DETAILED ACTION
Notice of Pre-AIA or AIA Status
The present application, filed on or after March 16, 2013, is being examined under the first inventor to file provisions of the AIA .
Continued Examination Under 37 CFR 1.114
A request for continued examination under 37 CFR 1.114, including the fee set forth in 37 CFR 1.17(e), was filed in this application after final rejection. Since this application is eligible for continued examination under 37 CFR 1.114, and the fee set forth in 37 CFR 1.17(e) has been timely paid, the finality of the previous Office action has been withdrawn pursuant to 37 CFR 1.114. Applicant's submission filed on 3/27/2026 has been entered.
Response to Amendment
Claims 16, 18, and 24 are amended. Claims 34 and 35 are newly added. Claims 16-35 are presented for examination.
Response to Arguments
Rejection under 35 U.S.C. 103
Applicant’s arguments regarding claims 16-34 have been considered but are moot because the new ground of rejection does not rely on any reference applied in the prior rejection of record for any teaching or matter specifically challenged in the argument.
Applicant's arguments regarding claim 35 have been fully considered but they are not persuasive. Applicant argues, “Nowhere do Ohtani or Vallabhan, whether taken alone or in combination, disclose or suggest calculating an adjusted reverberation suppression gain for a given frame based on ‘a value of the adjusted reverberation suppression gain for a previous frame of the input audio signal.’” However, Ohtani recites determining a reverberation suppression gain based on information of suppression gain applied in previous frames. Specifically, Ohtani discloses “[0142] on the basis of the suppression gain G(n-j, f) (where j=1 to m) of the last m frames and the corrected gain G'(n, f) for the nth frame, the correction controller 126 computes an index indicating the slope of the magnitude of the suppression gain G(n, f) in a period up to the nth frame. The correction controller 126 may compute an average gain Gav(n, f) as expressed in Eq. 16 as the index indicating the slope of the magnitude of the suppression gain G(n, f) up to the nth frame, for example.” Thus, Ohtani discloses calculating the adjusted reverberation suppression gain based on the adjusted reverberation suppression gain for a previous frame.
Claim Rejections - 35 USC § 103
The following is a quotation of 35 U.S.C. 103 which forms the basis for all obviousness rejections set forth in this Office action:
A patent for a claimed invention may not be obtained, notwithstanding that the claimed invention is not identically disclosed as set forth in section 102, if the differences between the claimed invention and the prior art are such that the claimed invention as a whole would have been obvious before the effective filing date of the claimed invention to a person having ordinary skill in the art to which the claimed invention pertains. Patentability shall not be negated by the manner in which the invention was made.
Claims 16-18, 31-33, and 35 are rejected under 35 U.S.C. 103 as being unpatentable over Ohtani et al. (US 20130077798 A1; hereinafter referred to as Ohtani) in view of Lester (US 8538038 B1) and Elko et al. (US 20180277137 A1).
Regarding claim 1, Ohtani discloses: a method for reverberation suppression, comprising: receiving an input audio signal, wherein the input audio signal comprises a plurality of frames ([0027] analyzes characteristics of the change over time of an input signal x(t) in a
reverb segment following the end of a segment in which sound is produced, on the basis of the input signal spectrum X(n, f) or the input power spectrum S(n, f) for each frame);
calculating an initial reverberation suppression gain for the input audio signal on a per-frame basis for the plurality of frames ([0073] The gain calculator 123 may use a function like that illustrated by the bold line in FIG. 7 to compute a standard suppression gain Gs(n, f) that corresponds to the signal-to-reverb ratio SRR(n, f) for the frequency number f in the nth frame);
calculating an adjusted reverberation suppression gain on a per-frame basis ([0043] The suppression gain G(n, f) determined for each frame by the suppression controller 120 on the basis of such analysis results becomes a suitable value for suppressing the reverb component included an input signal x(t)) for the plurality of frames of the input audio signal ([0077] the gain corrector 124 is able to determine whether or not the reverb component readily attenuates in the environment where the input signal x(t) was acquired, or in other words, whether or not reverberation suppression is desirable. The gain corrector determines an adjusted reverberation suppression gain.), wherein the adjusted reverberation suppression gain is based on the initial reverberation suppression gain ([0078] the suppression gain G(n, f) corrected by the gain corrector 124 becomes a standard suppression gain Gs(n, f) computed on the basis of the reverb characteristics .gamma.(f). However, the gain corrector 124 may also compute the suppression gain G(n, f) by subtracting a correction value depending on the value of the average change Dav(n) from the standard suppression gain Gs(n, f) in the case where the value of the average change Dav(n) is greater than the first threshold Th1 discussed earlier) and a reverberation intensity detected in the input audio signal ([0040] the analyzer 110 analyzes change in the input signal x(t) over time on the basis of the respective input power spectra S(j, f) (where j=1 to n) of the frames received thus far (step S302). In step S302, the analyzer 110 may also compute an index indicating the decrease per unit time in a reverb segment of the input signal x(t). The analyzer 110 may then output the computed index as an analysis result. The power spectra represents the intensity.),
and wherein, for a given frame of the plurality of frames, calculating the adjusted reverberation suppression gain comprises: calculating the reverberation intensity for the given frame of the input audio signal… ([0040] the analyzer 110 analyzes change in the input signal x(t) over time on the basis of the respective input power spectra S(j, f) (where j=1 to n) of the frames received thus far (step S302). In step S302, the analyzer 110 may also compute an index indicating the decrease per unit time in a reverb segment of the input signal x(t). The analyzer 110 may then output the computed index as an analysis result);
and generating an output audio signal by applying the adjusted reverberation suppression gain to the plurality of frames of the input audio signal on a per-frame basis ([0041] On the basis of the analysis result obtained by the processing in step S302, the suppression controller 120 illustrated by example in FIG. 1 determines a suppression gain G(n, f) to apply to the input signal spectrum X(n, f) of the current frame).
Ohtani does not explicitly, but Lester teaches: calculating an attack phase smoothing time constant and/or a release phase smoothing time constant ([col 18, lines 14-16] With some embodiments, the reverberation power calculation is not a single time constant approximation for both attack and decay constants. The amplitude of the reverberation was experimentally shown to be very highly variable in amplitude with respect to time) for the given frame of the input audio signal ([col 7, lines 17-25] since the reverberation effect is a very commonly perceived audio phenomenon, technique 207 is not typically used for a very long mute length. If the reverberation tail is allowed to trail for a long time, the reverberation may become perceptually apparent to a user and the perceptual illusion of uninterrupted audio is lost. The reverberation extension concealment technique 207 is then best suited for mute intervals of about 30 ms to 100 ms) based on the calculated reverberation intensity for the given frame… ([col 18, lines 29-36] It is a fast attack, slow decay system. The fast attack time constant allows the system to track the peaks of the reverberation channel so that the highest signal level is recorded. Similarly, the slow release time constant allows this highest power level to continue to be held so that if the power calculation occurs during a low energy portion, the high energy is still accounted for in case the energy of the reverberation signal rises again).
Ohtani and Lester are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani to combine the teachings of Lester because doing so would allow for the use of different time constants in order to reduce the possibility of the reverberation being too loud, leading to improved audio quality and flexible reverberation suppression (Lester [col 18, lines 14-27] the reverberation power calculation is not a single time constant approximation for both attack and decay constants. The amplitude of the reverberation was experimentally shown to be very highly variable in amplitude with respect to time. If the power matching calculation occurs when the reverb happens to have little energy, when the reverb is playing and increases back to its stochastic maximum, the reverberation may be perceived as too loud, as well as risking the chance of clipping the digital signal. Of course the opposite is true; the power matching can occur at the reverb's peak leading to a chance that the reverberation concealment will be too soft. It is better to err on the side of the reverb being too soft rather than too loud to minimize the chance of perceptual annoyance).
The combination of Ohtani and Lester does not explicitly, but Elko teaches: and calculating, for the given frame ([0071] For the single-pole recursions in Equations (15) and (16), each attack constant and each decay constant is computed using Equation (18)… where t is the desired time constant in seconds and ƒ.sub.s is the sampling rate of frame processing), the adjusted reverberation suppression ([0069] Processing block 1128 receives the short- and long-time envelope estimates 1127(1) and 1127(2) from processing blocks 1126(1) and 1126(2) and computes a suppression vector 1129 that suppresses the reverberant part of the signal from the main beamformer 1110(1)) gain based on the calculated attack phase smoothing time constant and/or the release phase smoothing time constant for the given frame of the input audio signal… ([0071] The attack and decay constants are chosen to result in recursive envelope estimators whose time response is coincident with the underlying physical quantities being tracked. For the single-pole recursions in Equations (15) and (16), each attack constant and each decay constant is computed using Equation (18)).
Ohtani, Lester, and Elko are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani and Lester to combine the teachings of Elko because doing so would allow for controlling of reverberation suppression based on frequency bands by using envelope estimate, leading to greater flexibility and control for reverberation suppression (Elko [0069] the signal-processing subsystem 1120 has two, independently controllable time delays 1122(1) and 1122(2) for time alignment of the two input audio signals y.sub.1 and y.sub.2 to account for possible differences in the propagation times from the sound source to the two beamformers 1110. In the embodiment of FIG. 11, envelope estimates are generated in frequency subbands that allow for frequency-dependent reverberation suppression).
Regarding claim 17, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Lester further teaches: wherein the attack phase smoothing time constant and/or the release phase smoothing time constant is scaled between two time constants ([col 18, lines 28-29] In order to reduce the chance of the reverb being too loud, a two time constant approximation may be used) based on the calculated reverberation intensity ([col 18, lines 29-36] It is a fast attack, slow decay system. The fast attack time constant allows the system to track the peaks of the reverberation channel so that the highest signal level is recorded. Similarly, the slow release time constant allows this highest power level to continue to be held so that if the power calculation occurs during a low energy portion, the high energy is still accounted for in case the energy of the reverberation signal rises again).
Regarding claim 18, the combination of Ohtani, Lester, and Elko teaches: the method of claim 17. Lester further teaches: wherein the calculated smoothing time constant is an attack phase smoothing time constant if the given frame of the input audio signal corresponds to an attack phase and a release phase smoothing time constant if the given frame of the input audio signal corresponds to a release phase ([col 18, lines 28-36] In order to reduce the chance of the reverb being too loud, a two time constant approximation may be used. It is a fast attack, slow decay system. The fast attack time constant allows the system to track the peaks of the reverberation channel so that the highest signal level is recorded. Similarly, the slow release time constant allows this highest power level to continue to be held so that if the power calculation occurs during a low energy portion, the high energy is still accounted for in case the energy of the reverberation signal rises again).
Regarding claim 31, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Ohtani further teaches: An apparatus configured for implementing the method of claim 16 ([0009] FIG. 7 is a diagram illustrating an embodiment of a reverberation suppression device).
Regarding claim 32, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Ohtani further teaches: a system configured for implementing the method of claim 16 ([0089] The memory 22 stores the operating system of the mobile device 10, as well as an application program by which the processor 21 executes the reverberation suppression process).
Regarding claim 33, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Lester further teaches: one or more non-transitory media having software stored thereon, the software including instructions for controlling one or more devices to perform the method of claim 16 ([Claim 23] A non-transitory computer-readable storage medium storing computer-executable instructions that, when executed, cause a processor to perform…).
Regarding claim 35, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Ohtani further teaches: wherein calculating the adjusted reverberation suppression gain is further based on a value of the adjusted reverberation suppression gain for a previous frame of the input audio signal ([0145] the correction controller 126 may determine that there is low desirability to apply reverberation suppression in the case where the average gain Gav(n, f) is less than or equal to the third threshold Th3, or in other words, in the case where the suppression effect over the past several frames is miniscule to a degree that might not be humanly perceivable. In this case, the correction controller 126 causes the gain corrector 124 to compute a suppression gain G(n, f) with a value smaller than the corrected gain G'(n, f)).
Claims 19 and 34 are rejected under 35 U.S.C. 103 as being unpatentable over Ohtani in view of Lester and Elko, as applied to claims 16-18, 31-33, and 35 above, and further in view of Melanson et al. (US 6104822 A; hereinafter referred to as Melanson).
Regarding claim 19, the combination of Ohtani, Lester, and Elko teaches: the method of claim 17. The combination of Ohtani, Lester, and Elko does not explicitly, but Melanson teaches: wherein the calculated time constant is calculated for a plurality of frequency bands of the input audio signal, and wherein the calculated time constant is smoothed across the plurality of frequency bands ([col 18, lines 10-18] adaptively determine the time constant based on input power level. For low input power, the time constant is made relatively long. For high input power, the time constant is made relatively short. This serves to prevent the reverberation artifact. The determination of adaptive smoothing time constants may be done in individual frequency bands based on power in the individual bands, or it may be done over the entire passband, with only one smoothing constant being used for all frequency bands).
Ohtani, Lester, Elko, and Melanson are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani, Lester, and Elko to combine the teachings of Melanson because doing so would allow for attack/release time constants to be adaptively determined based on input power of an audio signal, leading to better reverberation prevention and smoother audio (Melanson [col 18, lines 4-14] if the time constant is too short, then there is insufficient smoothing and the signal sounds choppy. This choppiness is most apparent at low input signal power levels, for example, during the silence periods in a conversation where there is an air conditioner or a computer fan in the background. A method for dealing with this problem is to adaptively determine the time constant based on input power level. For low input power, the time constant is made relatively long. For high input power, the time constant is made relatively short. This serves to prevent the reverberation artifact).
Regarding claim 34, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Lester further teaches: wherein calculating the attack phase smoothing time constant and/or the release phase smoothing time constant for the at least one frame of the plurality of frames of the input audio signal that is proportional to the calculated reverberation intensity ([col 18, lines 14-21] the reverberation power calculation is not a single time constant approximation for both attack and decay constants. The amplitude of the reverberation was experimentally shown to be very highly variable in amplitude with respect to time. If the power matching calculation occurs when the reverb happens to have little energy, when the reverb is playing and increases back to its stochastic maximum, the reverberation may be perceived as too loud) comprises calculating the attack phase smoothing time constant ([col 18, lines 28-32] In order to reduce the chance of the reverb being too loud, a two time constant approximation may be used. It is a fast attack, slow decay system. The fast attack time constant allows the system to track the peaks of the reverberation channel so that the highest signal level is recorded) that is a continuous value ([col 14-15, lines 61-1] FIG. 11 shows periodic extension concealment subsystem 1100 in accordance with aspects of the disclosure. Subsystem 1100 performs circular buffering. FIG. 12 shows an example of polarity flipping 1202 for periodic extension concealment in accordance with aspects of the disclosure. The output of the periodic extension is flipped in polarity so that when the samples start to get reversed at zero crossing 1201, the instantaneous slope is continuous. The reverberation is scaled with the periodic extension power.) scaled based on the calculated reverberation intensity… ([col 15, lines 31-35] Reverberation signal 1352 is scaled by scaler 1303 so that reverb out 1355 matches the power level of the concealment signal provided by periodic extension concealment subsystem 1100 in concert with power extension power indicator 1353 as processed by gain factor analyzer 1302).
The combination of Ohtani, Lester, and Elko does not explicitly, but Melanson teaches: and calculating the release phase smoothing time constant ([col 17, lines 46-58] The Smoothing Coefficient Generator (2307) determines the time constant of the compressor. Recall that it is generally desirable to have separate attack and release time constants. The Smoothing Coefficient Generator (2307) accomplishes this by comparing the incoming instantaneous power estimate with the current filter state and selecting the appropriate coefficient for attack or release based on this comparison. In addition, it is often desirable to scale the smoothing coefficient depending on the power of the input signal. This is particularly true when very fine frequency band noise reduction algorithms are implemented. In this case it is desirable to have a longer smoothing time constant for low power signals than for high power signals) that is switched between two values based on the calculated reverberation intensity ([col 18, lines 10-14] A method for dealing with this problem is to adaptively determine the time constant based on input power level. For low input power, the time constant is made relatively long. For high input power, the time constant is made relatively short. This serves to prevent the reverberation artifact).
Claims 20-24 are rejected under 35 U.S.C. 103 as being unpatentable over Ohtani in view of Lester and Elko, as applied to claims 16-18, 31-33, and 35 above, and further in view of Klein et al. (US 20060072766 A1; hereinafter referred to as Klein) and Fujita et al. (US 20160360330 A1; hereinafter referred to as Fujita).
Regarding claim 20, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. The combination of Ohtani, Lester, and Elko does not explicitly, but Klein teaches: wherein the adjusted reverberation suppression gain is combined with a calculated second adjusted reverberation suppression gain applied to different frequency bands of the input audio signal… ([0021] identifying the reverberation portion of the signals in each of the bands, establishing a gain function to remove the reverberant portion of the signal in each band, modifying the energy of the audio signals in each band using the gain function, resynthesizing the audio signals from the modified energy signals in each band, combining the processed signals into a single signal, and mixing the processed signals with other signals that mask undesirable processing artifacts).
Ohtani, Lester, Elko, and Klein are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani, Lester, and Elko to combine the teachings of Klein because doing so would allow for different reverberation suppression gain processes to be performed on different frequency bands of an input audio signal, as well as the combination of the results of those processes (Klein [0022] reverberation is removed by analyzing the amount of energy within frequency bands in the input signals in order to identify the main part of the signals and reverberant part of the signals and then to suppress the reverberant part of the signals. In some embodiments, the reverberant part of the signal is identified using delay stability between two or more signals. In some embodiments, the reverberant part of the signal is identified using one or more signals known to contain mostly or only reverberant energy. In some embodiments, the reverberant part of the signal is identified by using a known reverberant impulse response function).
The combination of Ohtani, Lester, Elko, and Klein does not explicitly, but Fujita teaches: based on the amount of room resonance detected in the input audio signal ([0008] the resonant band detecting means is configured to: detect a speaker distortion characteristic using a reference signal of the predetermined sweep signal and the measurement result of the predetermined sweep signal; and detect the resonant band based on the detected speaker distortion characteristic), and wherein calculating the second adjusted reverberation suppression gain comprises: dividing the input audio signal into a plurality of frequency bands ([0025] FIG. 10 is a diagram illustrating control gains of respective frequency bands for each of input levels);
for each frequency band of the plurality of frequency bands, calculating an amount of room resonance present in the input audio signal at the frequency band ([0047] The resonant band detecting unit 116 detects the resonant band at each input level based on the speaker distortion characteristic calculated by the speaker distortion characteristic calculating unit 114);
and calculating the second adjusted reverberation suppression gain for each frequency band based on the amount of room resonance present in the input audio signal at the frequency band ([0011] calculate, for each resonant band, the control gain based on a ratio between an attenuation inclination of a speaker response characteristic at an input level of the predetermined measurement signal and an attenuation inclination of a speaker response characteristic at the reference input level. The control parameter generating means may be configured to calculate, for each resonant band, the control time based on a ratio between the reverberation time at the input level of the predetermined measurement signal and the reverberation time at the reference input level).
Ohtani, Lester, Elko, Klein, and Fujita are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani, Lester, Elko, and Klein to combine the teachings of Fujita because doing so would allow for resonance to be determined and suppressed at different frequency bands in an input audio, leading to reverberation suppression as well (Fujita [0057] the dB conversion unit 118D converts a linear scale value of the calculated ratio R into a decibel scale value, and obtains, as the control parameter (the control gain), the converted ratio R1 (the decibel scale value). The control gain thus obtained provides advantageous effects of suppressing occurrence of resonant sound by making the attenuation inclination a of the speaker response characteristic become equal to or approximately equal to the reference attenuation inclination a in accordance with the input level and thereby attenuating the speaker response characteristic).
Regarding claim 21, the combination of Ohtani, Lester, Elko, Klein, and Fujita teaches: the method of claim 20. Fujita further teaches: wherein calculating the amount of room resonance present in the input audio signal ([0047] The resonant band detecting unit 116 detects the resonant band at each input level based on the speaker distortion characteristic calculated by the speaker distortion characteristic calculating unit 114) at the frequency band comprises calculating a Signal to Reverberant energy Ratio (SRR) for each frequency band ([0045] components other than the sin wave (harmonic distortion and noise) can be obtained, and the speaker distortion characteristic at each input level can be obtained. The speaker distortion characteristic means a ratio (unit: %) indicating how much undesirable components (harmonic distortion and noise) are contained with respect to the component of the reference wave (the measured sweep signal). The noise can be reverberation.).
Regarding claim 22, the combination of Ohtani, Lester, Elko, Klein, and Fujita teaches: the method of claim 21. Fujita further teaches: wherein the amount of room resonance is calculated as greater than 0 for a frequency band of the plurality of frequency bands in response to determining that the SRR for the frequency band is below a threshold ([0050] The reference level setting unit 118A sets, as the reference input level an input level of which the speaker distortion rate is smaller than or equal to the second threshold, within the resonant band detected by the resonant band detecting unit 116, based on the speaker distortion characteristic calculated by the speaker distortion characteristic calculating unit 114).
Regarding claim 23, the combination of Ohtani, Lester, Elko, Klein, and Fujita teaches: the method of claim 21. Fujita further teaches: wherein the amount of room resonance of a frequency band of the plurality of frequency bands is calculated based on an activation function applied to the SRR at the frequency band ([0053] regarding the frequency of 100 Hz, the inclination calculating unit 118B obtains the speaker response characteristic based on the cumulative spectral decay calculated by the cumulative spectral decay calculating unit 112, and calculates an approximation straight line of the obtained speaker response characteristic with a linear regression function).
Regarding claim 24, the combination of Ohtani, Lester, Elko, Klein, and Fujita teaches: the method of claim 20. Fujita further teaches: wherein the second adjusted reverberation suppression gain for each frequency band is based on: a scaled value of the amount of room resonance at each frequency band and for a frame of the plurality of frames of the input audio signal; or a scaled value of an average amount of room resonance at each frequency band averaged across a plurality of frames of the input audio signal ([0059] By converting into the decibel scale value by the dB conversion unit 118D, the ratio R1 becomes -11.53 (dB). The value of -11.53 (AB) is the control gain with respect to the speaker response characteristic of 100 Hz at the input level of 0 dB. By executing similar calculations for the input levels other than 0 dB, the control gain at each input level can be obtained for the resonant band of 100 Hz. By further executing similar calculations for the resonant bands other than 100 Hz, the control gain at each input level can be obtained for each of the resonant bands).
Claims 25-30 are rejected under 35 U.S.C. 103 as being unpatentable over Ohtani in view of Lester and Elko, as applied to claims 16-18, 31-33, and 35 above, and further in view of Klein and Sunohara et al. (US 20160064011 A1; hereinafter referred to as Sunohara).
Regarding claim 25, the combination of Ohtani, Lester, and Elko teaches: the method of claim 16. Ohtani further teaches: wherein calculating the third adjusted reverberation suppression gain comprises: selecting initial reverberation suppression gains for frames of the input audio signal that exceed a threshold ([0075] the gain corrector 124 takes the suppression gain G(n, f) to be the standard suppression gain Gs(n, f) in the case where the value of the average change Dav(n) is greater than the first threshold Th1 discussed earlier).
The combination of Ohtani, Lester, and Elko does not explicitly, but Klein teaches: wherein the adjusted reverberation suppression gain is combined with a calculated third adjusted reverberation suppression gain ([0021] identifying the reverberation portion of the signals in each of the bands, establishing a gain function to remove the reverberant portion of the signal in each band, modifying the energy of the audio signals in each band using the gain function, resynthesizing the audio signals from the modified energy signals in each band, combining the processed signals into a single signal, and mixing the processed signals with other signals that mask undesirable processing artifacts) that adjusts the loudness of the input audio signal based on the effect of the initial reverberation suppression gain on the direct part of the input audio signal ([0021] Reverberation removal by signal processing audio signals is disclosed. In various embodiments, the signal processing may involve calculating the energy of the audio signals in different frequency bands, filtering the energy over time in each band such that reverberant energy is reduced, identifying the reverberation portion of the signals in each of the bands, establishing a gain function to remove the reverberant portion of the signal in each band, modifying the energy of the audio signals in each band using the gain function, resynthesizing the audio signals from the modified energy signals in each band, combining the processed signals into a single signal. Removing reverberant energy can lower the volume of the audio signal.).
Ohtani, Lester, Elko, and Klein are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani, Lester, and Elko to combine the teachings of Klein because doing so would allow for different reverberation suppression gain processes to be performed on different frequency bands of an input audio signal, as well as the combination of the results of those processes (Klein [0022] reverberation is removed by analyzing the amount of energy within frequency bands in the input signals in order to identify the main part of the signals and reverberant part of the signals and then to suppress the reverberant part of the signals. In some embodiments, the reverberant part of the signal is identified using delay stability between two or more signals. In some embodiments, the reverberant part of the signal is identified using one or more signals known to contain mostly or only reverberant energy. In some embodiments, the reverberant part of the signal is identified by using a known reverberant impulse response function).
The combination of Ohtani, Lester, Elko, and Klein does not explicitly, but Sunohara teaches: and estimating statistics associated with the direct part of the input audio signal for the frames of the input audio signal based on the selected initial reverberation suppression gains, wherein the third adjusted reverberation suppression gain is based on the estimated statistics associated with the direct part of the input audio signal ([0047] For the period of time during which the instantaneous value X(k) is larger than the estimated reverberation component R(k) (that is, for the period of time during which it is estimated that the component of direct sound or initial reflected sound is more prominent than the reverberation component), the gain G1 (k) fluctuates as G(k) calculated by the calculation formula (1). By adjusting the gain in this manner, it is possible to suppress the amplitude of the output signal for the period of time during which the late reverberation component is more prominent).
Ohtani, Lester, Elko, Klein, and Sunohara are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani, Lester, Elko, and Klein to combine the teachings of Sunohara because doing so would allow for the reverberation suppression process to use estimated values associated with the direct part of an input audio to determine gain values, leading to more efficient reverberation suppression gain (Sunohara [0019] a reverberation estimation unit that calculates an exponential moving average of the instantaneous value(s) as an estimated reverberation component; a gain derivation unit that derives a gain corresponding to the input signal according to the estimated reverberation component and the instantaneous value(s) for a period of time during which the each instantaneous value is larger than the estimated reverberation component).
Regarding claim 26, the combination of Ohtani, Lester, Elko, Klein, and Sunohara teaches: the method of claim 25. Sunohara further teaches: calculating smoothed initial reverberation suppression gains based on the selected initial reverberation suppression gains ([0019] a smoothing unit that performs a smoothing process on the gain derived by the gain derivation unit; and a gain processing unit that applies the gain after the smoothing process to amplitude adjustment of the input signal), wherein the estimated statistics associated with the direct part of the input audio signal comprise estimated gains applied to the direct part of the input audio signal ([0047] For the period of time during which the instantaneous value X(k) is larger than the estimated reverberation component R(k) (that is, for the period of time during which it is estimated that the component of direct sound or initial reflected sound is more prominent than the reverberation component), the gain G1(k) fluctuates as G(k) calculated by the calculation formula (1). By adjusting the gain in this manner, it is possible to suppress the amplitude of the output signal for the period of time during which the late reverberation component is more prominent), and wherein the estimated gains applied to the direct part of the input audio signal are based on the smoothed initial reverberation suppression gains ([0066] for each of the predetermined number of frequency band components, derives a gain for the input signal based on the estimated reverberation component and the instantaneous value for the period of time during which the instantaneous value is larger than the estimated reverberation component, and fixes the gain to a lower limit for the period of time during which the instantaneous value is smaller than the estimated reverberation component, the smoothing unit performs a smoothing process on the gain derived by the gain derivation unit for each of the predetermined number of frequency band components, and the gain processing unit applies the gains after the smoothing process to the predetermined number of frequency band components).
Regarding claim 27, the combination of Ohtani, Lester, Elko, Klein and Sunohara teaches: the method of claim 26. Elko further teaches: wherein calculating smoothed initial reverberation suppression gains comprises applying a one-pole smoothing to the selected initial reverberation suppression gains ([0059] In one possible implementation, the smoothed average y.sub.s of the coherence estimates {circumflex over (y)} may be generated using a first-order (single-pole) recursive low-pass filter).
Ohtani, Lester, Elko, Klein, and Sunohara are considered analogous in the field of audio processing. Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify the teachings of Ohtani, Lester, Elko, Klein, and Sunohara to further combine the teachings of Elko because doing so would allow for reverberation suppression to use single-pole smoothing to reduce noise and suppress reverberation, leading to improved reverberation suppression (Elko [0060] Processing block 430 multiplies the frequency vector for the time-delayed main beampattern from block 424(1) by the exponentiated average coherence values computed in block 428 to generate a reverberation-suppressed version of the main beampattern in the frequency domain for application to synthesis block 432).
Regarding claim 28, the combination of Ohtani, Lester, Elko, Klein, and Sunohara teaches: the method of claim 26. Sunohara further teaches: wherein the third adjusted reverberation suppression gain is calculated by comparing the estimated gains applied to the direct part of the input audio signal to a target gain. wherein the third adjusted reverberation suppression gain is calculated by comparing the estimated gains applied to the direct part of the input audio signal to a target gain ([0034] At that time, the smoothing unit 7 compares the gain G1(k) at a certain point of time with the immediately preceding gain G2(k)).
Regarding claim 29, the combination of Ohtani, Lester, Elko, Klein, and Sunohara teaches: the method of claim 25. Sunohara further teaches: wherein the estimated statistics associated with the direct part of the input audio signal comprise smoothed loudness levels of the direct part of the audio signal for the frames of the input audio signal based on the selected initial reverberation suppression gains ([0006] a smoothing unit that performs a smoothing process on the gain derived by the gain derivation unit; and a gain processing unit that applies the gain after the smoothing process to amplitude adjustment of the input signal).
Regarding claim 30, the combination of Ohtani, Lester, Elko, Klein, and Sunohara teaches: the method of claim 29. Sunohara further teaches: wherein the third adjusted reverberation suppression gain is calculated by comparing the smoothed loudness levels of the direct part of the input audio signal to a target loudness level ([0034] The smoothing unit 7 then outputs to the gain processing unit 2 the calculated exponential moving average as a gain G2(k) after the smoothing process. At that time, the smoothing unit 7 compares the gain G1 (k) at a certain point of time with the immediately preceding gain G2(k)).
Conclusion
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/NATHAN TENGBUMROONG/Examiner, Art Unit 2654
/HAI PHAN/Supervisory Patent Examiner, Art Unit 2654
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