. In other words, the left earpiece (20L) of the acoustic headphones should emit a sound similar to the sound that the listener would perceive if a physical speaker were placed on the left side of the listener. Clearly, since such a speaker does not exist within the system, it is referred to as the left virtual speaker (DL) and is shown by a dashed line in FIG. 3. Instead, the right earpiece (20R) of the acoustic headphones should emit a sound similar to the sound that the listener would perceive if a physical speaker were placed on the right side of the listener. Clearly, since such a speaker does not exist within the system, it is referred to as the right virtual speaker (DL) and is shown by a dashed line in FIG. 3.

[0043] The system (100) comprises a first input (I1) and a second input (I2), which are suitable for connection to a sound source (1) suitable for transmitting a stereo electroacoustic signal, a first output (U1) and a second output (U2), which are suitable for connection to acoustic headphones (2) comprising a left earpiece (20L) and a right earpiece (20R). and comprises.

[0044] By way of example, the sound source (1) may be a CD player, a Hi-Fi system, an audio mixer, a smartphone, a tablet, or the like.

[0045] The stereo signal emitted from the sound source includes a first acoustic signal (E1) transmitted to the first input (I1) and a second acoustic signal (E2) transmitted to the second input (I2). In the case of stereo transmission with two speakers, the first acoustic signal (E1) is the signal transmitted to the left speaker, and the second acoustic signal (E2) is the signal transmitted to the right speaker.

[0046] The first input (I1) is connected to the input of the first addition block (S1).

[0047] The second input (I2) is connected to the input of the second addition block (S2).

[0048] The first input (I1) is connected to a first delay block (D1) connected to the second addition block (S2). The first delay block (D1) imparts a first delay (T1) to the first delay signal (E1), and a first delayed acoustic signal (ED1) that is added to the second acoustic signal (E2) is output from the first delay block (D1). The second addition block (S2) has an output connected to the second output (U2), from which the first delayed acoustic signal (ED1) is output in addition to the second acoustic signal (E2).

[0049] The second input (I2) is connected to a second delay block (D2) connected to the first addition block (S1). The second delay block (D2) imparts a second delay (T2) to the second acoustic signal (E2), and a second delayed acoustic signal (ED2) that is added to the first acoustic signal (E1) is output from the second delay block (D2). The first addition block (S1) has an output connected to the first output (U1), from which the second delayed acoustic signal (ED2) is output in addition to the first acoustic signal (E1).

[0050] In such a case, the first input signal (E1) corresponds to the acoustic signal (EL) emitted from the left speaker (DL) in the embodiment of FIG. 2, and the second input signal (E2) corresponds to the acoustic signal (ER) emitted from the right speaker (DR) in the embodiment of FIG. 2.

[0051] The first delay block (D1) is appropriately configured to produce a first delay time (T1) that is independent of the frequency of the signal but depends on the position of the left virtual speaker (DL) relative to the listener (U) and the distance (D) between the listener's ears (L, R). In other words, once the position where the listener wants to place the left virtual speaker (DL) and the distance (D) between the listener's ears are known, the first delay time (T1) is set, and this remains constant even when the frequency of the signal changes.

[0052] More precisely, the first delay time (T1) is equal to the time it takes for the sound signal emitted from the left speaker (DL) to cover the left-to-right movement (SL).

Number

[0053] Assuming that sound has an average speed (SS) of 343 m / s, the first delay time (T1) is a function of the distance (D) between the listener's ears and the inclination angle (A) of the left speaker with respect to the listener. Thus, the first delay time (T1) is between 0 when A = 0° and D / 343 when A = 90°. Considering that D is generally 15 cm to 18 cm, T1 can be in the range of 0 μs to 525 μs.

[0054] The second delay block (D2) is appropriately configured to produce a second delay time (T2) that is independent of the frequency of the signal but depends on the position of the right virtual speaker (DR) relative to the listener (U) and the distance (D) between the listener's ears. In other words, once the position where the listener wants to place the right virtual speaker (DR) and the distance between the listener's ears are known, the second delay time (T2) is set, and this remains constant even when the signal frequency changes.

[0055] More precisely, the second delay time (T2) is equal to the time it takes for the sound signal emitted from the right speaker to cover the right-to-left movement (SR) shown in Figure 2.

Number

[0056] Assuming that the sound has an average speed (SS) of 343 m / s, the second delay time (T2) is a function of the distance between the listener's ears and the tilt angle (B) of the right speaker with respect to the listener. Thus, the second delay time (T1) is between 0 when B = 0° and D / 343 when B = 90°. Considering that D is generally between 15 cm and 18 cm, T2 can be in the range of 0 μs to 525 μs.

[0057] The delay blocks (D1, D2) are of an adjustable type. Therefore, the user can adjust the delay times (T1, T2) according to the distance between the ears and according to the inclination of the virtual speaker to be simulated.

[0058] The first output (U1) and the second output (U2) of the system (100) are connected to the respective inputs (21L, 21R) of the headphones (2). The inputs (21L, 21R) of the headphones are connected to the left earpiece (20L) and the right earpiece (20R), respectively. Thus, the left earpiece (20L) emits a sound obtained by adding the second delayed acoustic signal (ED2) to the first acoustic signal (E1), while the right earpiece (20R) emits a sound obtained by adding the first delayed acoustic signal (ED1) to the second acoustic signal (E2).

[0059] The sound from the left earpiece (20L) and the sound from the right earpiece (20R) reproduce the sound that the listener would perceive from the physical left earpiece and the physical right earpiece, at positions set by the user by varying the delay times (T1, T2) of the first delay block and the second delay block (D1, D2).

[0060] Returning to FIG. 2, the delayed acoustic signal (EDL) reaching the right ear from the left speaker may be insufficient and attenuated at high frequencies (above 2 kHz) because the left-to-right movement (SL) is obstructed by the listener's facial features, causing attenuation of the acoustic signal. Also, the delayed acoustic signal (EDR) reaching the left ear from the right speaker (DR) may be insufficient and attenuated at high frequencies because the right-to-left movement (SR) is obstructed by the listener's facial features.

[0061] Referring to FIG. 4, to address such drawbacks, the system (100) advantageously comprises a first low-pass filter (F1) and a second low-pass filter (F2) respectively disposed at the outputs of the first delay block (D1) and the second delay block (D2).

[0062] The first low-pass filter (F1) and the second low-pass filter (F2) respectively block the high frequencies of the first delayed acoustic signal (ED1) and the second delayed acoustic signal (ED2). This is to mimic the high-frequency attenuation resulting from the fact that the first delayed acoustic signal (ED1) is obstructed by the listener's facial features during the left-to-right movement (SL), and the second delayed acoustic signal (ED2) is obstructed by the listener's facial features during the right-to-left cover (SR).

[0063] The cut-off frequencies of the first low-pass filter (F1) and the second low-pass filter (F2) are adjustable and are adjusted according to the configurations of the left-to-right movement (SL) and the right-to-left movement (SR). In any case, the cut-off frequencies of the first low-pass filter (F1) and the second low-pass filter (F2) can be set in the range of 900 Hz to 20,000 Hz, which corresponds to the null effect of audible frequencies.

[0064] In addition to, or replacing, the first low-pass filter (F1) and the second low-pass filter (F2), the system (100) comprises a first attenuator (AT1) and a second attenuator (AT2). If the low-pass filters (F1, F2) are not provided, the first attenuator (AT1) and the second attenuator (AT2) are respectively arranged at the outputs of the first delay block (D1) and the second delay block (D2). If the low-pass filters (F1, F2) are provided, the first attenuator (AT1) and the second attenuator (AT2) are respectively arranged at the outputs of the first low-pass filter (F1) and the second low-pass filter (F2).

[0065] The attenuators (AT1, AT2) attenuate their respective delayed acoustic signals (ED1, ED2) regardless of frequency. The attenuators (AT1, AT2) mimic the attenuation of the delayed acoustic signals due to facial artifact interference regardless of the frequency of the signal.

[0066] Referring to FIG. 5, the system (100) comprises a first active filter (FA1) arranged at the output of the first addition block (S1) and a second active filter (FA2) arranged at the output of the second addition block (S2). Accordingly, the first active filter (FA1) acts on the first acoustic signal (E1) and the second delayed acoustic signal (ED2). The second active filter (FA1) acts on the second acoustic signal (E2) and the first delayed acoustic signal (ED1).

[0067] The function of the active filters (FA1, FA2) is to enhance low-frequency sounds, for example when there is a lack of acoustic signals at low frequencies due to problems during the recording process.

[0068] The active filters (FA1, FA2) can be provided in the system (100) in the presence or absence of the low-pass filters (F1, F2) and / or the attenuators (AT2, AT3).

[0069] Advantageously, each attenuator (AT3, AT4) is arranged at the output of the active filter (FA1, FA2) and attenuates the signal from the active filter (volume attenuator) regardless of frequency.

[0070] As shown in FIG. 6a, an example will be described where there is a sound distribution (S1) from a 1960s music work at the input of the active filters (FA1, FA2), with a typical deficiency at frequencies below 100 Hz.

[0071] Each active filter (FA1, FA2) has an adjustable knob for adjusting the frequency range for enhancing the sound signal. As shown in FIG. 6b, the active filter has a frequency response according to the position of the knob, represented by a family of board diagrams.

[0072] Each board diagram is a first-order (6 dB / octave) transfer function (G1,... G12), characterized by a fixed low frequency, for example, in the range of 15 Hz to 25 Hz, preferably a frequency of 20 Hz for the pole, and a variable frequency according to the position of the knob, for example, in the range of 15 Hz to 1.2 KHz for the zero. The diagram in FIG. 6b shows the 12 transfer functions (G1,... G12) obtained at the 12 knob positions.

[0073] In such a case, for the purpose of illustration, to compensate for the sound deficiency at frequencies below 100 Hz in the sound distribution (S1) of FIG. 6a, the knob is set to 100 Hz to enhance the low-frequency sound below 100 Hz. In such a case, a filter with the transfer function (G5) shown in FIG. 6c is generated, but at frequencies above 100 Hz, the transfer function (G5) is equal to 0 dB.

[0074] From the transfer functions of the diagram in FIG. 6b, the transfer function (G5) shown in FIG. 6c having a zero at 100 HZ is selected. In other words, a transfer function having a zero at 100 Hz, a pole at 20 Hz, and a constant gain of 5x (about 14 dB) at frequencies below 20 Hz is selected.

[0075] In such a case, when the sound distribution (S1) in FIG. 6a is multiplied by the transfer function (G5) of the filter shown in FIG. 6c, the sound distribution (S2) shown in FIG. 6d is obtained, which becomes equal to zero dB at frequencies above 20 Hz. Therefore, the resulting sound distribution (S2) causes sound enhancement only at low frequencies above 20 Hz and below 100 Hz.

[0076] In such a case, as seen in the sound distribution (S2) in FIG. 6d, sound compensation is achieved only at low frequencies without changing the sound tone.

[0077] Regarding this embodiment of the present invention, many equivalent variations and modifications can be made within the scope possible for those skilled in the art, and these are within the scope of the present invention represented by the appended claims.

Claims

1. A system (100) for processing a stereo audio signal so that a listener (U) wearing a sound headphone (2) perceives that sound is heard from a left virtual speaker (DL) arranged on the left side of the listener and from a right virtual speaker (DR) arranged on the right side of the listener, wherein the system (100) comprises: a first input (I1) and a second input (I2), which are suitable for connecting to a sound source (1) capable of transmitting a stereo electroacoustic signal including a first acoustic signal (E1) suitable for being transmitted to the first input (I1) and a second acoustic signal (E2) suitable for being transmitted to the second input (I2), the first input (I1) and the second input (I2); a first output (U1) and a second output (U2), which are suitable for connecting to a sound headphone (2) provided with a left earpiece (20L) and a right earpiece (20R), the first output (U1) and the second output (U2); a first addition block (S1), which has an input connected to the first input (I1) and an output connected to the first output (U1), the first addition block (S1); a second addition block (S2), which has an input connected to the second input (I2) and an output connected to the second output (U2), the second addition block (S2); a first delay block (D1), which is connected to the first input (I1) and the second addition block (S2), and imparts a first delay time (T1) to the first acoustic signal (E1) to generate a first delayed acoustic signal (ED1) at the output from the first delay block (D1), and the first delayed acoustic signal (ED1) is added to the second acoustic signal (E2) transmitted to the second input (I2) in the second addition block (S2), the first delay block (D1); and a second delay block (D2), which is connected to the second input (I2) and the first addition block (S1), and imparts a second delay time (T2) to the second acoustic signal (E2) to generate a second delayed acoustic signal (ED2) at the output from the second delay block (D2), and the second delayed acoustic signal (ED2) is added to the first acoustic signal (E1) transmitted to the first input (I1) in the first addition block (S1), the second delay block (D2); and comprises. The first delay block (D1) is connected to the first input (I1), applies a first delay time (T1) to the first acoustic signal (E1) transmitted to the first input (I1), and is appropriately configured to generate the first delay time (T1) regardless of the signal frequency. The first delay time (T1) is maintained constant even when the frequency of the signal changes, and is adjustable according to the distance (D) between the position of the left virtual speaker (DL) with respect to the listener (U) and the ear of the listener (U), and the second delay block (D2) is connected to the second input (I2), applies a second delay time (T2) to the second acoustic signal (E2) transmitted to the second input (I2), and is appropriately configured to generate the second delay time (T2) regardless of the signal frequency. The system (100) is adjustable according to the distance between the position of the right virtual speaker (DR) with respect to the listener (U) and the ear (U) of the listener (U). **Claim 2** The position of the left virtual speaker (DL) with respect to the listener (U) is defined by a first inclination angle (A) of the left virtual speaker (DL) with respect to the listener (U), and the position of the right virtual speaker (DR) with respect to the listener (U) is defined by a second inclination angle (B) of the right virtual speaker (DR) with respect to the listener (U). The listener (U) has a left ear (L) and a right ear (R). Considering a segment (LR) connecting the left ear and the right ear, the segment (LR) has a center (O). A first straight line (r1) connects the center of the left virtual speaker (DL) to the center (O) of the segment (LR), a second straight line (r2) connects the center of the right virtual speaker (DR) to the center (O) of the segment (LR), and a central straight line (r) passes through the center (O) of the segment (LR) and is orthogonal to the segment (LR). The system (100) according to claim 1, wherein the first inclination angle (A) between the central straight line (r) and the first straight line (r1) defines the inclination of the left virtual speaker (DL), and the second inclination angle (B) between the central straight line (r) and the second straight line (r2) defines the inclination of the right virtual speaker (DR). **Claim 3** The first delay time (T1) of the first delay block (D1) is given by the following formula 【Number 1】 (where D = the distance between the ears of the listener, A = the first inclination angle of the left virtual speaker (DL) with respect to the listener, SS = the average speed of sound in air) The second delay time (T2) of the second delay block (D2) is given by the following formula: 【Number 2】 (where D = the distance between the ears of the listener, B = the second inclination angle of the right virtual speaker (DR) with respect to the listener, SS = the average speed of sound in air) The system (100) according to claim 2.

4. A first low-pass filter (F1) and a second low-pass filter (F2), the first low-pass filter (F1) and the second low-pass filter being respectively arranged at the outputs of the first delay block (D1) and the second delay block (D2), The system (100) according to claim 1, wherein the first low-pass filter (F1) and the second low-pass filter (F2) are configured to block the high frequencies of the first delayed acoustic signal (ED1) and the second delayed acoustic signal (ED2) respectively.

5. The first low-pass filter (F1) and the second low-pass filter (F2) are cut-off frequency adjustable filters, and the cut-off frequencies of the first low-pass filter (F1) and the second low-pass filter (F2) can be adjusted in the range of 900 Hz to 20,000 Hz so as to block high frequencies. The system (100) according to claim 4.

6. A first attenuator (AT1) and a second attenuator (AT2), the first attenuator (AT1) and the second attenuator being respectively arranged at the outputs of the first delay block (D1) and the second delay block (D2), The system (100) according to claim 1, wherein the first attenuator (AT1) and the second attenuator (AT2) are configured to attenuate the first delayed acoustic signal (ED1) and the second delayed acoustic signal (ED2) respectively regardless of frequency.

7. A first active filter (FA1) arranged at the output of the first addition block (S1) and a second active filter (FA2) arranged at the output of the second addition block (S2). The first active filter (FA1) acts on the first acoustic signal (E1) and the second delayed acoustic signal (ED2), and the second active filter (FA2) acts on the second acoustic signal (E2) and the first delayed acoustic signal (ED1). The system (100) according to claim 1, wherein each of the active filters (FA1, FA2) is configured to enhance low-frequency sounds. **Claim 8** Each of the active filters (FA1, FA2) has an adjustable knob for selecting a frequency range for enhancing the sound signal. Each of the active filters has a first-order transfer function (G1 ,... G12) represented by a family of board diagrams, where each board diagram is characterized by a fixed low-frequency pole and a variable-frequency zero, and has a frequency response represented by the family of board diagrams, the system (100) according to claim 7. **Claim 9** The poles of each of the transfer functions (G1,... G12) of each of the active filters are at frequencies in the range of 15 Hz to 25 Hz, and the zeros of each of the transfer functions (G1,... G12) of each of the active filters are at frequencies in the range of 15 Hz to 1.2 kHz, the system (100) according to claim 8.

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