Audio signal output device and method of processing an audio signal

The present invention is a method of processing an audio signal comprising outputting a first part of a first audio signal; picking up the output first part of the first audio signal as a second audio signal; comparing a second part of the first audio signal and the second audio signal; modifying the second part of the first audio signal based on the result of the comparison; and outputting the modified second part of the first audio signal. An audio signal output device is also disclosed.

TECHNICAL FIELD

Various embodiments generally relate to the field of audio signal processing, in particular, real-time adaptive audio head-related transfer function (HRTF) system.

BACKGROUND

Advances in digital signal processing (DSP) have led to a proliferation of hardware (HW) and software (SW) developments/solutions that have been applied to various audio systems ranging from traditional 2.1 up to virtual 7.1 audio systems including headphones/headsets. In particular, by taking advantage of these new DSP technologies to a great extent, there have been a significant number of changes in headphones/headsets. Users of headphones, headsets and ear buds are seeing virtualized 5.1 and 7.1 versions come to market. These expanded versions require a lot more audio/sound processing power to achieve audio (sonic) results desired, which closely approximate actual 5.1 and 7.1 sounds, and to achieve optimized audio for gaming purposes.

FIG. 1shows a top view of a schematic diagram of a user100wearing a headphone (or headset)102. The head-related transfer functions (HRTFs) at the right ear cup104and the left ear cup106of the headphone102are represented by HRR108and HLL110, respectively which are used to denote the direct transmission or audio impulses that the right ear and the left ear would respectively perceive. Ideally, in a contained environment, there should be no crosstalk between the right ear cup104and the left ear cup106, i.e., the HRTF from right to left ear cups (HRL112) and the HRTF from left to right ear cups (HLR114) are zero. The right ear cup104and the left ear cup106are independent from each other. However, it should be understood that in practice, audio signals may have inherent crosstalk that may affect the sound perceived by the user.

While advances in HRTF implementations have been realized, they are based on “fixed models” of implementations. This means that these implementations are not adaptive and do not take into account ambient noise or the physical aspect of a human listener's (or user's) ear(s). The listener's outer ear configuration or structure (or pinna) can compound the problem by way of applying an “amplification and/or attenuation factor”, which is related to the human hearing sensitivity, to the incoming audio signature (or signal).FIG. 2shows a schematic diagram of the listener's ear200. The pinna202of the listener's ear200acts as a receiver for the incoming audio signal204through the auditory canal206into the tympanic membrane208. Because of the spreading out of sound energy by inverse square law, a larger receiver, for example, a large pinna202picks up more energy, amplifying the human hearing sensitivity by a factor of about 2 or 3.

Due to the fixed nature of current HRTF implementations it is not possible to account for and adjust for the variables that are known to exist regardless of environment, for example, ambient noise, variability in size and shape of the outer/inner ear canals of a given listener, variable positions of the audio driver(s) in the headset, for example, the headset102ofFIG. 1in relation to the outer/inner ear canal.

Thus, there is a need to provide a method and apparatus for integration within audio devices such as headphones, headsets and ear buds a real-time adaptive audio adjustment system that would significantly improve the perceived sound quality; thereby seeking to address at least the above mentioned problems.

SUMMARY OF THE INVENTION

In a first aspect, the present invention relates to a method of processing an audio signal including outputting a first part of a first audio signal; picking up the output first part of the first audio signal as a second audio signal; comparing a second part of the first audio signal and the second audio signal; modifying the second part of the first audio signal based on the result of the comparison; and outputting the modified second part of the first audio signal.

According to a second aspect, the present invention relates to an audio signal output device including a speaker configured to output a first part of a first audio signal; a microphone configured to pick up the output first part of the first audio signal as a second audio signal; a comparator configured to compare a second part of the first audio signal and the second audio signal; and a circuit configured to modify the second part of the first audio signal based on the result of the comparison, wherein the speaker is further configured to output the modified second part of the first audio signal.

In a third aspect, the present invention relates to a headset including a pair of ear cups; a speaker or number of speakers located in each ear cup; and a microphone located within at least one of the pair of the ear cups, wherein the speaker is substantially centrally located with the ear cup; and wherein the microphone is located adjacent to the speaker.

DETAILED DESCRIPTION

In order that the invention may be readily understood and put into practical effect, particular embodiments will now be described by way of examples and not limitations, and with reference to the figures.

Unique adaptations or implementations of head related transfer functions (HRTFs) continue to evolve. Various embodiments provide a combination (or refined combination) of existing DSP HW technologies combined with unique SW/algorithms that allows for a specific implementation. The way in which various HW and SW elements are arranged within the ear cups and integrated at the SW level allows the raw audio stream to be altered, i.e., modified by way of applying complex real-time signal processing of the audio signature that enters the listen's ears so as to enable the listening experience to be clearer (or more pure). By doing so, this ensures the perceived audio matches as closely as possible the original/raw audio stream as it is intended to be heard.

Various embodiments comprise a unique combination or blend of audio DSP technologies and microphone elements positioned in the ear pieces in such a way that the ear pieces pick up the right/left audio signatures altered by how the sound bounces off the outer ear canal and then a comparison of the original/raw audio source left and right channel is performed. The real time adaptive DSP technologies invoke and alter the original raw audio stream at the DSP level and ensure that the perceived sound signature, at the outer ear matches as closely as possible the original/raw audio stream.

Various embodiments provide frequency corrections on the original raw audio stream based on a unique HW driver in the ear cup of the headphone. Frequency corrections may be related to or associated with other algorithmic functions, for example, amplitude corrections (that is, amplification corrections or attenuation corrections) and phase shift corrections (or delay corrections).FIG. 3shows a block diagram of an exemplary real-time adaptive inverse filtering process. InFIG. 3, an input signal300is fed into a desired transfer function D302and an adaptive filter A304. The output from the desired transfer function D302is a desired signal306which is compared with a measured signal308by a comparator310to give an error signal312. The measured signal308is obtained from the output of a real transfer function R314which accepts a driving signal316as its input. The driving signal316is in turn obtained from the output of the adaptive filter A304, which has filtering parameters adapted in accordance to the error signal312. The adaptive filter as seen inFIG. 3is an example of a specific underlying algorithm for adaptively processing an audio signal in real-time.

In other words, for example, wave synthesis may be comparing a base line audio wave to a reflected audio wave from the microphones that are placed in each ear cup. The microphones may be placed at various locations in each ear cup. However, when placed at certain locations or strategic locations, the microphones can receive, for example, the maximum level of reflected audio wave; thereby enhancing the picking up of the desired audio signal for processing.

Wave synthesis may be applied in real time and is the process whereby, for example inFIG. 3, the raw or incoming audio wave is digitally sampled and then compared to a digital sample of the reflected audio wave from each ear cup. A third or audio wave results after the correction factors are applied, (i.e. amplification, attenuation, phase shift, delay, echo and/or noise cancellation). Wave synthesis applies the correction factors in real time and produces a third and unique audio wave that is reconstructed by applying the correction factor to as closely as possible approximate the initial or raw audio wave.

FIG. 4shows an exemplary overview of a combination (or refined combination) of existing DSP HW technologies combined with unique SW/algorithms that allows for a specific implementation.

InFIG. 4, a raw audio stream (or signal)400is input into a system402including a DSP function404. The system402may be but is not limited to an external audio PUCK/MICX amplifier. The raw audio stream400may be modified by the DSP function404to a modified audio stream (or signal)406, output by the system402. The DSP function404may also be used to perform some amount of processing for changes in amplitude, attenuation and/or other signal anomalies such as echo and or noise cancellation. The modified audio stream406is then fed into the left and right ear cups408,410of a headset412. A user (not shown inFIG. 4) positions his/her head between the left and right ear cups408,410as shown by a directional symbol414. The ear cups408,410may be positioned against the user's respective ears (not shown inFIG. 4) as shown by arrows416,418respectively.

A microphone420(MIC “L”) in the left ear cup408and a microphone422(MIC “R”) in the right ear cup410respectively pick up a MIC (L/R) audio signal424that is fed back into a comparator426. The comparator426also receives the raw audio stream400and compares this raw audio stream400and the MIC (L/R) audio signal424. The comparator426outputs result(s) of the comparison428which is fed back into the system402. The system402receives the result(s)428and modifies the raw audio stream400based on the results(s)428.

In order for the comparator426to perform the comparison of the MIC (L/R) audio signal424with respect to the corresponding raw audio stream400, a delay is introduced to the raw audio stream400by a phase shifter430before entering the comparator426; thereby providing a form of timing synchronization between the two signals for comparison.

In order for the system402to perform the modification of the raw audio stream400based on the corresponding result(s) of the comparison428, another delay is introduced to the result(s) of the comparison428by another phase shifter432before entering the system402; thereby providing a form of timing synchronization between the signals for modification.

For the example inFIG. 4, all the audio signals may be digital signals.

In other examples, some audio signals at certain processing steps may be analog or digital. For example, the raw audio stream may be analog or digital. If the raw audio stream is analog, the system converts the raw audio stream into a digital signal so that DSP functions can be applied.

In a first aspect, a method of processing an audio signal500is provided as shown inFIG. 5. At502, a first part of a first audio signal is output. For example, the first part of the first audio signal may refer to the modified audio stream406ofFIG. 4and the first audio signal may refer to the raw audio stream400ofFIG. 4. The first part of the first audio signal refers to an audio signal over a period of time, for example, denoted as X. The term “audio signal” may interchangably be referred to as “audio stream” which may represent any audio signal originating from any audio signal source, for example, a playback audio track.

At504, the output first part of the first audio signal is picked up as a second audio signal. For example, the second audio signal may refer to the MIC (L/R) audio signal424ofFIG. 4. As used herein, the term “pick up” or “picked up” may generally refer to being received.

At506, a second part of the first audio signal and the second audio signal are compared. For example, the second part of the first audio signal may refer to an audio signal based on the raw audio stream400ofFIG. 4that is fed through the system402with the DSP function404and into an input of the comparator426. In another example (not shown), the second part of the first audio signal may be an audio signal based on the raw audio stream and is fed into an input of the comparator without going through the system with the DSP function.

At508, the second part of the first audio signal is modified based on the result of the comparison. For example, the result of the comparison refers to the result(s) of the comparison428ofFIG. 4.

As used herein, the term “modify” refers but is not limited to change, adjust, amplify, or attenuate. For example, the second part of the first audio signal may be modified by amplifying its amplitude based on the result of comparison which may be an amplification correction factor. In another non-limiting example, the second part of the first audio signal may be modified by changing its frequency based on the result of comparison which may be a frequency correction factor. It should be appreciated that modification can take any form of change or a combination of changes in accordance to the result of comparison. Due to the feedback mechanism, the modification may be referred to as an adaptive modification. The object of the modification is to obtain a perceived sound signature at a user's outer ear that matches the original/raw audio stream as closely as possible.

At510, the modified second part of the first audio signal is output.

For example, the modified second part of the first audio signal may refer to the modified audio stream406ofFIG. 4over another period of time, for example, denoted as Y. In one example, the time periods X and Y may be adjacent time periods. In another example, at least parts of the time periods X and Y may be overlapped.

In various embodiments, the steps of outputting at502,510, picking up at504, comparing at506and modifying at508are repeated at a predetermined time interval that allows substantially real-time processing of the audio signal. For example, after the modified second part of the first audio signal is output at510, the steps provided by the method500may be repeated such that the modified second part of the first audio signal now becomes the first part of the first audio signal at502. In this case, the first part of the first audio signal now refers to an audio signal over the other period of time, for example, denoted as Y.

The method500may be repeated at intervals or may be repeated continuously so as to provide substantially real-time audio signal processing. It should be appreciated and understood that the term “substantially” may include “exactly” and “similar” which is to an extent that it may be perceived as being “exact”. For illustration purposes only and not as a limiting example, the term “substantially” may be quantified as a variance of +/−5% from the exact or actual. For example, the phrase “A is (at least) substantially the same as B” may encompass embodiments where A is exactly the same as B, or where A may be within a variance of +/−5%, for example of a value, of B, or vice versa.

In various embodiments, the step of outputting the first part of the first audio signal at502may include outputting the first part of the first audio signal through a speaker of a headset.

In the context of various embodiments, the term “headset” may refer to a device having one or more earphones usually with a headband for holding them over the ears of a user. In some examples, the term “headset” may interchangably refer to headphone, ear piece, ear phone, or receiver.

In an example, a headset includes ear phones in the form of ear cups, for example, the ear cups408,410ofFIG. 4. Each ear cup may include a cushion that surrounds the peripheral circumference of the ear cup. When a user places the ear cup over the ear, the cushion covers the ear to provide an enclosed environment around the ear in order for an audio signal to be directed into the auditory canal of the ear.

As used herein, the term “speaker” generally refers to an audio transmitter of any general form and may be interchangably referred to as a loudspeaker. The speaker may include an audio driver. The speaker may be encased within the ear cup of the headset.

In various embodiments, the step of picking up the output first pan of the first audio signal as the second audio signal at504may include receiving the output first part of the first audio signal by a microphone. The microphone may be strategically positioned within the ear cup such that the microphone receives the maximum level of audio signal and/or the microphone receives the similar audio signal as received by the ear canal of a wearer of the headset.

As used herein, the term “microphone” generally refers to an audio receiver of any general form. For example, the microphone may be a microelectromechanical system (MEMS) microphone. A MEMS microphone is generally a microphone chip or silicon microphone. To form the MEMS microphone, a pressure-sensitive diaphragm is etched directly into a silicon chip by MEMS techniques, and is usually accompanied with integrated preamplifier. Most MEMS microphones are variants of the condenser microphone design. Often MEMS microphones have built in analog-to-digital converter (ADC) circuits on the same CMOS chip making the chip a digital microphone and so more readily integrated with digital products. The MEMS microphone is typically compact and small in size, and can receive audio signals across a wide angle of transmission. The MEMS microphone also has a flat response over a wide range of frequencies.

In various embodiments, the microphone may be located within an ear cup of the headset such that when a wearer wears the headset, the microphone may be configured to be positioned substantially near the entrance of the ear canal of the wearer.

As used herein, the term “wearer” may interchangably be referred to as the user. The term “substantially” may be as defined above. In this context, the term “near” refers to being in close proximity such that the microphone and ear canal both receive at least similar audio signals. The term “ear canal” refers to the auditory canal of the ear.

In various embodiments, the second audio signal may include a left channel audio signal and a right channel audio signal of the headset. For example, the left channel audio signal and the right channel audio signal may refer to MIC (L/R) audio signal424ofFIG. 4.

In an embodiment, the second audio signal may further include a noise signal.

As used herein, the phrase “noise signal” generally refers to any undesired signals which may include unwanted audio signals and/or electrical noise signals that is attributed by the various electronic components (eg. microphone or electrical conductor). Electrical noise signals may include, for example, crosstalk, thermal noise, shot noise. Unwanted audio signals may include, for example, sounds from the environment.

In various embodiments, the output first part of the first audio signal may include a reflection of the first part of the first audio signal. In the context of various embodiments, the term “reflection” refers to an echo.

In an embodiment, the reflection of the first part of the first audio signal may include a reflection of the first part of the first audio signal from at least part of a pinna of a wearer of the headset. The reflected signal may be conditioned by processing for echo and noise cancellation correction factors.

As used herein, the term “pinna” means the outer ear structure that form one's unique ear shape.

For example, when a wearer (or user) wears the headset, the audio signal is output from the speaker of the headset and travels to the ear. Parts of the audio signal may enter into the ear canal while other parts of the audio signal may reach the pinna of the ear. The other parts of the audio signal or parts thereof may bounce off or reflect from the surface of the pinna and may be picked up by the microphone.

In another example, parts of the audio signal may enter into the ear canal while other parts of the audio signal may reach a surface of the ear cup that forms an at least substantially enclosed area with the ear. The other parts of the audio signal or parts thereof may bouce off or reflect from this surface of the ear cup and may be picked up by the microphone.

In various embodiments, the step of comparing the second part of the first audio signal and the second audio signal at506may include comparing at least one of the amplitude of the second part of the first audio signal and the amplitude of the second audio signal to obtain an amplitude correction factor, the frequency of the second part of the first audio signal and the frequency of the second audio signal to obtain a frequency correction factor, or the phase of the second part of the first audio signal and the phase of the second audio signal to obtain a phase correction factor.

For example, the amplitude correction factor, the frequency correction factor, and/or the phase correction factor may be the result(s) of the comparison428ofFIG. 4.

The term “comparing” may refer but is not limited to taking the difference of two or more signals. For example, the term “comparing” may also include a weight or a multiplication factor applied on the difference.

In various embodiments, the step of modifying the second part of the first audio signal at508may include modifying the second part of the first audio signal based on at least one of the amplitude correction factor, the frequency correction factor or the phase correction factor. For example, the second part of the first audio signal may be modified based on the amplitude correction factor, or the frequency correction factor, or the phase correction factor, or the combination of the amplitude correction factor and the frequency correction factor, or the combination of the amplitude correction factor and the phase correction factor, or the combination of the phase correction factor and the frequency correction factor, or the combination of the amplitude correction factor and the frequency correction factor and the phase correction factor.

In various embodiments, the step of modifying the second part of the first audio signal at508may include increasing or decreasing at least one of the amplitude, the frequency or the phase of the second part of the first audio signal.

In various embodiments, the step of modifying the second part of the first audio signal at508may include modifying the second part of the first audio signal based on a Head Related Transfer Function (HRTF).

In the context of various embodiments, a head-related transfer function (HRTF) is a response that characterizes how an ear receives a sound from a point in space. A pair of HRTFs for two ears may be used to synthesize a binaural sound that seems to come from a particular point in space. In general, HRTF is a transfer function describing how a sound from a specific point arrives at the ear or the pinna.

In various embodiments, the second part of the first audio signal is modified based on a dynamic HRTF. In other words, the dynamic HRTF changes according to severals factors, for example, a change in the position of the ear and/or a change in the received audio signal. This is in contrast to existing HRTFs which are static and do not change. For example, existing stereo sound systems may use static HRTF for their respective signal processing.

In various embodiments, the method500may further include prior to comparing the second part of the first audio signal and the second audio signal at506, a delay may be added to the second part of the first audio signal.

The delay may be performed by a phase shifter such as the phase shifter430ofFIG. 4. The purpose of adding a delay is to provide a form of timing synchronization between the two signals for comparsion such that the second audio signal may be compared against the corresponding part of the first audio signal.

In various embodiments, the method500may further include prior to modifying the second part of the first audio signal at508, another delay may be added to the result of the comparison.

The other delay may be performed by a phase shifter such as the phase shifter432ofFIG. 4. The purpose of adding the other delay is to provide a form of timing synchronization between the signals for modification such that the second part of the first audio signal may be modified based on the corresponding result of the comparison.

In various embodiments, the second part of the first audio signal may be an analog signal or a digital signal. If the second part of the first audio signal is an analog signal, the method500may further include converting the analog second part of the first audio signal into a digital signal. The digital signal may be in any format, for example, represented by parallel bits or serial bits and may be of any resolution, for example but not limited to 8-bit representation, 16-bit representation, 32-bit representation, 64-bit representation, or other representations higher than 64-bit representation.

In a second apsect, an audio signal output device600is provided as shown inFIG. 6. The audio signal output device600includes a speaker602configured to output a first part of a first audio signal; a microphone604configured to pick up the output first part of the first audio signal as a second audio signal; a comparator606configured to compare a second part of the first audio signal and the second audio signal; and a circuit608configured to modify the second part of the first audio signal based on the result of the comparison, wherein the speaker602is further configured to output the modified second part of the first audio signal.

For example, the speaker602may be the respective speaker found in the left and right ear cups408,410ofFIG. 4. The microphone604may be as defined hereinabove and may be the microphone MIC “L”420or the microphone MIC “R”422ofFIG. 4. The comparator606may refer to the comparator426ofFIG. 4. The comparator606may be a summing circuit and may be a digital comparator (i.e., a comparator comparing digital signals). The circuit608may refer to the system402ofFIG. 4with the DSP function404.

In other examples, the circuit608may be integrated within the ear cup, for example, the left and/or right ear cups408,410ofFIG. 4.

In the context of various embodiments, a “circuit” may be understood as any kind of a logic implementing entity, which may be special purpose circuitry or a processor executing software stored in a memory, firmware, or any combination thereof. Thus, a “circuit” may be a hard-wired logic circuit or a programmable logic circuit such as a programmable processor, e.g. a microprocessor (e.g. a Complex Instruction Set Computer (CISC) processor or a Reduced Instruction Set Computer (RISC) processor). A “circuit” may also be a processor executing software, e.g. any kind of computer program, e.g. a computer program using a virtual machine code such as e.g. Java or e.g. digital signal processing algorithm. Any other kind of implementation of the respective functions which are described may also be understood as a “circuit” in accordance with an alternative aspect of this disclosure.

In various embodiments, the speaker602, the microphone604, the comparator606and the circuit608may be configured to operate repetitively at a predetermined time interval that allows substantially real-time audio signal processing.

The term “substantially” is as defined above. The term “real-time” means a time-frame in which an operation is performed that is acceptable to and perceived by a user to be similar or equivalent to actual clock times. “Real-time” may also refer to a deterministic time in response to real world events or transactions where there is no strict time related requirement. For example, in this context, “real-time” may relate to operations or events occuring in microseconds, milliseconds, seconds, or even minutes ago.

In an example, the predetermined time interval may be but is not limited to a range of about 1 μs to about 100 μs, or about 10 μs to about 50 μs, about 1 ms to about 100 ms, or about 10 ms to about 50 ms, about 1 s to about 10 s.

The term “repetitively” refers to performing over and over.

The terms “microphone”, “first part of the first audio signal”, “second audio signal”, “second part of the first audio signal”, “compare”, “modify”, “result of the comparison” and “modified second part of the first audio signal” may be as defined above.

In various embodiments, the comparator606may be configured to compare at least one of the amplitude of the second part of the first audio signal and the amplitude of the second audio signal to obtain an amplitude correction factor, the frequency of the second part of the first audio signal and the frequency of the second audio signal to obtain a frequency correction factor, or the phase of the second part of the first audio signal and the phase of the second audio signal to obtain a phase correction factor.

The phrases “amplitude correction factor”, “frequency correction factor” and “phase correction factor” may be defined as above.

In various embodiments, the circuit608may be configured to modify the second part of the first audio signal based on at least one of the amplitude correction factor, the frequency correction factor or the phase correction factor. For example, the circuit608may be configured to increase or decrease at least one of the amplitude, the frequency or the phase of the second part of the first audio signal. The circuit608may also be configured to modify the second part of the first audio signal based on a Head Related Transfer Function (HRTF).

The phrase “HRTF” may be as defined above.

In various embodiments, the audio signal output device600may further include a phase shifter configured to add a delay to the second part of the first audio signal.

In other embodiments, the audio signal output device600may further include another phase shifter configured to add another delay to the result of the comparison.

The phase shifter and the other phase shifter may refer to the phase shifter430and the phase shifter432ofFIG. 4, respectively. The phase shifter (or delay block) may be used if there is a phase or delay measured as a result of the signal going through the various components or devices during processing.

In various embodiments, the audio signal output device600may further include an analog-to-digital converter configured to convert the analog second part of the first audio signal into a digital signal.

In a third aspect, a headset700is provided as shown inFIG. 7. The headset700includes a pair of ear cups702; a speaker704located in each ear cup702; and a microphone706located within at least one of the pair of the ear cups702, wherein the speaker704is substantially centrally located with the ear cup702; and wherein the microphone706is located adjacent to the speaker704.

The term “adjacent” refers to neighbouring, next to or alongside.

For example, the pair of ear cups702may refer to the left and right ear cups408,410ofFIG. 4, the speaker704may be the respective speaker found in the left and right ear cups408,410ofFIG. 4, and the microphone706may be the microphone MIC “L”420and/or the microphone MIC “R”422ofFIG. 4.

In various embodiments, the microphone706may be located below the speaker704such that when a wearer wears the headset, the microphone706is configured to face a substantially lower part of the external auditory canal of the wearer.

As used herein, the phrase “external auditory canal” may interchangably be referred to as ear canal or auditory canal.

In an embodiment, the microphone706may be located within an area having a radius of about 1 cm to 2 cm from the substantially centrally located speaker704. In other examples, the microphone706may be located about 0.5 cm, about 1 cm, about 1.2 cm, about 1.5 cm, about 1.8 cm, about 2 cm, about 2.2 cm, or about 2.5 cm from the substantially centrally located speaker704.

In some embodiments, the headset700may include a plurality of speakers in each ear cup. For example, the headset700may include 2 or 3 or 4 or 5 speakers in each ear cup.

The term “microphone” may be as defined above.

Various embodiments provide an adaptive method and device that adjusts the (original) raw audio stream, e.g. the raw audio stream400inFIG. 4in real-time, allowing for altering the (original) raw audio stream in such a way as to give the listener (wearer) the perception regardless of the position of audio driver in relation to the outer ear and its unique shape that the audio content is whole, intact and retains the intended sound signature.

The real-time adaptive part of the approach, for example as described inFIG. 3may be based on a unique combination of specific HW driver frequency corrections specific to the headset and a SW wave synthesis algorithm that adjusts in real-time other critical audio factors for example phase, delay, signal amplitude, (attenuation/amplification) factors based on a comparison to the initial audio signal. In some examples, both the correction and algorithm may take place in a system with DSP function(s), for example, the system402ofFIG. 4.

By way of strategic and optimized placement of the digital silicon or MEMs microphone near the entry of the ear cannel leading to the tympanic membrane as depicted inFIG. 2and at a distance that allows the microphone to pick up key audio impulses from the outer ear or pinna, the adaptive method and device for processing the audio signal may be achieved.

FIG. 8Ashows a cross-sectional side view of an exemplary ear cup800of a headset. In this example, five speakers802,804,806,808and810are shown to be located within the ear cup800with speaker808being substantially centrally located in the ear cup800. The rest of the speakers802,804,806and810are positioned around the central speaker808. For example, speaker802is positioned top-left to speaker808; driver804is positioned bottom-left to speaker808; driver806is positioned top-right to speaker808; and driver810is positioned bottom-right to speaker808.

FIG. 8Bshows the exemplary ear cup800ofFIG. 8Adepicting the positions of various drivers.

InFIG. 8B, five (audio) drivers820,822,824,826,828are located at the respective speakers802,804,806,808,810. When a wearer wears the headset with the ear cup800over the ear resulting in the upright orientation of the ear cup800as shown inFIG. 8B, the wearer faces to the left and the ear cup800is the left ear cup for the wearer. Driver820may be a front driver with a diameter of about 30 mm; driver822may be a center driver with a diamater of about 30 mm; driver824may be a surround back driver with a diameter of about 20 mm; driver826may be a subwoofer driver with a diameter of about 40 mm; and driver828may be a surround driver with a diameter of about 20 mm.

FIG. 8Cshows the exemplary ear cup800ofFIG. 8Adepicting the preferred (or ideal) position of the MEMS microphone830. InFIG. 8C, the MEMS microphone is positioned along the central axis832and near the bottom of the ear cup800, that is, below the center driver822and the surround driver828.

FIG. 8Dshows the exemplary ear cup800ofFIG. 8Adepicting three possible areas840,842,844where a MEMS microphone may be located and the effects thereof.

For example, having the MEMS microphone located in the area840is non-ideal as the area840is located furthest from the ear canal of the wearer. The MEMS microphone located in the area842allows adaptive audio signal processing to work and is better as compared to being located in the area840. Having the MEMS microphone located in the area844is (most) ideal since the area844is located nearest to the ear canal of the wearer.

The method according to various embodiments as described above may adapt itself more to audio listening environment especially at the micro level (for example, at the inlet to the ear as the audio signal (or sound) enters the outer ear) where there are inherent differences in the surface (that is provided by the shape of a user's outer ear or pinna and inner ear canal) that channels the audio signal or sound to the tympanic membrane. The described method also can take into account the ambient noise levels and applying noise cancellation approaches that are different depending upon the listening environment. In contrast, existing HRTF functions are static in nature and cannot account for or correct for these eventualities/environmental factors.

By applying the described method, a comparison between a modified audio signal and the corresponding original audio signal was made.FIG. 9shows the modified audio signals900,902based on an amplitude correction factor and the corresponding original audio signals904,906over the frequency range of 100 Hz to 20 KHz for (A) the left ear and (B) the right ear. It is noted that an inherent difference of about 4 dB to about 8 dB between the right and left ear.

As seen inFIG. 9, the modified audio signals900,902are attenuated from the original audio signals904,906based on the amplitude correction factor. A user preceives the original audio signals904,906when wearing a headset outputting the modified audio signals900,902. Conclusively,FIG. 9shows an example of an original audio wave and the resulting wave after wave synthesis or correction factors have been applied.

In the context of various embodiments, the term “about” as applied to a numeric value encompasses the exact value and a variance of +/−5% of the value.