Audio spatialization and environment simulation

Abstract

Methods are disclosed for improving sound localization of the human ear. In some embodiments, the method may include creating virtual movement of a plurality of localized sources by applying a periodic function to one or more location parameters of a head related transfer function (hrtf).

Description

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a divisional of U.S. patent application Ser. No. 12/582,449, entitled “AUDIO SPATIALIZATION AND ENVIRONMENT SIMULATION”, filed on Oct. 20, 2009, now U.S. Pat. No. 8,520,873, which is incorporated by reference in its entirety as if fully disclosed herein.

This application claims priority to U.S. provisional patent application No. 61/106,872, filed Oct. 20, 2008, and entitled “Audio Spatialization and Environment Simulation”, the contents of which are incorporated herein by reference in their entirety.

This application is related to the following commonly owned patent applications, each of which are incorporated by reference as if set forth in full below:

U.S. Provisional Application No. 60/892,508, filed Mar. 1, 2007, entitled “Audio Spatialization and Environment Simulation”;

U.S. Utility application Ser. No. 12/041,191, filed Mar. 3, 2008, entitled “Audio Spatialization and Environment Simulation”; and

PCT Application PCT/US08/55669, filed Mar. 3, 2008, entitled “Audio Spatialization and Environment Simulation”.

SUMMARY

GenAudio's AstoundSound™ technology is a unique sound localization process that places a listener in the center of a virtual space of stationary and/or moving sound. Because of the psychoacoustic response of the human brain, the listener may perceive that these localized sounds emanate from arbitrary positions within space. The psychoacoustic effects from GenAudio's AstoundSound™ technology may be achieved through the application of digital signal processing (DSP) for head related transfer functions (HRTFs).

Generally speaking, HRTFs may model the shape and composition of a human being's head, shoulders, outer ear, torso, skin, and pinna. In some embodiments, two or more HRTFs (one for the left side of the head and one for the right side of the head) may modify an input sound signal so as to create the impression that sound emanates from a different (virtual) position in space. Using GenAudio's AstoundSound™ technology, a psychoacoustic effect may be realized from as few as two speakers.

In some embodiments this technology may be manifested through a software framework that implements the DSP HRTFs through a binaural filtering method such as splitting the audio signal into a left-ear and right-ear channel and applying a separate set of digital filters to each of the two channels. Furthermore, in some embodiments, the post filtering of localized audio output may be accomplished without using encoding/decoding or special playback equipment.

The AstoundSound™ technology may be realized through Model-View-Controller (MVC) software architecture. This type of architecture may enable the technology to be instantiated in many different forms. In some embodiments, applications of AstoundSound™ may have access to similar underlying processing code, via a set of common software interfaces. Further, the AstoundSound™ technology core may include Controllers and Models that may be used across multiple platforms (e.g., may operate on Macintosh, Windows and/or Linux). These Controllers and Models also may enable real-time DSP processing play-through of audio input signals.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates a model view controller for a potential system architecture.

FIG. 2A illustrates two virtual speakers in azimuth and elevation relative to a listener.

FIG. 2B illustrates one virtual speakers in azimuth and elevation relative to a listener.

FIG. 3 illustrates a process flow for an expander.

FIG. 4 illustrates a potential wiring diagram for the expander.

FIG. 5 illustrates a process flow for a plug-in.

FIG. 6 illustrates a potential wiring diagram for the plug-in.

FIG. 7 illustrates oscillating a virtual sound source in three dimensional space.

FIG. 8 illustrates a process flow for a plug-in.

FIG. 9 illustrates a potential wiring diagram.

FIG. 10 illustrates localization of source audio reflections.

FIG. 11 illustrates a process flow for audio localization.

FIG. 12 illustrates a biquad filter and equation.

DESCRIPTION

AstoundStereo™ Expander Application

In some embodiments, the AstoundStereo™ Expander application may be implemented as a stand-alone executable that may take as input normal stereo audio and process it such that the output has a significantly wider stereo image. Further, the center information from the input (e.g., vocals and/or center staged instruments) may be preserved. Thus, the listener may “hear” a wider stereo image because the underlying AstoundStereo™ DSP technology creates the psychoacoustic perception that virtual speakers emanating the audio have been placed at a predetermined angle of azimuth, elevation and distance relative to the listener's head. This virtual localization of the audio may appear to place the virtual speakers farther apart than the listener's physical speakers and/or headphones.

One embodiment of the Expander may be instantiated as an audio device driver for computers. As a result, the Expander application may be a globally executed audio processor capable of processing a substantial amount of the audio generated by and/or passing through the computer. For example, in some embodiments, the Expander application may process all 3^rd party applications producing or routing audio on the computer.

Another consequence of the Expander being instantiated as an audio device driver for computers is that the Expander may be present and active while a user is logged into his/her computer account. Thus, a substantial amount of audio may be routed to the Expander and processed in real-time without loading individual files for processing, which may be the case for 3^rd party applications such as iTunes and/or DVD Player.

Some of the features of the AstoundStereo™ Expander include:

Stereo Expanded Symmetric Virtual Speaker Localization (EL, AZ, DIST)

Stereo Expansion Intensity Adjustment

Active Bass™

Global Bypass

Selectable Output Devices

Process Flow

A software controller class, from the Products Controller library, may enable the process flow of the AstoundStereo™ Expander application. As mentioned previously, the controller class may be a common interface definition to the underlying DSP models and functionality. The controller class may define the DSP interactions that are appropriate for stereo expansion processing. FIG. 3 illustrates an exemplary DSP interaction titled “Digitally process audio for localization”, which may be appropriate for stereo expansion. The activity shown in FIG. 3 is depicted in greater detail in FIG. 11.

The controller may accept a two-channel stereo signal as input, where the signal may be separated into a left and right channel. Each channel then may be routed through the set of AstoundStereo linear DSP functions, as shown in FIG. 4, and localized to a particular point in space (e.g., the two virtual speaker positions).

The virtual speaker locations may be fixed by the view-based application to be at a particular azimuth, elevation and distance, relative to the listener (e.g., see Infinite Impulse Response Filters below), where one virtual speaker is located some distance away from the listener's left ear and the other some distance away from the listener's right ear. These positions may be combined with parameters for %-Center Bypass (described in greater detail below) for enhanced vocals and center stage instrument presence, parameters for low pass filtering and compensation (e.g., see Low Frequency Processing below) for enhanced low frequency response, and parameters for distance simulation (see e.g., distance simulation description in PCT Application PCT/US08/55669, filed Mar. 3, 2008, entitled “Audio Spatialization and Environment Simulation”).

Combining the positions with these parameters may give the listener the perception of a wider stereo field.

Notably, the virtual speaker locations may be non-symmetrical in some embodiments. Symmetric positioning may undesirably diminish the localization effect (e.g., due to signal cancellation), which is described in greater detail below with regard to Hemispherical Symmetry.

Because the AstoundStereo Expander is an application (rather than a plug-in), it may contain a global DSP bypass switch to circumvent the DSP processing and allow the listener to hear the audio signal in its original stereo form. Additionally, the Expander may include an integrated digital watermarking technology that may detect a unique and inaudible GenAudio digital watermark. Detection of this watermark may automatically cause the AstoundStereo Expander process to enable global bypass. A watermarked signal may indicate that the input signal has been altered to already contain AstoundSound™ functionality. Bypassing this type of signal may be done to avoid processing the input signal twice and diminishing or otherwise corrupting the localization effect.

In some embodiments, the AstoundStereo™ process may include a user definable stereo expansion intensity level. This adjustable parameter may combine all the parameters for low frequency processing, %-center bypass and localization gain. Furthermore, some embodiments may include predetermined minimum and maximum settings for the stereo expansion intensity level. This user definable adjustment may be a linear interpolation between the minimum and maximum values for all associated parameters.

The ActiveBass™ feature of the AstoundStereo™ technology may include a user selectable switch that may increase one or more of the low frequency parameters (described below in the Low Frequency Processing section) to a predetermined setting for a deeper, richer, and more present bass response from the listener's audio output device.

In some embodiments, the selectable output device feature may be a mechanism by which the listener can choose from among various output devices, such as, built-in computer speakers, headphones, external speakers via the computers line-out port, a USB/FireWire speaker/output device and/or any other installed port that can route audio to a speaker/output device.

AstoundStereo™ Expander Plug-in Application

Some embodiments may include an AstoundStereo™ Expander Plug-in that may be substantially similar the AstoundStereo™ Expander Executable. In some embodiments, the Expander Plug-in may differ from the Expander Executable in that it may be hosted by a 3^rd party executable. For example, the Expander Plug-in may reside within an audio playback executable such as Windows Media Player, iTunes, Real Player and/or WinAmp to name but a few. Notably, the Expander Plug-in may include substantially the same features and functionality as the Expander Executable.

Process Flow

While the Expander Plug-in may include substantially the same internal process flows as the Expander executable, the external flow may differ. For example, instead of the user or the system instantiating the Plug-in, this may be handled by the 3^rd party audio playback executable.

AstoundStereo™ Plug-in Application

The AstoundStereo™ Plug-in may be hosted by a 3^rd party executable (e.g. ProTools, Logic, Nuendo, Audacity, Garage Band, etc.) yet it may have some similarities to the AstoundStereo™ Expander. Similar to the Expander, it may create a wide stereo field, however, unlike the Expander it may be tailored for the professional sound engineer and may expose numerous DSP parameters and allow a wide range of tunable control of the parameters to be accessed via a 3D user interface. Also, unlike the Expander, some embodiments of the Plug-in may differ from the Expander by integrating a digital watermarking component that may encode a digital watermark into the final output audio signal. Watermarking in this fashion may enable GenAudio to uniquely identify a wide variety of audio processed with this technology. In some embodiments, the exposed parameters may include:

Localization Azimuth & Elevation

Independent Left & Right Localization Gain

Localization Distance & Distance Reverberation

Positional Vibrato in Azimuth & Elevation for increased perception of the localized audio output

Master Input & Output Gain

Center Bypass Spread & Gain

Center Band Pass Frequency & Bandwidth

Low Frequency Band Pass Frequency, Roll-off, Gain & ITD Compensation

4-Band HRTF Filter Equalization

Reflection Localization Azimuth & Elevation (discussed in further detail below in the Reverb Localization section)

Reflection Localization Amount, Room Size, Decay, Density & Damping

Process Flow

The Plug-in may be instantiated and destroyed by the 3^rd party host executable.

%-Center Bypass

The %-center bypass (referred to above in FIGS. 3 and 6) is a DSP element that allows, in some embodiments, at least a portion of the audio's center information (e.g. vocals or “center stage” instruments) to be left unprocessed. The amount of center information in a stereo audio input that may be allowed to bypass processing may vary between different embodiments.

By allowing certain stereo audio to be bypassed, center channel information may remain prominent, which is a more natural, true-to-life representation. Without this feature, center information may become lost or diminished and give an unnatural sound to the audio. During operation, before the actual localization processing takes place, the incoming audio signal may be split into a center signal and a stereo edge signal. In some embodiments, this process may include subtracting out the L+R mono sum from the left and right channels—i.e., M−S decoding. The center portion may be subsequently processed after the stereo edges have been processed. In this manner, Center Bypass may determine how much of the processed center signal is added back to the output.

Center Band Pass

The center band pass DSP element shown in FIG. 6 may enhance the results of the %-center bypass DSP element. The center signal may be processed with a variable band pass filter in order to emphasize the lead vocal or instrument (which are commonly present in the center channel of a recording). If only the entire center channel is attenuated, the vocals and lead instruments may be removed from the mix, creating a “Karaoke” effect, which is not desired for some applications. Applying a band pass filter may alleviate this problem by selectively removing frequencies that are less relevant for the lead vocal, and therefore, may widen the stereo image without losing the lead vocals.

Spatial Oscillator

The human brain may more accurately determine the location of a sound if there is relative movement between the sound source and human ear. For example, a listener may move their head from side to side to help determine a sound location when the sound source is stationary. The reverse is also true. Thus, the spatial oscillator DSP element may take a given localized sound source and vibrate and/or shake it in a localized space to provide additional spatialization to the listener. In other words, by vibrating and/or shaking both virtual speakers (localized sound sources) the listener can more easily detect the spatialization effect of the AstoundStereo™ process.

In some embodiments, the overall movement of the virtual speakers) may be very small, or nearly imperceptible. Even though the movement of the virtual speakers may be small, however, it may be enough for the brain to recognize and determine location. The spatial oscillation of a localized sound may be accomplished by applying a periodic function to the location parameters of the HRTF function. Such periodic functions may include, but are not limited to sinusoidal, square wave, and/or triangular to name but a few. Some embodiments may use a sine wave generator in conjunction with a frequency and depth variable to repeatedly adjust the azimuth of the localization point. In this manner, frequency is a multiplier that may indicate the speed of vibration, and depth is a multiplier that may indicate the absolute value of the distance traveled for the localization point. The update rate for this process may be on a per sample basis in some embodiments.

Hemispherical Symmetry

Since the listener's head is symmetric with regard to the sagittal plane of the body, this symmetry may be exploited to reduce the amount of stored filter coefficients by 1/a in some embodiments. Instead of storing filter coefficients for a given symmetric position to the left and right of the listener (such as at 90° and 270° azimuth) filter coefficients may be selectively stored for one side, and then reproduced for the reciprocal side by swapping both the position and the output channels. In other words, instead of processing the position at 270° azimuth, the filter corresponding to 90° azimuth may be used and then the left and right channels may be swapped to mirror the effect to the other side of the hemisphere.

AstoundSound™ Plug-in Application

The AstoundSound™ Plug-in for the professional sound engineer may have similarities to the AstoundStereo™ Plug-in. For example, it may be hosted by a 3^rd party executable and also may expose all DSP parameters for a wide range of tuning capability. The two may differ in that the AstoundSound Plug-in may take a mono signal as input and allow a full 4D (3-dimentional spatial localization with movement over time) control of a single sound source, via a 3D user interface. Unlike the other applications discussed in this document, the AstoundSound Plug-in may enable the use of a 3D input device for moving the virtual sound sources in 3D space (e.g., a “3D mouse”).

Furthermore, the AstoundSound Plug-in may integrate a watermarking component that encodes a digital watermark directly into the final output audio signal, enabling GenAudio to uniquely identify a wide variety of audio processed with this technology. Because some embodiments may implement this functionality as a plug-in, the host executable may instantiate multiple instances of the plug-in, which may allow multiple mono sound sources to be spatialized. In some embodiments, a consolidated user interface may show one or more localized positions of these independent instantiations of the AstoundSound Plug-in running within the host. In some embodiments, the exposed parameters may include:

Localization Azimuth & Elevation

Localization Distance & Distance Reverberation

Positional Vibrato in Azimuth & Elevation

Master Input & Output Gain

Low Frequency Band Pass Frequency, Roll-off, Gain & ITD Compensation

4-Band HRTF Filter Equalization

Reflection Localization Azimuth & Elevation (see section Reverb Localization for details)

Reflection Localization Amount, Room Size, Decay, Density & Damping

Process Flow

The plug-in this is instantiated and destroyed by the 3^rd party hosting executable.

Reverb Localization

In order to improve the spatialization effect, some embodiments may localize the reverberated (or reflected) signals by applying a different set of localization filters than the direct (“dry”) signal. We can therefore position the perceived origin of the direct signal's reflections out of the way of the direct signal itself. While the reflections can be localized anywhere (i.e. variable positioning), it has been determined that positioning them to the back of the listener results in higher clarity and better overall spatialization.

Common Technologies

Infinite Impulse Response Filters

Conventional AstoundSound™ DSP technology may define numerous (e.g., −7,000+) independent points on a notional unit sphere. For each of these points, two finite impulse response (FIR) filters were calculated, based on the right and left HRTFs for that point and the inverses of the right and left head-to-ear-canal transfer functions.

In some embodiments, the FIR filters may be supplanted by a set of Infinite Impulse Response (MR) filters. For example, a set of 64-coefficient NR filters may be created from the original 1,920-coefficient FIR HRTF filters using a least mean square error approximation. Unlike the block based processing necessary to do linear convolution in the frequency domain, MR filters may be convolved in the time domain without needing to perform a Fourier transform. This time domain convolution process may be used to calculate the localized result on a sample-by-sample basis. In some embodiments, the MR filters do not have an inherent latency, and therefore, they may be used for simulating both position updates and localizing sound waves without introducing a perceivable processing delay (latency). Furthermore, the reduction in the number of coefficients from 1,920 in the original FIR filters to 64 coefficients in the MR filters may reduce significantly the memory footprint and/or CPU cycles used to calculate the localized result. An Inter-aural Time Difference (ITD) may be added back into the signal by delaying the left and right signal according to the ITD measurements derived from the original FIR filters.

Because the HRTF measurements may be performed at regular intervals in space with a relatively fine resolution, spatial interpolation between neighboring filters may be minimized for position updates (i.e. when moving a sound source over time). In fact, some embodiments may accomplish this without any interpolation. That is, moving sound source directions may be simulated by loading the MR filters for the nearest measured direction. Position updates then may be smoothed across a small number of samples to avoid any zipper noise when switching between neighboring NR filters. A linearly interpolated delay line may be applied for ITD to both right and left channels allowing for sub-sample accuracy.

IIR filters are similar to FIR filters in that they also process samples by calculating a weighted sum of the past (and/or future) samples, where the weights may be determined by a set of coefficients. However, in the MR situation, this output may be fed back to the filter input thereby creating an asymptotically decaying impulse response that theoretically never decays to zero—hence the name “Infinite Impulse Response”. Feeding back the processed signal in this manner may “reprocess” the signal partially by running it through the filter multiple times, and therefore, increase the control or steepness of the filter for a given number of coefficients. A general diagram for an MR biquad structure as well as the formula for generating its output is shown below in FIG. 12:

Sample Rate Independence

Conventional FIR filters were sampled at a 44.1 kHz sample rate, and therefore due to Nyquist criterion, the FIR filters were capable of processing signals between 0 Hz and half the sampling rate (i.e., the Nyquist frequency). However, in today's audio production environments, higher sampling rates may be desired. In order to enable the AstoundSound™ filters to deal with higher sample rates without losing the high frequency content that comes with the higher sample rates, the frequencies above the Nyquist frequency of the original filters (22,050 Hz) may be bypassed. To accomplish this bypassing, the signal may be first split into low (<Nyquist) and high (>=Nyquist) frequency bands. The low frequency band then may be down-sampled to the sampling frequency of the conventional HRTF filters and subsequently processed by the localization algorithm at a 44.1 kHz sampling frequency. Meanwhile, the high frequency band may be retained for later processing. After the localization processing has been applied to the low frequency band, the resulting localized signal may be again up-sampled to the conventional sample rate and mixed with the high frequency band. In this manner, a bypass for the high frequencies may be created in the original signal that would not have survived sample rate conversion to 44.1 kHz.

Alternate embodiments may achieve the same effect by extending the sampling rate of the conventional FIR filters by re-designing them at a higher sample rate and/or converting them to an HR structure. However, this may imply two additional sample rate conversions that to be applied to the processed signal, and therefore, may represent a higher processing load when processing the more frequently encountered sample rates like 44.1 kHz. Because the 44.1 kHz sample rate has been well tested and is still a frequently encountered sample rate on today's consumer music reproduction systems, some embodiments may eliminate the extra bandwidth and only apply sample rate conversion in a more limited number of cases. Also, since a substantial portion of the AstoundSound™ DSP processing may be carried out at 44.1 kHz, fewer CPU instructions may be consumed per sample cycle.

Filter Equalization

“Filter equalization” generally refers to the process of attenuating certain frequency spectrum bands to reduce colorization that can be introduced in HRTF localization. Conventionally, for the numerous (e.g., −7,000+) independent filter points, an average magnitude response was calculated to determine the overall deviation of the filters from an idealized (flat) magnitude response process. This averaging process identified 4 distinct peaks in the frequency spectrum of the conventional filter set that deviated from a flat magnitude causing the filters to colorize the signal in potentially undesired ways. In order to define a localization/colorization tradeoff, some embodiments of the AstoundSound™ DSP implementation may add a 4-band equalizer at the 4 distinct frequencies, thereby attenuating the gain at these distinct points in frequency. Although 4 distinct frequencies have been discussed herein, it should be noted that any number of distinctive frequency equalization points are possible and a multi-band equalizer may be implemented, where each distinct frequency may be addressed by one or more bands of the equalizer.

Low Frequency Processing

Low Pass Filtering

In some embodiments, low frequencies may not need to be localized. Additionally, in some cases, localizing low frequencies may alter their presence and impact the final output audio. Thus, in some embodiments, the low frequencies present in the input signal may be bypassed. For example, the signal may be split in frequency allowing the low frequencies to pass through unaltered. It should be noted that the precise frequency threshold at which bypass begins (referred to herein as the “LP Frequency”) and/or the localization of the onset of the bypass in frequency (referred to herein as the “Q factor” or “rolloff”) may be variable.

ITD Compensation

When preparing the final mixing of the localized signal with the bypassed low frequency signal, prior to final output, the time delay introduced into the localized signal by the inter-aural time difference (ITD) may cause both signals to have different relative time delays. This time delay artifact may create a misalignment in phase for the low frequency content at the transition frequency when it is mixed with the localized signal. Thus, in some embodiments, delaying the low frequency signal by a predetermined amount using an ITD compensation parameter may compensate for the phase misalignment.

Phase Flip

In some cases, the phase misalignment between the localized signal and the bypassed low frequency signal may cause the low frequency signal to be attenuated to a point where it is almost cancelled out. Thus, in some embodiments, the phase of the signal may be flipped by reversing the polarity of the signal (which is equivalent to multiplying the signal by −1). Flipping the signal in this manner may change the attenuation into a boost, bringing back much of the original low frequency signal.

Low Pass Gain

In some embodiments, the low frequencies may have an adjustable output gain. This adjustment may allow for filtered low frequencies to have a more or less prominent presence in the final audio output.

Claims

1. A method for improving sound localization of a listener, the method comprising:

receiving a sound signal including a left-ear signal and a right-ear signal in a controller;

modifying the left-ear signal and the right-ear signal, the step of modifying comprising:

applying a plurality of infinite impulse response (IIR) filters to each of the left-ear signal and the right-ear signal, the filters being calculated based on each of a left head related transfer function (hrtf) and a right hrtf;

creating a plurality of virtual localized sources at a plurality of positions with distances away from the listener; and

vibrating the plurality of virtual localized sources by a spatial oscillator.

2. The method of claim 1, wherein each IIR filer is associated with a position at an angle of azimuth, elevation and distance relative to the listener.

3. The method of claim 1, further comprising embedding a watermark signal in the localized left-ear signal and the right-ear signal for indicating that the signals are altered.

4. The method of claim 1, the step of modifying comprising splitting each of the left-ear signal and the right-ear signal into a first frequency band and a second frequency band having lower frequencies than the first frequency band.

5. The method of claim 4, further comprising bypassing the left-ear and right-ear signal in the second frequency band.

6. The method of claim 4 further comprising splitting each of the left-ear signal and the right-ear signal in the first frequency band into a center signal and a stereo edge signal.

7. The method of claim 6 further comprising localizing the stereo edge signals of left-ear and right-ear signals in the first frequency band by convolving the plurality of filters in a time domain.

8. The method of claim 6 further comprising bypassing the center signals in the first frequency band.

9. The method of claim 4, further comprising adjusting output gain independently for each of the edge signal in the first frequency band, center signal in the first frequency band, and the signal in the second frequency band.

10. The method of claim 1, the step of outputting a localized left-ear signal and a localized right-ear signal comprising delaying the localized right-ear signal and the localized left-ear signal by a period of time that is an inter-aural time difference (ITD) and mixing the localized signals and bypassed signals in the first frequency band with respective bypassed signals in the second frequency band.

11. The method of claim 1, further comprising repeating the step of applying a periodic function through the step of outputting a localized left-ear signal and a localized right-ear signal to reprocess the localized left-ear and right-ear signals.

12. The method of claim 1, the step of vibrating comprising utilizing a sine wave generator with a frequency variable and a depth variable to dynamically adjust the plurality of positions of the virtual localized sources.

13. The method of claim 2, wherein the periodic function is selected from a group consisting of sinusoidal wave, square wave, and triangular wave.

14. The method of claim 3, further comprising detecting if there is a watermark in the left-ear signal and the right-ear signal by a decoder, if the watermark is present, bypassing the left-ear signal and the right-ear signal and if the watermark is absent, continuing with the step of modifying the signals.

15. The method of claim 1, further comprising dynamically adjusting the positions of the plurality of virtual localized sources in a three dimensional space.

16. A method for improving sound localization of a listener, the method comprising:

receiving a sound signal including a left-ear signal and a right-ear signal in a controller; and

modifying the left-ear signal and the right-ear signal, the step of modifying comprising:

splitting each of the left-ear signal and the right-ear signal into a first frequency band and a second frequency band, the first frequency band having higher frequencies than the second frequency band;

applying a plurality of infinite impulse response (IIR) filters to each of the left-ear signal and the right-ear signal in the first frequency band, the filters being calculated based on each of a left head related transfer function (hrtf) and a right hrtf;

bypassing each of the left-ear and right-ear signals in the second frequency band;

attenuating each of the left-ear signal and right-ear signal in the first band by a band equalizer;

reversing the polarity of the signals in the first frequency band; and

mixing the signals in the first frequency band with the respective bypassed signals in the second frequency band.