Patent Grant
9635484
The present invention relates to a method for reproducing surround audio signals.
Audio systems as well as headphones are known, which are able to produce a surround sound.
Headphones are also known, which are able to produce a ‘surround’ sound such that the listener can experience for example a 5.1 surround sound over headphones or earphones having merely two electric acoustic transducers.
On the one hand, the Room Reproduction may create an impression of an acoustic space and may create an impression that the sound comes from outside the user's head. On the other hand, the Room Reproduction may also color the sound, which can be unacceptable for high fidelity listening.
Accordingly, it is an object of the invention to provide a method for reproducing audio signals such that the auditory spatial and timbre cues are provided such that the human brain has the impression that a multichannel audio content is played.
This object is solved by a method according to claim 1.
This object is solved by a method for providing surround audio signals. Input surround audio signals are received and are binaurally filtered by means of at least one filter unit. On the input surround audio signals, a binaural equalizing processing is performed by at least one equalizing unit. The binaurally filtered signals and the equalized signals are combined as output signals.
According to an aspect of the invention, the filtering and the equalizing processing are performed in parallel.
Furthermore, the filtered and/or equalized signals can be weighted.
Furthermore, in a real-time implementation, the amount of room effect RE included in both signal paths can be weighted,
The invention also relates to a surround audio processing device. The device comprises an input unit for receiving surround audio signals, at least one filter unit for binaurally filtering the received input surround audio signals and at least one equalizing unit for performing a binaural equalizing processing on the input surround audio signals. The output signals of the filter units and the output signals of the equalizing units are combined.
Optionally, the binaural filtering unit can comprise a room model reproducing the acoustics of a target room, and may optionally do so as accurately as computing and memory resources allow for.
According to a further aspect of the invention, the surround audio processing device comprises a first delay unit arranged between the input unit and at least one equalizing unit for delaying the input surround audio signal before it is processed by the equalizing unit. The device furthermore comprises a second delay unit for delaying the output of the at least one equalizing unit.
According to a further aspect of the invention, the device comprises a controller for weighting the output signals of the filter units and/or the output signals of the equalization units.
The invention also relates to a headphone comprising an above described surround audio processing device.
The invention also relates to a headphone which comprises a head tracker for determining the position and/or direction of the headphone and an audio processing unit. The audio processing unit comprises at least one filter unit for binaurally filtering the received input surround audio signals and at least one equalizing unit for performing a binaural equalizing processing on the input surround audio signals. The output signals of the filter units and the equalizing units are combined as output signals.
The invention relates to a headphone reproduction of multichannel audio content, a reproduction on a home theatre system, headphone systems for musical playback and headphone systems for portable media devices. Here, binaural equalization is used for creating an impression of an acoustic space without coloring the audio sound. The binaural equalization is useful for providing excellent tonal clarity. However, it should be noted that the binaural equalization is not able to provide an externalization of a room impulse response or of a room model, i.e. the impression that the sound originates from outside the user's head. An audio signal convolved or filtered with a binaural filter providing spaciousness (with a binaural room impulse response or with a room model) and the same audio signal which is equalized, for example to correct for timbre changes in the filtered sound, is combined in parallel.
Optionally directional bands can be used during the creation of an equalization scheme for compensating for timbre changes in binaurally recorded sound or binaurally processed sound. Furthermore, stereo widening techniques in combination with the direction of frequency band boosting can be used in order to externalize an equalized signal which is added to a process sound to correct for timbre changes. Accordingly, a virtual surround sound can be created in a headphone or an earphone, in portable media devices or for a home theatre system. Furthermore, a controller can be provided for weighting the audio signal convolved or filtered with a binaural impulse response or the audio signal equalized to correct for timbre changes. Therefore, the user may decide for himself which setting is best for him.
By means of an equalizer that excites frequency bands corresponding to spatial cues, the spatial cues already rendered by the binaural filtering are reinforced or do not lead to an alteration of the spatial cues. By separating the rendering of the spatial cues provided by the binaural filters and by rendering the correct timbre by providing the equalizer, a flexible solution is provided which can be tuned by the end-user, wherein he can choose whether he wishes more spaciousness vs. more timbre preservation.
Other aspects of the invention are defined in the dependent claims.
Advantages and embodiments of the invention are now described in more detail with reference to the figures.
It should be noted that “Ipsi” and “Ipsilateral” relate to a signal which directly hits a first ear while “contra” and “contralateral” relate to a signal which arrives at the second ear. If in
In some embodiments, the filter units CU can cause attenuation in the low frequencies (e.g., 400 Hz and below) and in the high frequencies (e.g., 4 Hz and above) in the audio signals presented at the ears of the user. Also, the sound that is presented to the user can have many frequency peaks and notches that reduce the perceived sound quality. In these embodiments, the equalization filters EQFI, EQFC may be used to construct a flat-band representation of right and left signals (without externalization effects) for the user's ears which compensates for the above-noted problems. In other embodiments, the equalization filters may be configured to provide a mild amount of boost (e.g., 3 dB to 6 dB) in the above-noted low and high frequency ranges. As illustrated in the embodiment shown in
Binaural Filters Database and Binaural Equalizers Database can store the coefficients for the filter units or convolution units. The coefficients can optionally be based upon a given “virtual source” position of a loud speaker. The auditory image of this “virtual source” can be preserved despite the head movements of the listener thanks to a head tracker unit as described with respect to
The output of the filters can be summed (e.g., added) for the left ear and the right ear of a user, which can be provided to Output Ipsi and Output Contra. In certain embodiments, the surround audio processing unit of
Each equalizing unit EQF, EQR can have one or two outputs, wherein one output can relate to the Ipsi signal and one can relate to the contra signal. The delay unit and/or a gain unit G can be coupled to the outputs. One output can relate to the left side and one can relate to the right side. The outputs of the left side are summed together and the outputs of the right side are also summed together. The result of these two summations can constitute the left and right signal L, R for the headphone. Optionally, a stereo widening unit SWU can be provided.
In the stereo widening processing unit SWU the output signals of the equalization units EQF, EQR are phase inverted (−1) reduced in their level and added to the opposite channel to widen the sound image.
The outputs of all filters can enter a final gain stage, where the user can balance the equalization units EQFI, EQFC with the convolved signals from the convolution or filter units CU. The bands which are used for the binaural equalization process can be a front-localized band in the 4-5 kHz region and to back-localized bands localized in the 200 and 400 Hz ranges. In some instances, the back-localized bands can be localized in the 800-1500 Hz range.
The method or processing described above can be performed in or by an audio processing apparatus in or for consumer electronic devices. Furthermore, the processing may also be provided for virtual surround home theatre systems, headphone systems for music playback and headphone systems for portable media devices.
By means of the above described processing the user can have room impulses as well as a binaural equalizer. The user will be able to adjust the amount of either signal, i.e. the user will be able to weight the respective signals.
These sets of parameters can be derived from head-related transfer functions (HRTF), which can be measured as described in
The head position as determined by the head tracker HT is forwarded to the audio processing unit APU and the audio processing unit APU can extract the corresponding set of filter parameters and equalization parameters which correspond to the detected head position. Thereafter, the audio processing unit APU can perform an audio processing on the received multi-channel surround audio signal in order to provide a left and right signal L, R for the electro-acoustic transducers of the headset.
The audio processing unit according to the third embodiment can be implemented using the filter units CU and/or the equalization units EQFI, EQFC according to the first and second embodiments of
According to a fourth embodiment, a convolution and filter units CU and one of the equalization units EQFI, EQFC according to
According to a fifth embodiment, the audio processing unit as described according to the third embodiment can also be implemented as a dedicated device or be integrated in an audio processing apparatus. In such a case, the information from the head tracker of the headphone can be transmitted to the audio processing unit.
According to a sixth embodiment which can be based on the second embodiment, the programmable delay unit D is provided at each output of the equalization units EQF, EQR. These programmable delay units D can be set as stored in the parameter memory PM.
It should be noted that Ipsi relates to a signal which directly hits a first ear while the signal contra relates to a signal which arrives at the second ear. If in
It should be noted that a convolution unit or a pair of convolution units is provided for each of the multi-channel surround audio channels. Furthermore, an equalizing unit or a pair of equalizing units is provided for each of the multi-channel surround audio channels. In the embodiment of
It should be noted that in
The delay unit DU2 in
It should be noted that the equalizing units are merely serve to improve the quality of the signal. In further embodiments described below, the equalizing units can contribute to localization.
It should be noted that virtual surround solutions according to the prior art make for example use of a binaural filtering to reproduce the auditory spatial and timbre cues that the human brain would receive with a multichannel audio content. According to the prior art, binaurally filtered audio signals are used to deal with the timbre issues. Furthermore, the use of convolution reverb for binaural synthesis, the use of notch and peak filters to simulate head shadowing and the use of binaural recording for binaural synthesis is also known. However, the prior art does not address the use of an equalization used in parallel with a binaural filtering to correct for timbre. The filters used for the binaural filtering focus on reproducing accurate spatial cues and do not specifically care about the timbre produced by this filtering. However, a timbre changed by the binaural filtering is often perceived as altered by the listeners. Therefore, listeners often prefer to listen to a plain stereo down-mix of the multichannel audio content rather than the virtual surround processed version.
The above-described equalizer or equalizing unit can be an equalizer with directional bands or a standard equalizer without directional bands. If the equalizer is implemented without directional bands, the preservation of the timbre competes with the reproduction of spatial cues.
By measuring impulse responses of an audio processing method, it can be detected whether the above-described principles of the invention are implemented.
It may be appreciated that the above embodiments of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
Embodiments of a binaural filtering unit can comprise a room model reproducing the acoustics of a target room as accurately as computing and memory resources allow for. The filtering unit can produce a binaural representation of the early reflections ER that is accurate in terms of time of arrival and frequency content at the listener's ears (such as resources allow for). In certain embodiments, the method can use the combination of a binaural convolution as captured by a binaural room impulse response for the first early reflections and, for the later time section of the early reflections, of an approximation or model. This model can consist of two parts as shown in system 850 of
Embodiments disclosed herein include methods to reproduce as many geometrically accurate early reflections ER in a room model as resources allow for, using a geometrical simulation of the room. One exemplary method can simulate the geometry of the target room and can further simulate specular reflections on the room walls. Such simulation generates the filter parameters for the binaural filtering unit to use to provide the accurate time of arrival and filtering of the reflections at the centre of the listener's head. The simulation can be accomplished by one of ordinary skill in the acoustical arts without undue experimentation.
In certain embodiments, the reflections can be categorized based on the number of bounces of the sound on the wall, commonly referred to as first order reflections, second order reflections, etc. Thus, first order reflections have one bounce, second order reflections have two bounces, and so on.
The low order reflections may be chosen by determining the N tap-outs (835a through 835n) from the delay line 830. The delay of each tap-out may be chosen to be within the selectable time limit. For example, the selectable time limit may comprise 42 ms. In this example, six tap-outs may be chosen with delays of 17, 19, 22, 25, 28, and 31 ms. Other tap-outs may be chosen. Each tap-out can represent a low order reflection within the selectable time limit as shown by reflections 810 in
In certain embodiments, a five channel surround audio may be used. Each channel can comprise an input. Thus there may be five systems 850 per ear. The system 850 of
It may be appreciated that the above embodiments of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
Each tap-out (835a through 835n) of
The basis filters 713a, 713b, and 713c can then be used to process the reflection outputs, in place of filters 830a . . . 830n of
The fixed filter system can then connect to reflection 2 using connection 721 or other suitable connection, and repeat the process using the appropriate gains g0, g1, and g2. This result can also be stored in summing buses 1, 2, and 3, along with the previously stored reflection 1. This process can be repeated for all reflections. Thus, reflection 1 through reflection N can be split, multiplied by an appropriate gain, and stored in the summing buses. Once all N reflections are so stored, the summing buses can be activated so that the stored reflections are multiplied by the appropriate basis filters 713a, 713b, and 713c. The outputs of the basis filters can then be summed together to provide an output corresponding to section 820 of
Embodiments of the fixed filtering disclosed herein can provide a method to produce a binaural representation of the early reflections ER. Exemplary embodiments can create representations to be as accurate in terms of time of arrival (as described with respect to
It may be appreciated that the above embodiments of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
Appropriate Initial Echo Density from Feedback Delay Network
According to an exemplary embodiment, the filter units CU according to
The plurality of inputs 801 is connected to the mixing matrix 802 and an associated feedback loop (loop 0 . . . loop N). In certain embodiments, the mixing matrix 802 can have N inputs 801 by N outputs 804 (such as 12×12). The mixing matrix can take each input 801, and mix the inputs such that each individual output in the outputs 804 contains a mix of all inputs 801. Each output 804 can then feed into a delay line 806. Each delay line 806 can have a left tap-out 803 (L0 . . . LN), a right tap-out 804 (R0 . . . RN), and a feedback tap-out 807. Thus, each delay line 806 may have three discrete tap-outs. Each tap-out can comprise a delay, which can approximate the late reverberation LR with appropriate echo density. Each feedback tap-out can be added back to the input 801 of the mixing matrix 802. In exemplary embodiments, the right tap-out 804 and the left tap-out 803 may occur before the feedback tap-out 807 for the corresponding delay line (i.e., the delay line tap-out occurs after the left and right tap-outs for each delay line). In certain embodiments, every right tap-out 804 and the left tap-out 803 may also occur before the feedback tap-out for the shortest delay line. Thus, in the example shown in
Embodiments of the FDN 800 can be used in a model of the room effect RE that reproduces with perceptual accuracy the initial echo density of the room effect RE with minimal impact on the spectral coloration of the resulting late reverb. This is achieved by choosing appropriately the number and time index of the tap-outs 803 and 804 as described above along with the length of the delay lines 806. In one aspect, each individual left tap-out L0 . . . LN can each have a different delay. Likewise, each individual right tap-out R0 . . . RN can each have a different delay. The individual delays can be chosen so that the outputs have approximately flat frequencies and are approximately uncorrelated. In certain embodiments, the individual delays can be chosen so that the outputs each have an inverse logarithmic spacing in time so that the echo density increases appropriately as a function of time.
The left tap-outs can be summed to form the left output 805a, and the right tap-outs can be summed to form the right output 805b. The output of the FDN 800 preferably occurs after the early reflections ER, otherwise the spatialization can be compromised. Embodiments described herein can select the initial output timing of the FDN 800 (or tap-outs) to ensure that the first echoes generated by the FDN 800 arrive in the appropriate time frame.
The choice for the tap-outs 803 and 804 can also take into account the need for uncorrelated left and right FDN 800 outputs. This can ensure a spacious Room Reproduction. The tap-outs 803 and 804 may also be selected to minimize the perceived spectral coloration, or comb filtering, of the reproduced late reverberation LR. As shown in
In exemplary embodiments, the FDN will not overlap with the output of the system 850 shown in
It may be appreciated that the above embodiments of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
In some embodiments of the invention, the parameters of one or more filters may change in real time. For example, as the head tracker HT determines changes in the position and/or direction of the headphone, the audio processing unit APU extracts the corresponding set of filter parameters and/or equalization parameters and applies them to the appropriate filters. In such embodiments, there may be a need to effect the changes in parameters with the least impact on the sound quality. We present in this section an overlap-add method can be used to smooth the transition between the different parameters. This method also allows for a more efficient real-time implementation of a Room Reproduction.
After extracting the set of filter and/or equalization parameters for a given position and/or direction of the headphone, the audio processing unit APU transforms the parameters into the frequency domain. The input audio signal AS is segmented into a series of blocks with a length B that are zero padded. The zero padded portion of the block has a length one less than the filter (F−1). Additional zeros are added if necessary so that the length of the Fast Fourier Transform FFT is a power of two. The blocks are transformed into the frequency domain and multiplied with the transformed filter and/or equalization parameters. The processed blocks are then transformed back to the time domain. The tail due to the convolution is now within the zero padded portion of the block and gets added with the next block to form the output signals. Note that there is no additional latency when using this method.
According to an embodiment, the window length and/or the block length may be variable from block to block to smooth the time-varying parameters according to the methods illustrated in
According to an embodiment, the filter unit or the equalizing unit may acquire the set of filter and equalization parameters for a given position and/or direction and perform the signal process according to the methods illustrated in
It may be appreciated that the above embodiments of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
In the various embodiments disclosed herein, HRTFs may be used which have been modified to compensate for timbral coloration, such as to allow for an adjustable degree of timbral coloration and correction therefore. These modified HRTFs may be used in the above-described binaural filter units and binaurally filtering processes, without the need to use the equalizing units and equalizing processes. However, the modified HRTFs disclosed below may be used in the above-described equalizing units and equalizing processes, alone or in combination with their use of the above-described binaural filter units and binaurally filtering processes.
As is known in the art, an HRTF may be expressed as a time domain form or a frequency domain form. Each form may be converted to the other form by an appropriate Fourier transform or inverse Fourier transform. In each form, the HRTF is a function of the position of the source, which may be expressed as a function of azimuth angle (e.g., the angle in the horizontal plane), elevation angle, and radial distance. Simple HRTFs may use just the azimuth angle. Typically, the left and right HRTFs are measured and specified for a plurality of discrete source angles, and values for the HRTFs are interpolated for the other angles. The generation and structure of the modified HRTFs are best illustrated in the frequency domain form. For the sake of simplicity, and without loss of generality, we will use HRTFs that specify the source location with just the azimuth angle (e.g., simple HRTFs) with the understanding the generation of the modified forms can be readily extended to HRTFs that use elevation angle and radial distance to specify the location of the source.
In one exemplary embodiment, a set of modified HRTFs for left and right ears is generated from an initial set, which may be obtained from a library or directly measured in a anechoic chamber. (The HRTFs in the available libraries are also derived from measurements.) The values at one or more azimuth angles of the initial set of HRTFs are replaced with modified values to generate the modified HRTF. The modified values for each such azimuth angle may be generated as follows. The spectral envelope for a plurality k of audio frequency bands is generated. The spectral envelope may be generated as the root-mean-square (RMS) sum of the left and right HRTFs in each frequency band for the given azimuth angle, and may be mathematically denoted as:
RMSSpectrum(k)=sign(HRTFL(k)2+HRTFR(k)2); (F1)
where HRTFL denotes the HRTF for the left ear, HRTFR denotes the HRTF for the right ear, k is the index for the frequency bands, and “sqrt” denotes the square root function. Each frequency band k may be very narrow and cover one frequency value, or may cover several frequency values (currently one frequency value per band is considered best). A timbrally neutral, or “Flat”, set of HRTFs may then be generated from the RMSSpectrum(k) values as follows:
FlatHRTFL(k)=HRTFL(k)/RMSSpectrum(k);
FlatHRTFR(k)=HRTFR(k)/RMSSpectrum(k); (F2)
The RMS values of these FlatHRTFs are equal to 1 in each of the frequency bands k. Since the RMS values are representative of the energy in the bands, their values of unity indicate the lack of perceived coloration. However, the right and left values at each frequency band and source angle are different, and this difference generates the externalization effects.
A particular degree of coloration may be adjusted by generating modified HRTF values in a mathematical form equivalent to:
NewHRTFL(k)=FlatHRTFL(k)*(RMSSpectrum(k))C;
NewHRTFR(k)=FlatHRTFR(k)*(RMSSpectrum(k))C; (F3)
where parameter C is typically in the range of [0, 1], and it specifies the amount of coloration. A mathematically equivalent form of form (F3) is as follows:
NewHRTFL(k)=HRTFL(k)*(RMSSpectrum(k))(C-1);
NewHRTFR(k)=HRTFR(k)*(RMSSpectrum(k))(C-1); (F4)
A value of C=1 will recreate the original HRTFs. It is conceivable that C>1 could be used to enhance the features of an HRTF. The typical trade-off for reduced coloration is that externalization reduces for C<1 and, for small values, localization precision is also reduced. Smoothing of the reapplied RMSSpectrum in Equations (F3) may be done, and may be helpful.
The modified HRTFs may be generated for only a few source angles, such as those going from the front left speaker to the front right speaker, or may be generated for all source angles.
An important frequency band for distinguishing localization effects lies from 2 kHz to 8 kHz. In this band, most normalized sets of HRTFs have dynamic ranges in their spectral envelopes of more than 10 dB over a major span of the source azimuth angle (e.g., over more than 180 degrees). The dynamic ranges of unnormalized sets of HRTFs are the same or greater.
Thus, sets of HRTFs modified according to the present invention can have spectral envelopes in the audio frequency range of 2 kHz to 8 kHz that are equal to or less than 10 dB over a majority of the span of the source azimuth angle (e.g., over more than 180 degrees), and more typically equal to or less than 6 dB.
In considering a pair of angles disposed asymmetrically about the median plane, such as the above source angles of 0 and 30 degrees, the dynamic ranges in the spectral envelopes can both be less than 10 dB in the audio frequency range of 2 kHz to 8 kHz, with at least one of them being less than 6 dB. With lower values of C, such as between C=0.3 to C=0.5, the dynamic ranges in both the spectral envelopes can both be less than 6 dB in the audio frequency range of 2 kHz to 8 kHz, with at least one of them being less than 4 dB, or less than 3 dB.
The modified HRTFs (NewHRTFL and NewHRTFR) may be generated by corresponding modifications of the time-domain forms. Accordingly, it may be appreciated that a set of modified HRTFs may be generated by modifying the set of original HRTFs such that the associated spectral envelope becomes more flat across the frequency domain, and in further embodiments, becomes closer to unity across the frequency domain.
In further embodiments of the above, the modified HRTFs may be further modified to reduce comb effects. Such effects occur when a substantially monoaural signal is filtered with HRTFs that are symmetrical relative to the median plane, such as with simulated front left and right speakers (which occurs frequently in virtual surround sound systems). In essence, the left and right signals substantially cancel one another to create notches of reduced amplitude at certain audio frequencies at each ear. The further modification may include “anti-comb” processing of the modified Head-Related Transfer Functions to counter this effect. In a first “anti-comb” process, slight notches are created in the contralateral HRTF at the frequencies where the amplitude sum of the left and right HRTFs (with ITD) would normally produce a notch of the comb. The slight notches in the contralateral HRTFs reduce the notches in the amplitude sums received by the ears. The processing may be accomplished by multiplying each NewHRTF for each source angle with a comb function having the slight notches. The processing modifies ILDs and should be used with slight notches in order to not introduce significant localization errors. In a second “anti-comb” process the RMSSpectrum is partially amplified or attenuated inversely proportional to the amplitude sum of the left and right HRTFs (with ITD). This process is especially effective in reducing the bass boost that often follows from virtual stereo reproduction since low frequencies in recordings tend to be substantially pretty monoaural. This process does not modify the ILDs, but should be used in moderation. Both “anti-comb” processes, particularly the second one, add coloration to a single source hard panned to any single virtual channel, so there are trade-offs between making typical stereo sound better and making special cases sound worse.
It may be appreciated that this embodiment of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
As described above with reference to
However, a given set of HRTFs does not precisely fit each individual human user, and there are always slight variations between what a given HRTF set provides and what best suits a particular human individual. As such, the above-described straightforward compensation may lead to varying degrees of error in the perceived angular localization for a particular individual. Within the context of head-tracked binaural audio, such varying errors may lead to a perceived movement of the source as a function of head-movements. According to another embodiment of the present invention, the perceived movement of the sources can be compensated for by mapping the current desired source angle (or current measured head angle) to a modified source angle (or modified head angle) that yields a perception closest to the desired direction. The mapping function can be determined from angular localization errors for each direction within the tracked range if these errors are known. As another approach, controls may be provided to the user to allow adjustment to the mapping function so as to minimize the perceived motion of the sources.
Any mapping function known to those with skill in the relevant arts can be used. In one embodiment of the present invention, the mapping function is implemented as a parametrizable cubic spline that can be easily adjusted for a given positional filters database or even for an individual listener. The mapping can be implemented by a set of computer instructions embodied on a tangible computer readable medium that direct a processor in the audio processor unit to generate the modified signal from the input signal and the mapping function. The set of instructions may include further instructions that direct the processor to receive commands from a user to modify the form of the mapping function. The processor may then control the processing of the input surround audio signals by the above-described filters in relation to the modified angle signal.
An embodiment of an exemplary audio processing unit is shown by way of an augmented headset H′ in
It may be appreciated that this embodiment of the invention may be combined with any other embodiment or combination of embodiments of the invention described herein.
The terms and expressions which have been employed herein are used as terms of description and not of limitation, and there is no intention in the use of such terms and expressions of excluding equivalents of the features shown and described, it being recognized that various modifications are possible within the scope of the invention claimed. Moreover, one or more features of one or more embodiments of the invention may be combined with one or more features of other embodiments of the invention without departing from the scope of the invention. While the present invention has been particularly described with respect to the illustrated embodiments, it will be appreciated that various alterations, modifications, adaptations, and equivalent arrangements may be made based on the present disclosure, and are intended to be within the scope of the invention and the appended claims.
| Number | Date | Country | Kind |
|---|---|---|---|
| 08152448 | Mar 2008 | EP | regional |
This application is a division of U.S. application Ser. No. 12/920,578, filed Dec. 17, 2010, which is a U.S. National Stage of PCT/US2009/036575, filed Mar. 9, 2009, which claims priority to European patent application No. EP-08152448.0, filed Mar. 7, 2008, both of which are commonly assigned and incorporated by reference herein for all purposes.
| Number | Name | Date | Kind |
|---|---|---|---|
| 5371799 | Lowe et al. | Dec 1994 | A |
| 6078669 | Maher | Jun 2000 | A |
| 6118875 | Møller et al. | Sep 2000 | A |
| 6370256 | McGrath | Apr 2002 | B1 |
| 6978027 | Dahl et al. | Dec 2005 | B1 |
| 6990205 | Chen | Jan 2006 | B1 |
| 7099482 | Jot et al. | Aug 2006 | B1 |
| 20020067836 | Paranjpe | Jun 2002 | A1 |
| 20020164037 | Sekine | Nov 2002 | A1 |
| 20070154020 | Katayama | Jul 2007 | A1 |
| 20070172086 | Dickins et al. | Jul 2007 | A1 |
| 20090086982 | Kulkarni | Apr 2009 | A1 |
| Number | Date | Country |
|---|---|---|
| WO02098172 | Dec 2002 | WO |
| Entry |
|---|
| Hur, Yoomi, et al. “Efficient Individualization of HRTF Using Critical-Band Based Spectral Cues Control.” Audio Engineering Society Convention 124. Audio Engineering Society, 2008. |
| U.S. Appl. No. 12/920,578, Notice of Allowance mailed on Jun. 18, 2014, 12 pages. |
| U.S. Appl. No. 12/920,578, Non-Final Office Action mailed on Sep. 4, 2013, 15 pages. |
| International Search Report for PCT application PCT/US2009/036575 (Mar. 22, 2010). |
| International Preliminary Report on Patentabilty for PCT application PCT/US2009/036575 (Sep. 7, 2010). |
| Kendall, G., et al., “A Spatial Sound Processor for Loudspeaker and Headphone Reproduction,” AES 8th International Conference, p. 209-221 (May 30, 1990). |
| Smith, J., “Chapter 6: Transfer Function Analysis,” Introduction to Digital Filters, p. 121-129 (2007, W3K Publishing, http://www.w3k.org). |
| J. O. Smith, “Spectral Audio Signal Processing”, Center for Computer Research in Music and Acoustics (CCRMA), Stanford University, Mar. 2007 Version, http://ccrma.stanford.edu/˜jos/sasp/Time—Varying—OLA—Modifications.html. |
| Number | Date | Country | |
|---|---|---|---|
| 20140334650 A1 | Nov 2014 | US |
| Number | Date | Country | |
|---|---|---|---|
| Parent | 12920578 | US | |
| Child | 14341597 | US |
