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 .
Response to Arguments
Applicant's arguments filed 10/17/25 have been fully considered but they are not persuasive.
Regarding Applicant’s first argument:
“the new alleged combination of references still fails to disclose or suggest at least to "perform acoustic echo cancellation (AEC) of the microphone signal ... using first adaptive filters to estimate and cancel feedback that is a result of the environment," and also to "perform acoustic feedback cancellation (AFC) of the echo cancelled microphone signal ... using second adaptive filters to estimate and cancel feedback resulting from application of the reinforced voice signal within the environment." […]
Simply put, neither reference discloses or suggests the two different adaptive filters as claimed, let alone the sequential operation of "first adaptive filters to estimate and cancel feedback" followed by "second adaptive filters to estimate and cancel feedback resulting from application of the reinforced voice signal within the environment."”
The examiner respectfully disagrees. Hetherington and Suzuki both describe using adaptive filters to perform acoustic echo cancellation and to perform acoustic feedback cancellation.
See Hetherington at: col. 2 lns. 25-30: “Due to feedback and echo that accompanies the in car environment, echo and feedback cancellation is performed at the audio processor and thereafter amplified. Here, adaptive filters model the loudspeaker-to-microphone impulse responses that are executed by the audio processor to cancel echo and feedback.” (emphasis added)
Also see col. 6 ln. 65 – col. 7 ln. 9: “At 504, the process models the acoustic environment of the vehicle by modeling the physical paths from the loudspeakers to the microphones and updates the echo canceller coefficients per each reference signal and each microphone. […] The echo canceller coefficients to be updated in 506 may be Finite Impulse Response (FIR) or Infinite Impulse Response (IIR) adaptive filter coefficients per each microphone and each loudspeaker.” (emphasis added)
Further, see col. 4 lns. 18-21: “Because the signals are unique, the echo paths are optimally modeled by the echo & feedback cancellation module 314 that may comprise one or more instances of an adaptive filter, for example, before the signals are post-processed by an optional post processor 316.” (emphasis added)
Still further, see col. 14 lns. 5-8: “The memory 604 and/or 1004 may store information in data structures including, for example, feedback and or echo canceller coefficients that render or estimate echo signal levels.” (emphasis added).
Thus, Hetherington describes using adaptive filters to cancel echo and feedback, where the adaptive filters include feedback canceller coefficients and echo canceller coefficients.
Hetherington doesn’t describe that the echo & feedback cancellation module 314 first performs acoustic echo cancellation of the microphone signal, and then performs acoustic feedback cancellation of the echo cancelled microphone signal to produce a processed microphone signal, which is why the rejection below proposes modifying Hetherington based on the teachings of Suzuki.
As an initial point of clarification, Suzuki does describe using adaptive filters to perform acoustic echo cancellation and to perform acoustic feedback cancellation.
Regarding the adaptive filters to perform acoustic echo cancellation, See Suzuki at col. 6 ln. 1 – col. 7 ln. 35: “First echo and crosstalk canceller 50 is a circuit for estimating and calculating, using an output signal of first acoustic feedback canceller 40, a first interference signal indicative of degrees of first echo 31 caused when a voice output from second loudspeaker 24 comes around and enters into first microphone 21 and first crosstalk 32 caused when a voice of second conversation participant 12 enters into first microphone 21, and for removing the calculated first interference signal from an output signal of first microphone 21. […]
More specifically, first echo and crosstalk canceller 50 includes second transfer function storage circuit 54, second storage circuit 52, second convolution arithmetic operation unit 53, second subtractor 51, and second transfer function update circuit 55.
Second transfer function storage circuit 54 stores a transfer function estimated as a transfer function with respect to first echo 31 and first crosstalk 32 combined to each other.
Second storage circuit 52 stores an output signal of first acoustic feedback canceller 40.
Second convolution arithmetic operation unit 53 performs a convolution with the signal stored in second storage circuit 52 and the transfer function stored in second transfer function storage circuit 54 to generate a first interference signal. For example, second convolution arithmetic operation unit 53 is an N-tap FIR filter for performing a convolution arithmetic operation represented by expression 4 shown below. […]
Second subtractor 51 removes from an output signal of first microphone 21 a calculated first interference signal output from second convolution arithmetic operation unit 53, and outputs an obtained signal as an output signal of first echo and crosstalk canceller 50. For example, second subtractor 51 performs a subtraction represented by expression 5 shown below. […]
Second transfer function update circuit 55 updates the transfer function stored in second transfer function storage circuit 54 based on the output signal of second subtractor 51 and the signal stored in second storage circuit 52. For example, second transfer function update circuit 55 uses an independent component analysis, as represented by expression 6 shown below, to update the transfer function stored in second transfer function storage circuit 54 based on the output signal of second subtractor 51 and the signal stored in second storage circuit 52 such that the output signal of second subtractor 51 and the signal stored in second storage circuit 52 are independent from each other. […]
As described above, second transfer function update circuit 55 performs nonlinear processing using the nonlinear function on the output signal of second subtractor 51, performs a multiplication on an obtained result with the signal stored in second storage circuit 52 and the second step size parameter for controlling the learning speed in estimating the transfer function with respect to first echo 31 and first crosstalk 32 combined to each other, and calculates a second update coefficient. Then, the calculated second update coefficient is added to the transfer function stored in second transfer function storage circuit 54 for updating.” (emphasis added).
Thus, the transfer function coefficients are adaptively updated based on the first echo and first crosstalk, in order to accurately model and remove the echo and crosstalk, which means they are used by adaptive filters to perform echo cancellation.
Regarding the adaptive filters to perform acoustic feedback cancellation of the echo cancelled microphone signal, See Suzuki at col. 7 ln. 56 – col. 9 ln. 27: “Second acoustic feedback canceller 60 is provided between first echo and crosstalk canceller 50 and first loudspeaker 22. Second acoustic feedback canceller 60 is a circuit for estimating and calculating a second acoustic feedback signal indicative of a degree of second acoustic feedback 33 caused when a voice output from first loudspeaker 22 returns and enters into first microphone 21, and for removing the calculated second acoustic feedback signal from an output signal of first microphone 21. In this exemplary embodiment, second acoustic feedback canceller 60 is a circuit for further removing the second acoustic feedback signal from an output signal of first echo and crosstalk canceller 50, in which a calculated first interference signal is removed from the output signal of first microphone 21, and for outputting a signal obtained after the removal to first loudspeaker 22, and is also a digital signal processing circuit for processing digital voice data in a time domain.
More specifically, second acoustic feedback canceller 60 includes third transfer function storage circuit 64, second delay device 66, third storage circuit 62, third convolution arithmetic operation unit 63, third subtractor 61, and third transfer function update circuit 65.
Third transfer function storage circuit 64 stores a transfer function estimated as a transfer function with respect to second acoustic feedback 33.
Second delay device 66 delays an output signal of second acoustic feedback canceller 60.
Third storage circuit 62 stores a signal output from second delay device 66.
Third convolution arithmetic operation unit 63 performs a convolution with the signal stored in third storage circuit 62 and the transfer function stored in third transfer function storage circuit 64 to generate a second acoustic feedback signal. For example, third convolution arithmetic operation unit 63 is an N-tap FIR filter for performing a convolution arithmetic operation represented by expression 7 shown below. […]
Third subtractor 61 removes from an output signal of first echo and crosstalk canceller 50 a calculated second acoustic feedback signal output from third convolution arithmetic operation unit 63, and outputs an obtained signal as an output signal of second acoustic feedback canceller 60. For example, third subtractor 61 performs a subtraction represented by expression 8 shown below. […]
Third transfer function update circuit 65 updates the transfer function stored in third transfer function storage circuit 64 based on the output signal of third subtractor 61 and the signal stored in third storage circuit 62. For example, third transfer function update circuit 65 uses an independent component analysis, as represented by expression 9 shown below, to update the transfer function stored in third transfer function storage circuit 64 based on the output signal of third subtractor 61 and the signal stored in third storage circuit 62 such that the output signal of third subtractor 61 and the signal stored in third storage circuit 62 are independent from each other. […]
As described above, third transfer function update circuit 65 performs nonlinear processing using a nonlinear function on the output signal of third subtractor 61, performs a multiplication on an obtained result with the signal stored in third storage circuit 62 and the third step size parameter for controlling the learning speed in estimating the transfer function with respect to second acoustic feedback 33, and calculates a third update coefficient. Then, the calculated third update coefficient is added to the transfer function stored in third transfer function storage circuit 64 for updating.” (emphasis added).
Thus, the transfer function coefficients are adaptively updated based on the second feedback, in order to accurately model and remove the feedback, which means they are used by adaptive filters to perform feedback cancellation.
Further, as described above, the second acoustic feedback canceller removes the feedback signal from the output signal of the echo and crosstalk canceller. Thus, Suzuki describes the two different adaptive filters as claimed, including the sequential operation of "first adaptive filters to estimate and cancel feedback" followed by "second adaptive filters to estimate and cancel feedback resulting from application of the reinforced voice signal within the environment."
Regarding Applicant’s second argument:
“The Office Action relied on the following portion of Suzuki when making the alleged combination:
The two-way conversation assisting device and the two-way conversation assisting method according to the present disclosure are effective for removing an acoustic noise including not only an echo (resound) but also crosstalk (overhearing), and for amplifying and assisting a two-way conversation.
(Suzuki, col. 2, lines 22-27.)
Yet, this portion of Suzuki does not explain why one would replace a single adaptive module with the sequential operation of "first adaptive filters to estimate and cancel feedback" followed by "second adaptive filters to estimate and cancel feedback resulting from application of the reinforced voice signal within the environment."
Applicant respectfully submits that the alleged combination can only be derived through impermissible hindsight reconstruction based on Applicant's own disclosure. Neither Hetherington nor Suzuki provides any teaching, suggestion, or motivation to modify Hetherington's single-module architecture to include the claimed sequential AEC and AFC adaptive filtering. ”
The examiner respectfully disagrees. In response to applicant's argument that the examiner's conclusion of obviousness is based upon improper hindsight reasoning, it must be recognized that any judgment on obviousness is in a sense necessarily a reconstruction based upon hindsight reasoning. But so long as it takes into account only knowledge which was within the level of ordinary skill at the time the claimed invention was made, and does not include knowledge gleaned only from the applicant's disclosure, such a reconstruction is proper. See In re McLaughlin, 443 F.2d 1392, 170 USPQ 209 (CCPA 1971).
In this case, since Hetherington discloses that the echo & feedback cancellation module 314 removes echo and feedback from the audio signal, as described above, 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 echo & feedback cancellation module 314 such that it performs the sequential operation of “first adaptive filters to estimate and cancel feedback" followed by "second adaptive filters to estimate and cancel feedback resulting from application of the reinforced voice signal within the environment", as suggested by Suzuki, since there a limited number of possible configurations regarding the process of removing feedback and echo from a voice signal, and Suzuki describes his process of removing echo and then removing feedback from the echo-cancelled signal as effective for removing noise including echo, which would have motivated one of ordinary skill in the art to try Suzuki’s technique, in order to determine if it would result in a higher quality echo and feedback cancelled voice signal.
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.
Claim(s) 1, 5, 7, 9, 13, 15, 17, 21, and 23 is/are rejected under 35 U.S.C. 103 as being unpatentable over Hetherington et al. (US 11,348,595), herein “Hetherington”, in view of Suzuki et al. (US 10,542,154), herein “Suzuki”.
RE claims 1, 9, and 17, Hetherington describes a system, method, and non-transitory computer-readable medium comprising instructions for sound signal processing in a vehicle multimedia system, comprising:
loudspeakers configured to reproduce, within an environment, an audio signal from an audio source and a reinforced voice signal (col. 3 lns .51-63: “In this front-to-back reinforcement in which the reinforced signal is conveyed by a single channel and the infotainment comprises music in stereo the four loudspeaker signals, x_1[n], x_4[n] can be represented as: x_1[n]=FL=music left x_2[n]=FR=music right x_3[n]=RL=music left+reinforcement signal x_4[n]=RR=music right+reinforcement signal”);
at least one microphone for detection of a microphone signal, where the microphone signal includes a first voice signal component that corresponds to uttered speech, a second voice signal component that corresponds to the reinforced voice signal as reproduced by the loudspeakers, and an audio signal component corresponding to the audio signal as reproduced by the loudspeakers (col. 2 lns. 39-51: “In vehicle 200 of FIG. 2, the driver and one or more co-driver's (not shown) microphone signals are captured by microphones 202 A and B and then processed and played in rear zone 206 B of the vehicle 200 through loudspeakers 204 C and D. These loudspeakers are provided with front-to-back reinforcing signals 208 C and D. Likewise, one or more rear microphone signals can be captured by microphones 202 C and D, and thereafter processed and converted into audible sound in the front zone 206 A of the vehicle 200 through loudspeakers 204 A and B if there were rear passengers communicating in the vehicle 200. These loudspeakers are provided with back-to-front re-enforcing signals 208 A and B.” Also see col. 3 lns. 27-34: “In other alternative configurations, the entertainment signals and reinforcement signals may be rendered by additional loudspeakers, e.g., tweeters or subwoofer.”); and
a voice processor system configured to
receive the microphone signal from the at least one microphone (col. 4 lns. 1-5 “In FIG. 3 echo and feedback cancellation estimates the impulse response paths {h_j[n]; j=1, . . . , J} given the reference channels {x_j[n]]; j=1, . . . , J} and the microphone signal Y[n], and then subtracts the echo E[n] from microphone signal Y[n].”),
perform acoustic echo cancellation (AEC) of the microphone signal to produce an echo cancelled microphone signal, the AEC using first adaptive filters to estimate and cancel feedback that is a result of the environment (col. 2 lns. 25-30 “Due to feedback and echo that accompanies the in car environment, echo and feedback cancellation is performed at the audio processor and thereafter amplified. Here, adaptive filters model the loudspeaker-to-microphone impulse responses that are executed by the audio processor to cancel echo and feedback.” Also see col. 6 ln. 65 – col. 7 ln. 9: “At 504, the process models the acoustic environment of the vehicle by modeling the physical paths from the loudspeakers to the microphones and updates the echo canceller coefficients per each reference signal and each microphone. […] The echo canceller coefficients to be updated in 506 may be Finite Impulse Response (FIR) or Infinite Impulse Response (IIR) adaptive filter coefficients per each microphone and each loudspeaker.”),
perform acoustic feedback cancellation (AFC) of the microphone signal to produce a processed microphone signal, the AFC using second adaptive filters to estimate and cancel feedback resulting from application of the reinforced voice signal within the environment (col. 2 lns. 25-30 “Due to feedback and echo that accompanies the in car environment, echo and feedback cancellation is performed at the audio processor and thereafter amplified. Here, adaptive filters model the loudspeaker-to-microphone impulse responses that are executed by the audio processor to cancel echo and feedback.” Also see col. 4 lns. 18-21: “Because the signals are unique, the echo paths are optimally modeled by the echo & feedback cancellation module 314 that may comprise one or more instances of an adaptive filter, for example, before the signals are post-processed by an optional post processor 316.” Further, see col. 14 lns. 5-8: “The memory 604 and/or 1004 may store information in data structures including, for example, feedback and or echo canceller coefficients that render or estimate echo signal levels.”),
reinforce the uttered speech in the processed microphone signal to produce the reinforced voice signal (col. 4 lns. 1-13: “In FIG. 3 echo and feedback cancellation estimates the impulse response paths {h_j[n]; j=1, . . . , J} given the reference channels {x_j[n]]; j=1, . . . , J} and the microphone signal Y[n], and then subtracts the echo E[n] from microphone signal Y[n]. In FIG. 3, synthesizer 312, such as a real-time sound synthesizer differentiates the signals by making a non-linear modification to the reinforcing signals and/or by adding an uncorrelated signal to each channel making each of the signals unique.”), and
apply the reinforced voice signal and the audio signal to the loudspeakers for reproduction in the environment (col. 4 lns. 16-18: “In FIG. 3, the signal adder circuit 320 L and R adds the echo cancelled audio processed signal to the infotainment signals.”).
While Hetherington describes performing echo cancellation and feedback cancellation, as described above, Hetherington doesn’t explicitly describe performing acoustic feedback cancellation (AFC) of the echo cancelled microphone signal to produce a processed microphone signal. In other words, Hetherington doesn’t explicitly describe performing echo cancellation first, followed by feedback cancellation of the echo cancelled signal.
However, Suzuki describes performing acoustic feedback cancellation (AFC) of the echo cancelled microphone signal to produce a processed microphone signal (col. 7 ln. 65 – col. 8 ln. 5: “In this exemplary embodiment, second acoustic feedback canceller 60 is a circuit for further removing the second acoustic feedback signal from an output signal of first echo and crosstalk canceller 50, in which a calculated first interference signal is removed from the output signal of first microphone 21, and for outputting a signal obtained after the removal to first loudspeaker 22”).
It would have been obvious before the effective filing date of the claimed invention to include in Hetherington a system and method of performing acoustic feedback cancellation (AFC) of the echo cancelled microphone signal to produce a processed microphone signal, as taught by Suzuki, in order to implement an effective technique to remove all interfering noise and feedback signals from a reinforced voice signal, which results in a high-quality reinforced voice signal (Suzuki col. 2 lns. 22-27).
RE claims 5, 13, and 21 Hetherington describes the system of claim 1, method of claim 9, and medium of claim 17, wherein the environment includes a plurality of sound zones, the at least one microphone includes a first microphone in a first sound zone of the plurality of sound zones and a second microphone in a second sound zone of the plurality of sound zones (col. 2 lns. 34-58: “n FIG. 2 the audio processing system is part of the vehicle 200 and provides entertainment and echo and feedback cancellation. In other systems it is an accessory or a component of a motor vehicle and in other systems part of an audio system used in a room, which may be divided into zones. In vehicle 200 of FIG. 2, the driver and one or more co-driver's (not shown) microphone signals are captured by microphones 202 A and B and then processed and played in rear zone 206 B of the vehicle 200 through loudspeakers 204 C and D. These loudspeakers are provided with front-to-back reinforcing signals 208 C and D. Likewise, one or more rear microphone signals can be captured by microphones 202 C and D, and thereafter processed and converted into audible sound in the front zone 206 A of the vehicle 200 through loudspeakers 204 A and B if there were rear passengers communicating in the vehicle 200. These loudspeakers are provided with back-to-front re-enforcing signals 208 A and B.”), and the voice processor system is further configured to:
reinforce first speech received to the first microphone to produce a first aspect of the reinforced voice signal in the first sound zone (col. 10 ln. 48 – col. 11 ln. 6: “In yet another application, the entertainment post processor 704 may execute synthesis signal processing that modifies the isolated speech from the multiple zones of the vehicle—where the zones comprise a front-left (or driver zone—zone one), front-right (co-driver zone or zone two), rear left (a passenger zone behind the driver or zone three), and rear-rear right (a passenger zone behind the co-driver—zone four). In this application the synthesis signal processing modifies the isolated voices coming from the different zones or alternatively, each of the occupants and modifies the spoken utterances before rending them through selected loudspeakers. The modification may occur by pitch shifting the audio of each zone and then rendering the processed utterances in different zones or combinations of zones out of selected loudspeakers. For example, the front-right zone may be pitch shifted up a half of an octave and projected into the vehicle cabin through rear loudspeaker 306 A, the front-left zone may be pitch shifted up two tenths of an octave and projected into the vehicle cabin through rear loudspeaker 306 B, the rear right zone may be pitch shifted up eight tenths of an octave and projected into the vehicle cabin through front loudspeakers 304 A and B, and the rear-left zone may be pitch shifted up an octave and projected into the vehicle cabin through front and rear loudspeakers 304 A and B and 306 A and B to render an in car harmony”); and
reinforce second speech received to the second microphone to produce a second component of the reinforced voice signal in the second sound zone (col. 10 ln. 48 – col. 11 ln. 6: “In yet another application, the entertainment post processor 704 may execute synthesis signal processing that modifies the isolated speech from the multiple zones of the vehicle—where the zones comprise a front-left (or driver zone—zone one), front-right (co-driver zone or zone two), rear left (a passenger zone behind the driver or zone three), and rear-rear right (a passenger zone behind the co-driver—zone four). In this application the synthesis signal processing modifies the isolated voices coming from the different zones or alternatively, each of the occupants and modifies the spoken utterances before rending them through selected loudspeakers. The modification may occur by pitch shifting the audio of each zone and then rendering the processed utterances in different zones or combinations of zones out of selected loudspeakers. For example, the front-right zone may be pitch shifted up a half of an octave and projected into the vehicle cabin through rear loudspeaker 306 A, the front-left zone may be pitch shifted up two tenths of an octave and projected into the vehicle cabin through rear loudspeaker 306 B, the rear right zone may be pitch shifted up eight tenths of an octave and projected into the vehicle cabin through front loudspeakers 304 A and B, and the rear-left zone may be pitch shifted up an octave and projected into the vehicle cabin through front and rear loudspeakers 304 A and B and 306 A and B to render an in car harmony”).
RE claims 7, 15, and 23, Hetherington teaches the system of claim 1, , method of claim 9, and medium of claim 17, wherein the voice processor system is further configured to apply vocal effects to the reinforced voice signal, the vocal effects including the addition of artificial reverberation (col. 4 lns. 46-58: “Other audio effects such as chorus, flange, and pitch shift may also be generated by synthesizer 312 that enhance the reinforced vocals by rendering a richer, more pleasing and professional sound. Reverberation may also be added by synthesizer 312 to render a sound that simulates an in-car talker's sound (e.g., speech, song, utterances, etc.) being reflected off of a large number of surfaces and simulating a large number of reflections that build up and then decay as if the sound were absorbed by the surfaces in a much larger and/or different space. It can provide the illusion of speaking, singing, or performing in a larger acoustic space such as a night club, concert hall, or cathedral, rather than in the small confines of the vehicle's cabin.”).
Claim(s) 2, 10, and 18 is/are rejected under 35 U.S.C. 103 as being unpatentable over Hetherington in view of Suzuki, as applied to claims 1, 9, and 17 above, and further in view of Christoph et al. (US 2007/0110254), herein “Christoph”.
RE claims 2, 10, and 18, Hetherington in view of Suzuki doesn’t describe, but Christoph describes the system of claim 1, method of claim 9, and medium of claim 17, wherein the AEC is performed using a first subset of the loudspeakers, and the AFC is performed using a second, different subset of the loudspeakers ([0010]: “A dereverberation and feedback compensation system reduces the echo of audio signals received from a first audio device while reducing feedback of speech signals received from a second audio device.” [0020]: “The communication system may also comprise an audio device 9, such as a radio, a CD player, and/or a DVD player. If a back passenger 6 is in dialog with front passenger 5, the conversation may be aided by the communication system by detecting the passengers' utterances through the use of microphones 3 close to the passengers 5 and/or 6 respectively, processing the received signals, and broadcasting the processed signals to the loudspeakers close to the passengers 5 and/or 6 respectively. Because the microphones 3 may also detect audio signals generated by audio device 9 and broadcast by loudspeakers 2, the communication system may also process these signals.” [0029]: “After estimating the impulse response h.sub.i(n) between loudspeaker 2 and microphone 3, the feedback components of an audio signal from a second audio device 15, such as a loudspeaker that transmits a passenger's verbal utterances (e.g., a speech signal), may also be estimated.” [0030]: “Where the audio signals x.sub.i(n) may be broadcasted by all of the loudspeakers, the output signals of the communication system y.sub.i(n) may be transmitted by the loudspeaker(s) close to the listening communication partner only”).
It would have been obvious before the effective filing date of the claimed invention to include in Hetherington in view of Suzuki a system and method wherein the AEC is performed using a first subset of the loudspeakers, and the AFC is performed using a second, different subset of the loudspeakers, as taught by Christoph, in order to reduce the processing requirements associated with the echo cancellation and feedback cancellation processes, without degrading system performance, by only performing echo and/or feedback cancellation on the loudspeakers that are outputting audio signals and/or reinforced voice signals.
Claim(s) 3, 11 and 19 is/are rejected under 35 U.S.C. 103 as being unpatentable over Hetherington in view of Suzuki, as applied to claims 1, 9, and 17 above, and further in view of Li (US 11,211,061), herein “Li”.
RE claims 3, 11, and 19, Hetherington describes the system of claim 1, method of claim 9, and medium of claim 17, wherein the voice processor system is further configured to perform automatic speech recognition (ASR) on the processed microphone signal (col. 9 ln. 34 – col. 10 ln. 47: “In FIG. 7, an entertainment post processing system 704 may deliver entertainment, services, or a grammar-based or a natural language-based automatic speech recognition (ASR). Since the in-car entertainment communication system isolates speech and/or other content delivered in the vehicle 200 a parallel architecture through a tree-based ASR structure may execute speech recognition of a limited vocabulary size through one or more processing branches (or paths) when resources are limited or through an unlimited vocabulary through a natural language vocabulary that can include a dictionary in one or more or all processing branches or a combination of ASRs.”).
Hetherington in view of Suzuki doesn’t explicitly describe a system or method wherein the voice processor system is further configured to perform automatic speech recognition (ASR) on the processed microphone signal to receive commands to control the voice processor system.
However, Li describes a system and method wherein the voice processor system is further configured to perform automatic speech recognition (ASR) on the processed microphone signal to receive commands to control the voice processor system (col. 16 lns. 22-47: “The ASR system 2000 uses various modules to perform tasks such as audio capture and import and to provide prompt services. ASR services may be launched through a Persistent Publish/Subscribe (PPS) service when the user activates Push-to-Talk (PPT) functionally, for examples, by touching and activating a PTT button or tab on a human-machine interface (HMI) interface displayed on the touchscreen 1836. The audio module 2010 include an audio capture module that detects speech commands, including the beginning and end of sentences, and forwards the audio stream to the speech recognition modules 2015.” […] “For example, the speech recognition module 2015 would take the utterance “search media for Hero” and create a results structure” Also see col. 17 lns. 1-7: “Applications such as Media Player and Navigation may subscribe to PPS objects for changes. For example, if the user activates PTT and says “play Arcade Fire”, the speech recognition modules 2015 parse the speech command. The Media conversation module 2020 then activates the media engine, causing tracks from the desired artist to play.”).
It would have been obvious before the effective filing date of the claimed invention to include in Hetherington in view of Suzuki a system or method wherein the voice processor system is further configured to perform automatic speech recognition (ASR) on the processed microphone signal to receive commands to control the voice processor system, as taught by Li, in order to enable users to interact with and control the voice processor system using voice commands, which improves the user experience by making user input easier.
Claim(s) 4, 12, and 20 is/are rejected under 35 U.S.C. 103 as being unpatentable over Hetherington in view of Suzuki, and further in view of Li, as applied to claims 3, 11, and 19, and further in view of Thoresz et al. (US 2020/0314486), herein “Thoresz”.
RE claims 4, 12, and 20, Hetherington in view of Suzuki and further in view of Li doesn’t teach, but Thoresz teaches the system of claim 3, method of claim 11, and medium of claim 19, wherein the commands include one or more of to: skip a song, repeat a song, repeat a section, adjust vocal effects and/or multichannel effects, add a user for voice reinforcement, turn off a user for voice reinforcement, turn off voice reinforcement for all users, or to request to turn on a voice processor mode to send uttered speech from one user to another of the users ([0080]: “The playback control region 133d can include selectable (e.g., via touch input and/or via a cursor or another suitable selector) icons to cause one or more playback devices in a selected playback zone or zone group to perform playback actions such as, for example, play or pause, fast forward, rewind, skip to next, skip to previous, enter/exit shuffle mode, enter/exit repeat mode, enter/exit cross fade mode, etc. The playback control region 133d may also include selectable icons to modify equalization settings, playback volume, and/or other suitable playback actions.” [0081]: “in some embodiments the control device 130a is configured as an NMD (e.g., one of the NMDs 120), receiving voice commands and other sounds via the one or more microphones 135.”).
It would have been obvious before the effective filing date of the claimed invention to include in Hetherington in view of Suzuki in view of Li a system or method wherein the commands include one or more of to: skip a song, repeat a song, repeat a section, adjust vocal effects and/or multichannel effects, add a user for voice reinforcement, turn off a user for voice reinforcement, turn off voice reinforcement for all users, or to request to turn on a voice processor mode to send uttered speech from one user to another of the users, as taught by Thoresz, in order to enable the user to easily control the entertainment system via voice commands, which improves the user experience by giving a wider range of commands that that the system can recognize.
Claim(s) 6, 14, and 22 is/are rejected under 35 U.S.C. 103 as being unpatentable over Hetherington in view of Suzuki, as applied to claims 5, 13, and 21 above, and further in view of Roberge (US 2015/0255088), herein “Roberge”.
RE claims 6, 14, and 21, Hetherington teaches the system of claim 5, method of claim 13, and medium of claim 21, wherein the first speech is received from a first singer, the second speech is received from a second singer (col. 10 ln. 62 – col. 11 ln. 6: “For example, the front-right zone may be pitch shifted up a half of an octave and projected into the vehicle cabin through rear loudspeaker 306 A, the front-left zone may be pitch shifted up two tenths of an octave and projected into the vehicle cabin through rear loudspeaker 306 B, the rear right zone may be pitch shifted up eight tenths of an octave and projected into the vehicle cabin through front loudspeakers 304 A and B, and the rear-left zone may be pitch shifted up an octave and projected into the vehicle cabin through front and rear loudspeakers 304 A and B and 306 A and B to render an in car harmony.” Also see col. 11 lns. 20-26: “In some in-car entertainment communication system that may include the functions shown in FIG. 7, the speech recognized lyrics are stored locally in memory or in a cloud-based storage as metadata with the original music or the processed music (with and/or without the original vocal tracks) so that the processing need only occur once a music track or segment is played. When the content is rendered, the original music or the track without vocals may be rendered in the vehicle cabin through loudspeaker 302 A and B and 306 A and B. The lyrics may be displayed on one or more heads-up display for each of the occupants or transmitted to the occupants wireless or mobile devices. In these alternatives a carpool karaoke system is rendered.”)
Hetherington in view of Suzuki doesn’t teach, but Roberge teaches the voice processor system is further configured to:
perform an evaluation of pitch of each of the first speech and the second speech against a reference pitch ([0003]: “There is further provided a system for scoring a singer, comprising a processing module determining notes duration and pitch of a melody of a reference song and notes duration and pitch of a melody of the singer's rendering of the reference song; and a scoring processing module comparing the notes duration and the pitch of the melody of the reference song and the notes and the pitch of the melody of the singer's rendering of the reference song.”); and
identify whether the first singer or the second singer provided a performance closest to the reference pitch ([0037]: “ In FIG. 4, a fixed trip set point T.sub.a is shown. In practice, the trip set point T.sub.a is set at half the value of the energy of the first peak, so as to adapt to amplitude variations of the input signal. Hence, the envelope of a first singer singing louder than a second singer stops at the same point as the envelope of a second singer singing in a lower voice, which allows an equitable scoring between the different users.” [0041]: “Thus, the processing module 100 generates a set R of N parameters, defining the melody (notes) of a song, in terms of pitch and duration (i.e. time envelope). It serves as a reference when assessing the quality of the song as sung by a karaoke user.”).
It would have been obvious before the effective filing date of the claimed invention to include in Hetherington in view of Suzuki a system and method wherein the voice processor system is further configured to: perform an evaluation of pitch of each of the first speech and the second speech against a reference pitch; and identify whether the first singer or the second singer provided a performance closest to the reference pitch, as taught by Roberge, which increases interest in the karaoke system by using it as a teaching system to improve a person’s singing skills, or by using it as a game or competition, where users compete to see who is the best singer.
Claim(s) 8, 16, and 24 is/are rejected under 35 U.S.C. 103 as being unpatentable over Hetherington in view of Suzuki, as applied to claims 7, 15, and 23 above, and further in view of Buck et al. (US 10,056,092), herein “Buck”.
RE claims 8, 16, and 24, Hetherington in view of Suzuki doesn’t teach, but Buck describes the system of claim 7, method of claim 15, and medium of claim 23, wherein the voice processor system is further configured to
increase a step size of adjustment of the second adaptive filters responsive to detection of reverberation in the processed microphone signal (col. 4 lns. 26-33: “controlling a step size of the adaptation AIC filter, dynamically adjusting a length of the FIR filter corresponding to a reverberation time and/or the ratio of early and late residual interference, the reference signal comprises a loudspeaker signal and the RSV estimate is applied for residual echo suppression, and/or the reference signal comprises a microphone signal and the second component corresponding to late RSV is used for dereverberation.” Also see col. 10 lns. 42-62: “The optimal step size for adaptation can be computed as: [see equation 20] Control of the step size μ(k) enables good convergence behavior. In embodiments, one aim is to get a better estimate of the residual of Φϵϵ(k) and thus, to a better convergence of the AIC filter. Generally, the dynamic step size enables the filter to adapt (and converge) quickly when Φϵϵ(k) is large (i.e., the filter is not well converged) and also ensures that the filter adapts slowly when Φϵϵ(k) is small (i.e., it prevents the filter from losing good convergence). A benefit of modelling the late reverb here is to get an estimate for the early residual PSD that is not affected by the late residual PSD. As a consequence, the AIC-step-size will be small even if there is significant late reverberant energy. This improves the convergence of the AIC-filter compared to conventional AIC control methods.” Further, see col. 11 lns. 31-46: “FIG. 7A shows an illustrative sequence of steps for residual interference suppression processing. In step 700, an AIC output signal having residual interference is received. In step 752, the residual interference is estimated including estimating a power spectral density of a first part of the residual interference corresponding to early reverberation and a second part corresponding to late reverberation. In one embodiment, the first part is estimated using a real-valued FIR filter operating on a power spectral density (PSD) of a reference signal and the second part is estimated using an exponential decay over time corresponding to a reverberation time using the PSD of the reference signal. In step 754, filter parameters can be adjusted. In step 756, a filter step size can be optimized for filter convergence. In step 758, a filter length can be adjusted based upon the reverberation time.”); and
decrease the step size of the adjustment of the second adaptive filters responsive to a lack of reverberation in the processed microphone signal (col. 4 lns. 26-33: “controlling a step size of the adaptation AIC filter, dynamically adjusting a length of the FIR filter corresponding to a reverberation time and/or the ratio of early and late residual interference, the reference signal comprises a loudspeaker signal and the RSV estimate is applied for residual echo suppression, and/or the reference signal comprises a microphone signal and the second component corresponding to late RSV is used for dereverberation.” Also see col. 10 lns. 42-62: “The optimal step size for adaptation can be computed as: [see equation 20] Control of the step size μ(k) enables good convergence behavior. In embodiments, one aim is to get a better estimate of the residual of Φϵϵ(k) and thus, to a better convergence of the AIC filter. Generally, the dynamic step size enables the filter to adapt (and converge) quickly when Φϵϵ(k) is large (i.e., the filter is not well converged) and also ensures that the filter adapts slowly when Φϵϵ(k) is small (i.e., it prevents the filter from losing good convergence). A benefit of modelling the late reverb here is to get an estimate for the early residual PSD that is not affected by the late residual PSD. As a consequence, the AIC-step-size will be small even if there is significant late reverberant energy. This improves the convergence of the AIC-filter compared to conventional AIC control methods.” Further, see col. 11 lns. 31-46: “FIG. 7A shows an illustrative sequence of steps for residual interference suppression processing. In step 700, an AIC output signal having residual interference is received. In step 752, the residual interference is estimated including estimating a power spectral density of a first part of the residual interference corresponding to early reverberation and a second part corresponding to late reverberation. In one embodiment, the first part is estimated using a real-valued FIR filter operating on a power spectral density (PSD) of a reference signal and the second part is estimated using an exponential decay over time corresponding to a reverberation time using the PSD of the reference signal. In step 754, filter parameters can be adjusted. In step 756, a filter step size can be optimized for filter convergence. In step 758, a filter length can be adjusted based upon the reverberation time.”).
It would have been obvious before the effective filing date of the claimed invention to include in Hetherington in view of Suzuki a system and method wherein the voice processor system is further configured to increase a step size of adjustment of the second adaptive filters responsive to detection of reverberation in the processed microphone signal; and decrease the step size of the adjustment of the second adaptive filters responsive to a lack of reverberation in the processed microphone signal, as taught by Buck, in order to improve the echo cancellation process by quickly removing reverberation when it is detected, while also maintaining good echo cancellation performance when reverberation is not detected, through the use of a dynamic step size in the filter, which results in a better user experience.
Conclusion
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