Stereophonic 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 system for processing stereophonic audio signals suitable for transmission to audio headphones, such that the signal heard by a user wearing said headphones resembles as closely as possible an imaginary hearing obtained with two or more loudspeakers in an anechoic listening room.

[0002] As is known to all users, professional users, audiophiles, sound operators etc. who are accustomed to listening to acoustic signals (such as musical pieces) using headphones, the perception of a musical piece using headphones differs from the perception obtained without headphones and from two or more loudspeakers in a listening room.

[0003] In the case of basic stereophony with two loudspeakers, the main reason for the difference between the signal perceived with headphones and that perceived from two loudspeakers is that the sound signal from the left loudspeaker arrives at the left ear first and then the right ear after a time interval, which is due to the propagation of sound in the space between the left and right ears.

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

[0005] Thus, with two speakers and one listener, a direct acoustic signal from the left speaker and a delayed acoustic signal from the right speaker reach the left ear, a direct acoustic signal from the right speaker and a delayed acoustic signal from the left speaker reach the right ear, whereas with headphones, the sounds reproduced at the right and left earpieces are perceived at the right and left ears without any delay.

[0006] Such phenomena were analyzed as early as the 1960s (Robert Larson and John Eagle, Audio Magazine, November 1962). A clearer electronic system for improving headphone listening was devised by engineer Siegfried Linkwitz and published in "Improving Headphone Listening" in Audio Magazine, December 1971, and is still on the market today. The system has been improved over the years, but is based on empirical principles and intuition.

[0007] A document by Siegfried Linkwitz describes a device suitable for placement between a sound source and headphones. Such a device includes two delay blocks, which delay the sound signal going to the right and left earpiece of the headphones. The delay blocks are RC resonators, i.e. resonant circuits made of capacitance and resistance that generate a delay that varies with the change in the frequency of the signal. This results in different delays for low and high frequency signals, which results in poor fidelity sound reproduction.

[0008] In "Improvement of the sound image in stereo headphones", Thomas MV describes a device suitable for placement between a sound source and headphones. Such a device comprises two delay blocks, which 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 keep the phase of the signal constant. To keep the phase of the signal constant, it is necessary to vary the delay as the frequency of the signal changes. Figure 6 of said document shows that the delay varies with frequency and decreases for frequencies above 4 kHz. As a result, such a device does not guarantee high fidelity sound reproduction at high frequencies above 4 KHz.

[0009] In "Externalization of Headphone Signals and Improvement of Front Perception" by Weinrich, Søren and Gret, a device suitable for placement between a sound source and headphones is described. Such a device comprises two delay blocks, which delay the sound signal going to the right earpiece of the headphones and the sound signal going to the left earpiece. The document does not disclose how the delay of each delay block is calculated, and certainly does not disclose or suggest any relationship between the delay and the position of the loudspeaker with respect to the user wearing the headphones. The document states that the delay of the blocks is 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 loudspeaker with respect to the listener. In fact, assuming an inclination angle of the loudspeaker with respect to the listener of 90°, the distance between the earpieces must be about 68 cm to give a delay of 2000 μs, which is clearly illogical, since the distance between the two earpieces is typically in the range of 15 cm to 18 cm.

[0010] There are other phenomena that can cause differences between listening with headphones and listening without them, like internal reflections in the listening room, high-power signals entering the nasal passages, and low-frequency signals that vibrate the stomach and bones.

[0011] Musical works from the 1960s, when high-fidelity techniques in recording studios were not advanced, often suffer from a lack of low-frequency sounds. Figure 1a shows the distribution of possible sounds (S1) from musical works from the 1960s, with a typical lack of frequencies below 100 Hz.

[0012] To overcome these shortcomings, a type of tone control known as a "Baxandal" is found in the majority of audio amplifiers and allows the sound to be boosted or attenuated in two or more frequency bands of the audio signal.

[0013] A Baxandal tone control has an adjustable knob for selecting a frequency band for boosting. Depending on the knob position, the Baxandal tone control has a response of boosting low frequencies of the type shown in FIG. 1b. FIG. 1b shows a family of 12 Bode plots as a function of knob position. Specifically, the knob has 12 positions. The Bode plots are first order (6 dB / octave) transfer functions (G1,...G12) characterized by a variable frequency pole and a fixed zero at an average frequency, e.g., 1 kHz.

[0014] Moving the knob to boost frequencies below 100 Hz selects the transfer function G7 shown in Figure 1c, which has a pole at 100 Hz, from the transfer function of the diagram in Figure 1b. In other words, the transfer function of knob position 7 is selected, which has a pole at 100 Hz and a zero at 1 KHz, resulting in a gain of 20 dB. Thus, a filter with transfer function G7 is generated.

[0015] However, if we multiply the sound distribution (S1) in Figure 1a by the G7 transfer function of the knob-selected filter (thus adding up the dB values ​​of the two diagrams), we get the distribution (S2) shown in Figure 1d, which has a zero at the origin, a double pole at 100Hz and a zero at 1KHz. The resulting distribution (S2) therefore boosts the sound even at low and mid frequencies (100Hz-1KHz) and changes the tone of the final sound.

[0016] The object of the present invention is to overcome the drawbacks of the prior art by providing a system for processing stereo audio signals suitable for reproducing audio signals with high fidelity and high realism similar to the perception of a user listening through loudspeakers.

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

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

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

[0020] These objects are achieved according to the invention with the features of the attached independent claims.

[0021] Advantageous developments of the invention emerge from the dependent claims. [Brief description of the drawings]

[0022] Further features of the invention will emerge more clearly from the following detailed description with reference to purely exemplary and non-limiting embodiments illustrated in the attached drawings, in which: [Figure 1a] Figure 1a shows the sound distribution of frequencies from a musical piece from the 1960s, with a typical deficiency of frequencies below 100 Hz. [Figure 1b] FIG. 1b shows a family of Bode plots as a function of knob position for a Baxter-type tone control according to the prior art. [Figure 1c] Figure 1c shows the transfer function when the Baxter-type tone control knob is set to 100 Hz. [Figure 1d] FIG. 1d shows the sound distribution resulting from multiplying the sound distribution of FIG. 1a by the transfer function of FIG. 1c. [Diagram 2] FIG. 2 is a schematic diagram showing two speakers each emitting an acoustic signal towards a listener. [Diagram 3] FIG. 3 is a block diagram of a basic version of a system for processing stereophonic signals according to the invention. [Figure 4]FIG. 4 is a block diagram of an improved stereo audio signal processing system according to the present invention, in which a low pass filter and an attenuator are added. [Diagram 5] FIG. 5 is a block diagram of a further improved stereo audio signal processing system according to the present invention, in which active filters and attenuators are added. [Figure 6a] Similar to Figure 1a, Figure 6a shows the sound distribution of frequencies from a musical piece from the 1960s, with the typical deficiency of frequencies below 100 Hz. [Figure 6b] FIG. 6b shows a family of Bode plots as a function of knob position for an active filter in accordance with the present invention. [Figure 6c] Figure 6c shows the transfer function when the active filter knob is set to 100 Hz. [Figure 6d] FIG. 6d shows the sound distribution resulting from multiplying the sound distribution of FIG. 6a by the transfer function of FIG. 6c.

[0023] Figure 2 shows a typical listening situation using speakers or loudspeakers in an anechoic chamber with a left speaker (DL) placed to the left of the listener (U) and a right speaker (DR) placed to the right of the listener (U).

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

[0025] A first angle (A) between the center line (r) and the first line (r1) defines the tilt of the left speaker (DL). A second angle (B) between the center line (r) and the second line (r2) defines the tilt 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), which typically lies in the range of 140 mm to 180 mm.

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

[0028] In fact, once the sound from the left speaker (DL) reaches the left ear, it must cover a left-to-right distance (SL) before it can reach the right ear. This left-to-right distance (SL) depends on the position of the left speaker (DL) and is given by: SL=D*sin(A)

[0029] In this case, considering that sound travels through 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 sound is applied with a left to right delay (TL) given by: TL=SL / SS

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

[0032] For a sound from the right speaker (DR) to reach the right ear and then the left ear, the sound must cover a right-to-left shift (SR), which is given by the following equation: SR = D * sin(B)

[0033] To cover right-to-left movements (SR), the sounds are applied with a right-to-left delay (TR) given by: TR=SR / SS

[0034] The sound emitted from the left speaker and the sound emitted from the right speaker are identifiable as audio signals.

[0035] EL denotes the direct acoustic signal emitted by the left loudspeaker (DL), which reaches the left ear (L) without delay.

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

[0037] ER denotes the direct acoustic signal emanating from the right loudspeaker (DR), which reaches the right ear (R) without delay.

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

[0039] Thus, with two speakers and one listener, the left ear (L) receives a direct acoustic signal (EL) from the left speaker and a delayed acoustic signal (EDR) from the right speaker, and conversely, the right ear (R) receives a direct acoustic signal (ER) from the right speaker and a delayed acoustic signal (EDL) from the left speaker.

[0040] The object of the invention is to mimic the sound conditions as described above in the right and left earpieces of an acoustic headphone.

[0041] Referring to FIG. 3, there is shown a system, generally designated 100, in accordance with the present invention.

[0042] This system (100) is used to process stereophonic audio signals so that a listener (U) wearing acoustic headphones (2) perceives sounds as coming from a left virtual speaker (DL) located to the left of the listener and from a right virtual speaker (DR) located to the right 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 located to the left of the listener. Obviously, such a speaker does not exist in the system, so it is called the left virtual speaker (DL) and is shown in dashed lines 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 located to the right of the listener. Obviously, such a speaker does not exist in the system, so it is called the right virtual speaker (DL) and is shown in dashed lines in FIG. 3.

[0043] The system (100) comprises: a first input (I1) and a second input (I2) suitable for connection to a sound source (1) suitable for transmitting a stereo electric audio signal; a first output (U1) and a second output (U2) suitable for connection to an acoustic headphone (2) having a left earpiece (20L) and a right earpiece (20R); Equipped with.

[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, etc.

[0045] The stereo signal emanating from the sound source comprises a first acoustic signal (E1) sent to a first input (11) and a second acoustic signal (E2) sent to a second input (I2). In case of a stereo transmission with two speakers, the first acoustic signal (E1) is the signal sent to the left speaker and the second acoustic signal (E2) is the signal sent to the right speaker.

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

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

[0048] The first input (11) is connected to a first delay block (D1) which is connected to a second summing block (S2). The first delay block (D1) applies a first delay (T1) to a first delayed signal (E1) and outputs a first delayed acoustic signal (ED1) which is added to a second acoustic signal (E2). The second summing block (S2) has an output connected to a 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) which is connected to the first summing block (S1). The second delay block (D2) applies a second delay (T2) to the second acoustic signal (E2) and outputs a second delayed acoustic signal (ED2) which is added to the first acoustic signal (E1). The first summing block (S1) has an output which is 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 suitably 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 (L,R) are known, the first delay time (T1) is set and remains constant even if the frequency of the signal changes.

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

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[0053] Assuming that the 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 tilt angle (A) of the left speaker relative 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 typically between 15cm and 18cm, T1 can be in the range of 0μs to 525μs.

[0054] The second delay block (D2) is suitably 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 loudspeaker (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 loudspeaker (DR) and the distance between the listener's ears is known, the second delay time (T2) is set and 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 by the right speaker to cover the right-to-left movement (SR) shown in FIG.

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[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 relative 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 typically between 15cm and 18cm, T2 can range from 0μs to 525μs.

[0057] The delay blocks (D1, D2) are of adjustable type, so the user can adjust the delay times (T1, T2) according to the distance between the ears and according to the tilt of the virtual speakers to be emulated.

[0058] The first output (U1) and the second output (U2) of the system (100) are connected to 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 given by the first acoustic signal (E1) plus the second delayed acoustic signal (ED2), whereas the right earpiece (20R) emits a sound given by the second acoustic signal (E2) plus the first delayed acoustic signal (ED1).

[0059] The sound from the left earpiece (20L) and the sound from the right earpiece (20R) are reproduced as a listener would perceive it from the physical left earpiece and the physical right earpiece, located in the positions set by the user, by varying the delay times (T1, T2) of the first and second delay blocks (D1, D2).

[0060] Returning to Fig. 2, it should be considered that the delayed acoustic signal (EDL) arriving at the right ear from the left loudspeaker may be deficient and attenuated at high frequencies (above 2KHz) because its movement from left to right (SL) is blocked by the listener's facial features, causing attenuation of said acoustic signal, and the delayed acoustic signal (EDR) arriving at the left ear from the right loudspeaker (DR) may be deficient and attenuated at high frequencies because its movement from right to left (SR) is blocked by the listener's facial features.

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

[0062] The first low pass filter (F1) and the second low pass filter (F2) respectively block high frequencies of the first delayed acoustic signal (ED1) and the second delayed acoustic signal (ED2) in order to mimic the attenuation of high frequencies due to the fact that the first delayed acoustic signal (ED1) is blocked by the listener's facial features in left to right translation (SL) and the second delayed acoustic signal (ED2) is blocked by the listener's facial features in right to left coverage (SR).

[0063] The first low pass filter (F1) and the second low pass filter (F2) have adjustable cut-off frequencies, which are adjusted according to the configurations of left-to-right shift (SL) and right-to-left shift (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 900Hz to 20,000Hz, which corresponds to the nulling effect of audio frequencies.

[0064] In addition to or instead of the first low-pass filter (F1) and the second low-pass filter (F2), the system (100) includes 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 arranged at the output of the first delay block (D1) and the second delay block (D2), respectively. If the low-pass filters (F1, F2) are provided, the first attenuator (AT1) and the second attenuator (AT2) are arranged at the output of the first low-pass filter (F1) and the second low-pass filter (F2), respectively.

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

[0066] 5, the system (100) comprises a first active filter (FA1) arranged at the output of the first summing block (S1) and a second active filter (FA2) arranged at the output of the second summing block (S2). The first active filter (FA1) thus 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 boost low frequency sounds when there is a lack of acoustic signal at these frequencies, for example due to problems in the recording process.

[0068] The active filters (FA1, FA2) may be provided in the system (100) with or without the presence of low pass filters (F1, F2) and / or attenuators (AT2, AT3).

[0069] Advantageously, a respective attenuator (AT3, AT4) is placed at the output of an active filter (FA1, FA2) to attenuate the signal from the active filter (volume attenuator) regardless of frequency.

[0070] Let us consider the case where at the input of the active filters (FA1, FA2) there is a sound distribution (S1) from a musical piece of the 1960s, with a typical deficiency in frequencies below 100 Hz, as shown in FIG. 6a.

[0071] Each active filter (FA1, FA2) has an adjustable knob to adjust the frequency range in which it enhances the sound signal. The active filters have a frequency response depending on the knob position, which is represented by a family of Bode plots, as shown in Figure 6b.

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

[0073] In such a case, for illustrative purposes, to compensate for the lack of sound at frequencies below 100 Hz in the sound distribution (S1) of Figure 6a, the knob is set to 100 Hz to enhance the low frequency sounds below 100 Hz. In such a case, a filter is produced having a transfer function (G5) as shown in Figure 6c, but at frequencies above 100 Hz, said transfer function (G5) is equal to zero dB.

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

[0075] In such a case, multiplying the sound distribution (S1) of Fig. 6a by the transfer function (G5) of the filter shown in Fig. 6c gives the sound distribution (S2) shown in Fig. 6d, which is equal to zero dB for frequencies above 20 Hz. Thus, the resulting sound distribution (S2) produces sound enhancement only at low frequencies above 20 Hz and below 100 Hz.

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

[0077] Numerous equivalent variations and modifications within the scope of those skilled in the art may be made to the present embodiment of the invention, which remain within the scope of the invention as expressed by the appended claims.

Claims

1. A system (100) for processing stereo audio signals such that a listener (U) wearing audio headphones (2) perceives sounds as coming from a left virtual speaker (DL) located to the left of the listener and from a right virtual speaker (DL) located to the right of the listener, comprising: The system (100) comprises: a first input (I1) and a second input (I2) suitable for connection to a sound source (1) capable of transmitting a stereo electric sound signal comprising a first sound signal (E1) suitable for transmission to said first input (I1) and a second sound signal (E2) suitable for transmission to said second input (I2); a first output (U1) and a second output (U2) suitable for connection to an acoustic headphone (2) comprising a left earpiece (20L) and a right earpiece (20R); a first summing block (S1) having an input connected to said first input (I1) and an output connected to said first output (U1); a second summing block (S2) having an input connected to said second input (I2) and an output connected to said second output (U2); a first delay block (D1) connected to the first input (I1) and to the second summing block (S2) for applying a first delay time (T1) to the first acoustic signal (E1) to produce a first delayed acoustic signal (ED1) at an output from the first delay block (D1), the first delayed acoustic signal (ED1) being added to the second acoustic signal (E2) in the second summing block (S2); and a second delay block (D2) connected to the second input (I2) and to the first summing block (S1), for applying a second delay time (T2) to the second acoustic signal to produce a second delayed acoustic signal (ED2) at an output from the second delay block (D2), the second delayed acoustic signal (ED2) being added to the first acoustic signal (E1) in the first summing block (S1); Equipped with the first delay block (D1) is suitably configured to generate a first delay time (T1) independent of the signal frequency and adjustable according to the distance (D) between the position of the left virtual speaker (DL) relative to the listener (U) and the ear of the listener (U); and The second delay block (D2) is suitably configured to generate a second delay time (T1) independent of the signal frequency and adjustable depending on the distance between the position of the right virtual speaker (DR) relative to the listener (U) and the ear (U) of the listener (U), in the system (100).

2. The position of the left virtual speaker (DL) relative to the listener (U) is defined by a first tilt angle (A) of the left virtual speaker (DL) relative to the listener (U), and the position of the right virtual speaker (DR) relative to the listener (U) is defined by a second tilt angle (B) of the right virtual speaker (DR) relative to the listener (U); The listener (U) has a left ear (L) and a right ear (R). Consider a segment (LR) connecting the left ear and the right ear. The segment (LR) has a center (O), a first line (r1) connects the center of the left virtual speaker (DL) to the center (O) of the segment (LR), a second line (r2) connects the center of the right virtual speaker (DR) to the center (O) of the segment (LR), and a central line (r) passes through the center (O) of the segment (LR) and is perpendicular to the segment (LR); 2. The system (100) of claim 1, wherein the first angle (A) between the center line (r) and the first line (r1) defines a tilt of the left virtual speaker (DL) and the second angle (B) between the center line (r) and the second line (r2) defines a tilt of the right virtual speaker (DR).

3. The first delay time (T1) of the first delay block (D1) is given by the following formula: [0010] (where D=the distance between the listener's ears, A=the first tilt angle of the left virtual speaker (DL) relative to the listener, and 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: [0025] (Where D = the distance between the listener's ears, B = the second tilt angle of the right virtual speaker (D) relative to the listener, and SS = the average speed of sound in air.) The system (100) of claim 2.

4. a first low pass filter (F1) and a second low pass filter (F2) arranged at the outputs of the first delay block (D1) and the second delay block (D2), respectively; The system (100) of any one of claims 1 to 3, wherein the first low-pass filter (F1) and the second low-pass filter (F2) are configured to block high frequencies of the first delayed acoustic signal (ED1) and the second delayed acoustic signal (ED2), respectively.

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

6. a first attenuator (AT1) and a second attenuator (AT2) arranged at the outputs of the first delay block (D1) and the second delay block (D2), respectively; 6. The system (100) of 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 summing block (S1) and a second active filter (FA2) arranged at the output of the second summing 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 (FA1) acts on the second acoustic signal (E2) and the first delayed acoustic signal (ED1); The system (100) according to any one of claims 1 to 6, wherein each active filter (FA1, FA2) is configured to enhance low frequency sounds.

8. Each of the active filters (FA1, FA2) has an adjustable knob for selecting a frequency range in which to enhance the sound signal; Each of the active filters has a first order transfer function (G1) whose respective Bode plots are characterized by a fixed low frequency pole and a variable frequency zero as a function of the knob position.

8. The system (100) of claim 7, having a frequency response represented by a family of Bode plots where the Bode plots are G1, G2, G3, G4, G5, G6, G7, G8, G9, G10, G11, G12.

9. 10. The system (100) of claim 9, wherein 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, preferably 20 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.

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