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 .
Claim Rejections - 35 USC § 103
1. In the event the determination of the status of the application as subject to AIA 35 U.S.C. 102 and 103 (or as subject to pre-AIA 35 U.S.C. 102 and 103) is incorrect, any correction of the statutory basis (i.e., changing from AIA to pre-AIA ) for the rejection will not be considered a new ground of rejection if the prior art relied upon, and the rationale supporting the rejection, would be the same under either status.
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.
The factual inquiries for establishing a background for determining obviousness under 35 U.S.C. 103 are summarized as follows:
1. Determining the scope and contents of the prior art.
2. Ascertaining the differences between the prior art and the claims at issue.
3. Resolving the level of ordinary skill in the pertinent art.
4. Considering objective evidence present in the application indicating obviousness or nonobviousness.
2. Claim(s) 1, 11, 20 are rejected under 35 U.S.C. 103 as being unpatentable over Master et al. (US 2017/0041724) in view of Vitaladevuni et al. (US 9,704,478) in further view of Maher et al. (US 2013/0216071).
Regarding claim 1, Master teaches a playback device comprising: at least one audio transducer; at least one microphone; a network interface; at least one processor; a housing carrying the at least one audio transducer, the at least one microphone, the network interface, and the at least one processor; and at least one non-transitory computer-readable medium comprising program instructions that are executable by the at least one processor such that the playback device is configured to: receive, via the network interface, a first audio signal representing first audio content; play back the first audio content via the at least one audio transducer (see fig. 1,-2A, ¶ 0021, 0025-0027, 0030, 0054. Sound played through speakers of a device having a microphone. The inputting an original audio signal to a processing component and an output stage of the device for playback through the speakers, receiving playback sound output from the speakers through the microphone to generate a microphone signal and inputting the microphone signal into the processing component to calibrate the speakers for optimal playback of the original audio signal, wherein the processing component is configured to compare the original audio signal to the microphone signal and correct the microphone signal by one or more audio processing functions.).
Master discloses a speaker device which has a circuit board, microphone, speaker and is able to receive a audio signal wirelessly from an audio input device.
However Master does not disclose during playback of the first audio content, capture a second audio signal via the at least one microphone; apply an adaptive acoustic echo canceller to the second audio signal using the first audio signal as a reference; estimate environment acoustic characteristics based on output of the adaptive acoustic echo canceller; determine a calibration that at least partially offsets the estimated environment acoustic characteristics; and apply the determined calibration during playback of second audio content.
Vitaladevuni teaches during playback of the first audio content, capture a second audio signal via the at least one microphone; apply an adaptive acoustic echo canceller to the second audio signal using the first audio signal as a reference; estimate environment acoustic characteristics based on output of the adaptive acoustic echo canceller (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
The combination of Vitaladevuni to Master provides AEC filtering the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal.
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Maher teaches determine a calibration that at least partially offsets the estimated environment acoustic characteristics; and apply the determined calibration during playback of second audio content (see fig. 2. ¶ 0027. Calculation of the optimal equalization parameters is performed in a way that accommodates the transfer function of the microphone. Once the measurements and the calculations are complete, the optimal equalization parameters can be made available to the digital signal processor 102 which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers (208). This can include, for example, equalization for room reflections, cancellation of crosstalk from multiple channels, and/or the like. When additional audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback.).
The combination of Maher to Master and Vitaladevuni provides optimal equalization parameters that can be made available to the digital signal processor which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers.
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Vitaladevuni to incorporate audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback. The modification provides for calibration that at least partially offsets the estimated environment acoustic characteristics and apply the determined calibration during playback of second audio content.
Regarding claim 11, Master teaches a system comprising: a playback device, the playback device comprising: at least one audio transducer; at least one microphone; a network interface; a housing carrying the at least one audio transducer, the at least one microphone, and the network interface; and at least one processor; at least one non-transitory computer-readable medium comprising program instructions that are executable by the at least one processor such that the system is configured to: receive, via the network interface, a first audio signal representing first audio content; play back the first audio content via the at least one audio transducer (see fig. 1,-2A, ¶ 0021, 0025-0027, 0030, 0054. Sound played through speakers of a device having a microphone. The inputting an original audio signal to a processing component and an output stage of the device for playback through the speakers, receiving playback sound output from the speakers through the microphone to generate a microphone signal and inputting the microphone signal into the processing component to calibrate the speakers for optimal playback of the original audio signal, wherein the processing component is configured to compare the original audio signal to the microphone signal and correct the microphone signal by one or more audio processing functions.).
Master discloses a speaker device which has a circuit board, microphone, speaker and is able to receive an audio signal wirelessly from an audio input device.
However Master does not disclose during playback of the first audio content, capture a second audio signal via the at least one microphone; apply an adaptive acoustic echo canceller to the second audio signal using the first audio signal as a reference; estimate environment acoustic characteristics based on output of the adaptive acoustic echo canceller; determine a calibration that at least partially offsets the estimated environment acoustic characteristics; and apply the determined calibration during playback of second audio content.
Vitaladevuni teaches during playback of the first audio content, capture a second audio signal via the at least one microphone; apply an adaptive acoustic echo canceller to the second audio signal using the first audio signal as a reference; estimate environment acoustic characteristics based on output of the adaptive acoustic echo canceller (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
The combination of Vitaladevuni to Master provides AEC filtering the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal.
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Maher teaches determine a calibration that at least partially offsets the estimated environment acoustic characteristics; and apply the determined calibration during playback of second audio content (see fig. 2. ¶ 0027. Calculation of the optimal equalization parameters is performed in a way that accommodates the transfer function of the microphone. Once the measurements and the calculations are complete, the optimal equalization parameters can be made available to the digital signal processor 102 which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers (208). This can include, for example, equalization for room reflections, cancellation of crosstalk from multiple channels, and/or the like. When additional audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback.).
The combination of Maher to Master and Vitaladevuni provides optimal equalization parameters that can be made available to the digital signal processor which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers.
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Vitaladevuni to incorporate audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback. The modification provides for calibration that at least partially offsets the estimated environment acoustic characteristics and apply the determined calibration during playback of second audio content.
Regarding claim 20, Master teaches a method to be performed by a playback device, the method comprising: receiving, via a network interface, a first audio signal representing first audio content, wherein the playback device comprises a housing carrying at least one audio transducer, at least one microphone, and the network interface; playing back the first audio content via the at least one audio transducer (see fig. 1,-2A, ¶ 0021, 0025-0027, 0030, 0054. Sound played through speakers of a device having a microphone. The inputting an original audio signal to a processing component and an output stage of the device for playback through the speakers, receiving playback sound output from the speakers through the microphone to generate a microphone signal and inputting the microphone signal into the processing component to calibrate the speakers for optimal playback of the original audio signal, wherein the processing component is configured to compare the original audio signal to the microphone signal and correct the microphone signal by one or more audio processing functions.).
Master discloses a speaker device which has a circuit board, microphone, speaker and is able to receive an audio signal wirelessly from an audio input device.
However Master does not disclose while playing back the first audio content, capturing a second audio signal via the at least one microphone; applying an adaptive acoustic echo canceller to the second audio signal using the first audio signal as a reference; estimating environment acoustic characteristics based on output of the adaptive acoustic echo canceller; determining a calibration that at least partially offsets the estimated environment acoustic characteristics; and applying the determined calibration during playback of second audio content.
Vitaladevuni teaches while playing back the first audio content, capturing a second audio signal via the at least one microphone; applying an adaptive acoustic echo canceller to the second audio signal using the first audio signal as a reference; estimating environment acoustic characteristics based on output of the adaptive acoustic echo canceller (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
The combination of Vitaladevuni to Master provides AEC filtering the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal.
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Maher teaches determining a calibration that at least partially offsets the estimated environment acoustic characteristics; and applying the determined calibration during playback of second audio content (see fig. 2. ¶ 0027. Calculation of the optimal equalization parameters is performed in a way that accommodates the transfer function of the microphone. Once the measurements and the calculations are complete, the optimal equalization parameters can be made available to the digital signal processor 102 which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers (208). This can include, for example, equalization for room reflections, cancellation of crosstalk from multiple channels, and/or the like. When additional audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback.).
The combination of Maher to Master and Vitaladevuni provides optimal equalization parameters that can be made available to the digital signal processor which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers.
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Vitaladevuni to incorporate audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback. The modification provides for calibration that at least partially offsets the estimated environment acoustic characteristics and apply the determined calibration during playback of second audio content.
3. Claim(s) 2, 3, 4, 12, 13 are rejected under 35 U.S.C. 103 as being unpatentable over Master et al. (US 2017/0041724) in view of Vitaladevuni et al. (US 9,704,478) in further view of Maher et al. (US 2013/0216071) in further view of Calatayud (US 2015/0091691).
Regarding claim 2, Master, Vitaladevuni and Maher do not teach the playback device of claim 1, further comprising an audio pipeline comprising a digital to analog converter, a signal processor, and an audio amplifier, and wherein the program instructions that are executable by the at least one processor such that the playback device is configured to play back the first audio content comprise program instructions that are executable by the at least one processor such that the playback device is configured to: apply signal processing to the first audio signal; convert the signal-processed first audio signal to an analog signal; amplify the analog signal; and play back the amplified analog signal via the at least one audio transducer.
Calatayud teaches further comprising an audio pipeline comprising a digital to analog converter, a signal processor, and an audio amplifier, and wherein the program instructions that are executable by the at least one processor such that the playback device is configured to play back the first audio content comprise program instructions that are executable by the at least one processor such that the playback device is configured to: apply signal processing to the first audio signal; convert the signal-processed first audio signal to an analog signal; amplify the analog signal; and play back the amplified analog signal via the at least one audio transducer (see fig. 4, ¶ 0068-0069. The device having a DAC and converts the audio signal to analog and processing the audio signal to the audio amplifier for a playback.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Maher to incorporate apply signal processing to the first audio signal; convert the signal-processed first audio signal to an analog signal, amplify the analog signal and play back the amplified analog signal via the at least one audio transducer. The modification provides converting an audio signal to analog and amplify the analog signal and play back the amplified analog signal via the at least one audio transduce.
Regarding claim 3, Master and Maher do not teach the playback device of claim 2, wherein the program instructions that are executable by the at least one processor such that the playback device is configured to apply adaptive acoustic echo canceller to the second audio signal using the first audio signal as the reference comprise program instructions that are executable by the at least one processor such that the playback device is configured to: obtain the first audio signal as the reference from a tap in the audio pipeline after the signal processing is applied.
Vitaladevuni teaches wherein the program instructions that are executable by the at least one processor such that the playback device is configured to apply adaptive acoustic echo canceller to the second audio signal using the first audio signal as the reference comprise program instructions that are executable by the at least one processor such that the playback device is configured to: obtain the first audio signal as the reference from a tap in the audio pipeline after the signal processing is applied (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Calatayud teaches an audio pipeline (see fig. 4, ¶ 0068-0069. The device having a DAC and converts the audio signal to analog and processing the audio signal to the audio amplifier for a playback.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Vitaladevuni and Maher to incorporate apply signal processing to the first audio signal, convert the signal-processed first audio signal to an analog signal, amplify the analog signal and play back the amplified analog signal via the at least one audio transducer. The modification provides converting an audio signal to analog and amplify the analog signal and play back the amplified analog signal via the at least one audio transduce
Regarding claim 4, Master, Maher and Calatayud do not teach the playback device of claim 2, wherein the program instructions that are executable by the at least one processor such that the playback device is configured to apply adaptive acoustic echo canceller to the second audio signal using the first audio signal as the reference comprise program instructions that are executable by the at least one processor such that the playback device is configured to: obtain the first audio signal as the reference from a tap in the audio pipeline before the signal processing is applied; and apply the signal processing to the reference.
Vitaladevuni teaches wherein the program instructions that are executable by the at least one processor such that the playback device is configured to apply adaptive acoustic echo canceller to the second audio signal using the first audio signal as the reference comprise program instructions that are executable by the at least one processor such that the playback device is configured to: obtain the first audio signal as the reference from a tap in the audio pipeline before the signal processing is applied; and apply the signal processing to the reference (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Maher and Calatayud to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Regarding claim 12, Master, Vitaladevuni and Maher do not teach the system of claim 11, further comprising an audio pipeline comprising a digital to analog converter, a signal processor, and an audio amplifier, and wherein the program instructions that are executable by the at least one processor such that the system is configured to play back the first audio content comprise program instructions that are executable by the at least one processor such that the system is configured to: apply signal processing to the first audio signal; convert the signal-processed first audio signal to an analog signal; amplify the analog signal; and play back the amplified analog signal via the at least one audio transducer.
Calatayud teaches further comprising an audio pipeline comprising a digital to analog converter, a signal processor, and an audio amplifier, and wherein the program instructions that are executable by the at least one processor such that the system is configured to play back the first audio content comprise program instructions that are executable by the at least one processor such that the system is configured to: apply signal processing to the first audio signal; convert the signal-processed first audio signal to an analog signal; amplify the analog signal; and play back the amplified analog signal via the at least one audio transducer (see fig. 4, ¶ 0068-0069. The device having a DAC and converts the audio signal to analog and processing the audio signal to the audio amplifier for a playback.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Vitaladevuni and Maher to incorporate apply signal processing to the first audio signal; convert the signal-processed first audio signal to an analog signal, amplify the analog signal and play back the amplified analog signal via the at least one audio transducer. The modification provides converting an audio signal to analog and amplify the analog signal and play back the amplified analog signal via the at least one audio transduce
Regarding claim 13, Master, Maher and Calatayud do not teach the system of claim 12, wherein the program instructions that are executable by the at least one processor such that the system is configured to apply adaptive acoustic echo canceller to the second audio signal using the first audio signal as the reference comprise program instructions that are executable by the at least one processor such that the system is configured to: obtain the first audio signal as the reference from a tap in the audio pipeline before the signal processing is applied; and apply the signal processing to the reference.
Vitaladevuni teaches system of claim 12, wherein the program instructions that are executable by the at least one processor such that the system is configured to apply adaptive acoustic echo canceller to the second audio signal using the first audio signal as the reference comprise program instructions that are executable by the at least one processor such that the system is configured to: obtain the first audio signal as the reference from a tap in the audio pipeline before the signal processing is applied; and apply the signal processing to the reference (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Maher and Calatayud to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
4. Claim(s) 5, 14 are rejected under 35 U.S.C. 103 as being unpatentable over Master et al. (US 2017/0041724) in view of Vitaladevuni et al. (US 9,704,478) in further view of Maher et al. (US 2013/0216071) in further view of Goodson et al. (US 8,731,183).
Regarding claim 5, Master, Vitaladevuni and Maher do not teach the playback device of claim 1, wherein the output of the adaptive acoustic echo canceller comprises a transfer function representing a difference between the first audio signal and the second audio signal after the adaptive acoustic echo canceller has converged.
Goodson teaches wherein the output of the adaptive acoustic echo canceller comprises a transfer function representing a difference between the first audio signal and the second audio signal after the adaptive acoustic echo canceller has converged (see fig. 3, col. 1, lines 44-65, col. 3, lines 61-col. 4, lines 13. Cancellation relation comprises H1-H2.sub.N-H3. H1 represents the transfer function of the first voltage divider. H2.sub.N represents the transfer function of the Nth second voltage divider and H3 represents the transfer function of the second echo canceller as a digital echo canceller after the taps have converged.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Vitaladevuni and Maher to incorporate echo canceller comprising a transfer function representing a difference between the first audio signal and the second audio signal. The modification provides converge of the signals for echo cancellation.
Regarding claim 14, Master, Vitaladevuni and Maher do not teach the system of claim 11, wherein the output of the adaptive acoustic echo canceller comprises a transfer function representing a difference between the first audio signal and the second audio signal after the adaptive acoustic echo canceller has converged.
Goodson teaches wherein the output of the adaptive acoustic echo canceller comprises a transfer function representing a difference between the first audio signal and the second audio signal after the adaptive acoustic echo canceller has converged (see fig. 3, col. 1, lines 44-65, col. 3, lines 61-col. 4, lines 13. Cancellation relation comprises H1-H2.sub.N-H3. H1 represents the transfer function of the first voltage divider. H2.sub.N represents the transfer function of the Nth second voltage divider and H3 represents the transfer function of the second echo canceller as a digital echo canceller after the taps have converged.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Vitaladevuni and Maher to incorporate echo canceller comprising a transfer function representing a difference between the first audio signal and the second audio signal. The modification provides converge of the signals for echo cancellation.
5. Claim(s) 6, 15 are rejected under 35 U.S.C. 103 as being unpatentable over Master et al. (US 2017/0041724) in view of Vitaladevuni et al. (US 9,704,478) in further view of Maher et al. (US 2013/0216071) in further view of Goodson et al. (US 8,731,183) in further view of Craven et al. (US 6,760,451).
Regarding claim 6, Master, Maher and Goodson(New) The playback device of claim 5, wherein the program instructions that are executable by the at least one processor such that the playback device is configured to estimate the environment acoustic characteristics based on the output of the adaptive acoustic echo canceller comprise program instructions that are executable by the at least one processor such that the playback device is configured to: estimate the environment acoustic characteristics as a difference between a first response represented by the transfer function and a second response representing output of the playback device in an anechoic chamber.
Vitaladevuni teaches wherein the program instructions that are executable by the at least one processor such that the playback device is configured to estimate the environment acoustic characteristics based on the output of the adaptive acoustic echo canceller comprise program instructions that are executable by the at least one processor such that the playback device (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Maher and Calatayud to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Craven teaches estimate the environment acoustic characteristics as a difference between a first response represented by the transfer function and a second response representing output of the playback device in an anechoic chamber (see fig. 1a, col. 5, line 37-45, col. 6, lines 42-64, col. 7, lines 19-60, col. 8, lines 53-65. Craven discloses a microphone and speaker in an acoustic environment and enclosed in a anechoic chamber. The microphone pics up the source signal fed to the filter via the loudspeaker. The microphone picks up the audio signal and supplies the audio pick up to the filter, the audio signal generated by the loudspeaker is picked up by the microphone through the acoustic environment which imposes on the audio signal a transfer function. The loudspeaker response is provided in an anechoic chamber with microphone. the audio signal outputted by the playback device would provide corrections and adjustments to the device (speaker) based on the computation of the transfer function computed from the two audio signals).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify Master, Maher, Vitaladevuni and Calatayud to incorporate frequency response representation in order to determine the transfer function and frequency response in a closed room. The medication would provide the adjustments to the speaker so the audio level is adjusted for output by the speaker. Thus the audio signal outputted by the playback device would provide corrections and adjustments to the device (speaker) based on the computation of the transfer function computed from the two audio signals.
Regarding claim 15, Master, Maher and Goodson do not teach the system of claim 14, wherein the program instructions that are executable by the at least one processor such that the system is configured to estimate the environment acoustic characteristics based on the output of the adaptive acoustic echo canceller comprise program instructions that are executable by the at least one processor such that the system is configured to: estimate the environment acoustic characteristics as a difference between a first response represented by the transfer function and a second response representing output of the playback device in an anechoic chamber.
Vitaladevuni teaches wherein the program instructions that are executable by the at least one processor such that the system is configured to estimate the environment acoustic characteristics based on the output of the adaptive acoustic echo canceller comprise program instructions that are executable by the at least one processor such that the system (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. The AEC typically filters the microphone signal by determining an estimate of the acoustic echo (e.g., the remote audio signal emitted from the loudspeaker and reflected in the local environment). The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal (e.g., the user's utterance). The estimate is obtained by applying a transformation to a reference signal that corresponds to the remote signal emitted from the loudspeaker. The transformation is implemented using an adaptive algorithm. Adaptive transformation relies on a feedback loop, which continuously adjusts a set of coefficients that are used to calculate the estimated echo from the far-end signal. By using a feedback loop to continuously adjust the coefficients, an AEC to can adapt its echo estimates to the local environment in which it operates. AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Maher and Calatayud to incorporate AEC filtering the microphone signal by determining an estimate of the acoustic echo. The AEC can then subtract the estimate from the microphone signal to produce an approximation of the true local signal. The modification provides for AECs use a reference signal representing the audio signal that is output through the speaker in order to determine what to remove from the input signal received via a microphone.
Craven teaches estimate the environment acoustic characteristics as a difference between a first response represented by the transfer function and a second response representing output of the playback device in an anechoic chamber (see fig. 1a, col. 5, line 37-45, col. 6, lines 42-64, col. 7, lines 19-60, col. 8, lines 53-65. Craven discloses a microphone and speaker in an acoustic environment and enclosed in a anechoic chamber. The microphone pics up the source signal fed to the filter via the loudspeaker. The microphone picks up the audio signal and supplies the audio pick up to the filter, the audio signal generated by the loudspeaker is picked up by the microphone through the acoustic environment which imposes on the audio signal a transfer function. The loudspeaker response is provided in an anechoic chamber with microphone. the audio signal outputted by the playback device would provide corrections and adjustments to the device (speaker) based on the computation of the transfer function computed from the two audio signals).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to modify Master, Maher, Vitaladevuni and Calatayud to incorporate frequency response representation in order to determine the transfer function and frequency response in a closed room. The medication would provide the adjustments to the speaker so the audio level is adjusted for output by the speaker. Thus the audio signal outputted by the playback device would provide corrections and adjustments to the device (speaker) based on the computation of the transfer function computed from the two audio signals.
6. Claim(s) 7, 16 are rejected under 35 U.S.C. 103 as being unpatentable over Master et al. (US 2017/0041724) in view of Vitaladevuni et al. (US 9,704,478) in further view of Maher et al. (US 2013/0216071) in further view of Relyea et al. (US 2008/0120099).
Regarding claim 7, Master and Vitaladevuni do not teach the playback device of claim 1, wherein the program instructions that are executable by the at least one processor such that the playback device is configured to determine the calibration that at least partially offsets the estimated environment acoustic characteristics comprise program instructions that are executable by the at least one processor such that the playback device is configured to: select a particular calibration from among a plurality of pre-determined calibrations, the particular calibration corresponding to the estimated environment acoustic characteristics.
Maher teaches wherein the program instructions that are executable by the at least one processor such that the playback device is configured to determine the calibration that at least partially offsets the estimated environment acoustic characteristics comprise program instructions that are executable by the at least one processor such that the playback device (see fig. 2. ¶ 0027. Calculation of the optimal equalization parameters is performed in a way that accommodates the transfer function of the microphone. Once the measurements and the calculations are complete, the optimal equalization parameters can be made available to the digital signal processor 102 which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers (208). This can include, for example, equalization for room reflections, cancellation of crosstalk from multiple channels, and/or the like. When additional audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Vitaladevuni to incorporate audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback. The modification provides for calibration that at least partially offsets the estimated environment acoustic characteristics and apply the determined calibration during playback of second audio content.
Relyea teach select a particular calibration from among a plurality of pre-determined calibrations, the particular calibration corresponding to the estimated environment acoustic characteristics (see fig. 7-8, ¶ 0015, 0023, 0058, 0072. Predetermined calibration setting may estimate and account for differences between the audio content as defined by the audio output signal and the audio content actually broadcast by the output device. filtration may be based on the audio output signal provided to the output device and at least one predetermined calibration setting.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Maher and Vitaladevuni to incorporate filtration performed on the audio input may be based on the audio output signal and at least one predetermined calibration setting. The modification provides for predetermined calibration setting may estimate and account for differences between the audio content as defined by the audio output signal and the audio content.
Regarding claim 16, Master and Vitaladevuni do not teach the system of claim 11, wherein the program instructions that are executable by the at least one processor such that the system is configured to determine the calibration that at least partially offsets the estimated environment acoustic characteristics comprise program instructions that are executable by the at least one processor such that the system is configured to: select a particular calibration from among a plurality of pre-determined calibrations, the particular calibration corresponding to the estimated environment acoustic characteristics.
Maher teaches wherein the program instructions that are executable by the at least one processor such that the system is configured to determine the calibration that at least partially offsets the estimated environment acoustic characteristics comprise program instructions that are executable by the at least one processor such that the system (see fig. 2. ¶ 0027. Calculation of the optimal equalization parameters is performed in a way that accommodates the transfer function of the microphone. Once the measurements and the calculations are complete, the optimal equalization parameters can be made available to the digital signal processor 102 which can implement filters for equalizing the non-ideal responses of the room environment, and the speakers (208). This can include, for example, equalization for room reflections, cancellation of crosstalk from multiple channels, and/or the like. When additional audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Vitaladevuni to incorporate audio content is sent to the speakers for playback, DSP 102 applies the equalization parameters to the audio content signal before sending the appropriately processed signal to the speakers for playback. The modification provides for calibration that at least partially offsets the estimated environment acoustic characteristics and apply the determined calibration during playback of second audio content.
Relyea teaches select a particular calibration from among a plurality of pre-determined calibrations, the particular calibration corresponding to the estimated environment acoustic characteristics (see fig. 7-8, ¶ 0015, 0023, 0058, 0072. Predetermined calibration setting may estimate and account for differences between the audio content as defined by the audio output signal and the audio content actually broadcast by the output device. filtration may be based on the audio output signal provided to the output device and at least one predetermined calibration setting.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master, Maher and Vitaladevuni to incorporate filtration performed on the audio input may be based on the audio output signal and at least one predetermined calibration setting. The modification provides for predetermined calibration setting may estimate and account for differences between the audio content as defined by the audio output signal and the audio content.
7. Claim(s) 8, 9, 10, 17, 18, 19 are rejected under 35 U.S.C. 103 as being unpatentable over Master et al. (US 2017/0041724) in view of Vitaladevuni et al. (US 9,704,478) in further view of Maher et al. (US 2013/0216071).
Regarding claim 8, Master teaches the playback device of claim 1, wherein the program instructions that are executable by the at least one processor such that the playback device is configured to determine the calibration that at least partially offsets the estimated environment acoustic characteristics comprise program instructions that are executable by the at least one processor such that the playback device is configured to: compute an audio processing algorithm that at least partially offsets the estimated environment acoustic characteristics when applied to playback by the playback device (see ¶ 0007, 0053-0054. Certain device characteristics stored in one or more databases may be provided to the processing component to further inform the tuning algorithm of the calibration process. The device characteristics may comprise at least one of speaker characteristics and microphone characteristics. Certain device information, such as speaker and/or microphone data may also be provided to the processing component to inform the calibration algorithms that optimize the processed sound for output to the speakers.).
Regarding claim 9, Master teaches the playback device of claim 8, wherein the program instructions that are executable by the at least one processor such that the playback device is configured to compute the audio processing algorithm comprise program instructions that are executable by the at least one processor such that the playback device is configured to: determine filter coefficients of a filter represented by the audio processing algorithm, the filter adjusting frequency components of audio signals output by the playback device to offset the acoustic characteristics (see ¶ 0007, 0053-0054. Certain device characteristics stored in one or more databases may be provided to the processing component to further inform the tuning algorithm of the calibration process. The device characteristics may comprise at least one of speaker characteristics and microphone characteristics. Certain device information, such as speaker and/or microphone data may also be provided to the processing component to inform the calibration algorithms that optimize the processed sound for output to the speakers.).
Regarding claim 10, Master and Maher do not teach the playback device of claim 1, further comprising a network microphone device, and wherein the at least one non-transitory computer-readable medium further comprises program instructions that are executable by the at least one processor such that the network microphone device is configured to: capture voice inputs via the at least one microphone.
Vitaladevuni teaches further comprising a network microphone device, and wherein the at least one non-transitory computer-readable medium further comprises program instructions that are executable by the at least one processor such that the network microphone device is configured to: capture voice inputs via the at least one microphone (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47, col. 9, line 22-60. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Maher to incorporate audio content that is received by the microphone. The modification provides for capture voice inputs via the at least one microphone.
Regarding claim 17, Master teaches the system of claim 11, wherein the program instructions that are executable by the at least one processor such that the system is configured to determine the calibration that at least partially offsets the estimated environment acoustic characteristics comprise program instructions that are executable by the at least one processor such that the system is configured to: compute an audio processing algorithm that at least partially offsets the estimated environment acoustic characteristics when applied to playback by the playback device (see ¶ 0007, 0053-0054. Certain device characteristics stored in one or more databases may be provided to the processing component to further inform the tuning algorithm of the calibration process. The device characteristics may comprise at least one of speaker characteristics and microphone characteristics. Certain device information, such as speaker and/or microphone data may also be provided to the processing component to inform the calibration algorithms that optimize the processed sound for output to the speakers.).
Regarding claim 18, Master teaches the system of claim 17, wherein the program instructions that are executable by the at least one processor such that the system is configured to compute the audio processing algorithm comprise program instructions that are executable by the at least one processor such that the system is configured to: determine filter coefficients of a filter represented by the audio processing algorithm, the filter adjusting frequency components of audio signals output by the playback device to offset the acoustic characteristics (see ¶ 0007, 0053-0054. Certain device characteristics stored in one or more databases may be provided to the processing component to further inform the tuning algorithm of the calibration process. The device characteristics may comprise at least one of speaker characteristics and microphone characteristics. Certain device information, such as speaker and/or microphone data may also be provided to the processing component to inform the calibration algorithms that optimize the processed sound for output to the speakers.).
Regarding claim 19, Master and Maher do not teach the system of claim 11, further comprising a network microphone device, and wherein the at least one non-transitory computer-readable medium further comprises program instructions that are executable by the at least one processor such that the network microphone device is configured to: capture voice inputs via the at least one microphone.
Vitaladevuni teaches further comprising a network microphone device, and wherein the at least one non-transitory computer-readable medium further comprises program instructions that are executable by the at least one processor such that the network microphone device is configured to: capture voice inputs via the at least one microphone (see fig. 5, col. 1, lines 36-57, col. 2, line 20-47, col. 9, line 22-60. An acoustic echo canceller (“AEC”) used to remove acoustic echo from an audio signal captured by a microphone in order to facilitate improved communication. When music is presented via a speaker, the signal from the music source is the reference signal. AECs typically include adaptive filters which transform the internal representation of the reference signal according to the acoustic reflections caused by the local environment.).
It would have been obvious to one of ordinary skill the art before the effective filing date of the claimed invention to modify Master and Maher to incorporate audio content that is received by the microphone. The modification provides for capture voice inputs via the at least one microphone.
Conclusion
8. Any inquiry concerning this communication or earlier communications from the examiner should be directed to ASSAD MOHAMMED whose telephone number is (571)270-7253. The examiner can normally be reached 9:00AM-5:00PM.
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/ASSAD MOHAMMED/Examiner, Art Unit 2691
/DUC NGUYEN/Supervisory Patent Examiner, Art Unit 2691
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