Patent Description:
Options for accessing and listening to digital audio in an out-loud setting were limited until in <NUM>, when SONOS, Inc. filed for one of its first patent applications, entitled "Method for Synchronizing Audio Playback between Multiple Networked Devices," and began offering a media playback system for sale in <NUM>. The Sonos Wireless HiFi System enables people to experience music from many sources via one or more networked playback devices. Through a software control application installed on a smartphone, tablet, or computer, one can play what he or she wants in any room that has a networked playback device. Additionally, using the controller, for example, different songs can be streamed to each room with a playback device, rooms can be grouped together for synchronous playback, or the same song can be heard in all rooms synchronously.

Given the ever-growing interest in digital media, there continues to be a need to develop consumer-accessible technologies to further enhance the listening experience.

<CIT> describes a method comprising: receiving, by a mobile device, acoustic input from an environment of the mobile device, detecting whether the acoustic input includes a voice command from a user without requiring receipt of an explicit trigger from the user, and initiating responding to the detected voice command. <CIT> describes an echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system. <CIT> describes a method for reducing disturbing tone signals from acoustic echoes or background noises in speech/audio signal enhancement processing. <CIT> describes a system for detecting a designated wake-up word.

According to a first aspect of the present invention, there is provided a method according to appended claim <NUM>. According to a second aspect of the present invention there is provided a computer-readable medium according to appended claim <NUM>. According to a third aspect of the present invention there is provided a playback device according to appended claim <NUM>. Preferable aspects are set in the dependent claims.

The drawings are for purposes of illustrating example embodiments, but it is understood that the inventions are not limited to the arrangements and instrumentality shown in the drawings. In the drawings, identical reference numbers identify at least generally similar elements. To facilitate the discussion of any particular element, the most significant digit or digits of any reference number refers to the Figure in which that element is first introduced. For example, element <NUM> is first introduced and discussed with reference to <FIG>.

Networked microphone device may be used to control a household using voice control. Voice control can be beneficial for a "smart" home having a system of smart devices, such as playback devices, wireless illumination devices, thermostats, door locks, home-automation devices, as well as other examples. In some implementations, the system of smart devices includes a networked microphone device configured to detect voice inputs. A voice assistant service facilitates processing of the voice inputs. Traditionally, the voice assistant service includes remote servers that receive and process voice inputs. The voice service may return responses to voice inputs, which might include control of various smart devices or audio or video information (e.g., a weather report), among other examples.

A voice input typically includes an utterance with a wake word followed by an utterance containing a user request. A wake word, when uttered, may invoke a particular voice assistance service. For instance, in querying the AMAZON® voice assistant service, a user might speak a wake word "Alexa. " Other examples include "Ok, Google" for invoking the GOOGLE® voice assistant service and "Hey, Siri" for invoking the APPLE® voice assistant service.

Upon detecting a wake word, a networked microphone device may listen for the user request in the voice utterance following the wake word. In some instances, the user request may include a command to control a third party device, such as a smart illumination device (e.g., a PHILIPS HUE ® lighting device), a thermostat (e.g., NEST® thermostat), or a media playback device (e.g., a Sonos® playback device). For example, a user might speak the wake word "Alexa" followed by the utterance "turn on the living room" to turn on illumination devices. A user might speak the same wake word followed by the utterance "set the thermostat to <NUM> degrees. " The user may also utter a request for a playback device to play a particular song, an album, or a playlist of music.

When a networked microphone device detects a wake word, the networked microphone device provides an acknowledgement of the wake word to the user, so that the user can be informed that the networked microphone device has detected the wake word. In some implementations, this acknowledgement is provided by way of a light response (e.g., the illumination of one or more light emitting diodes, perhaps in certain colors and/or patterns). A possible disadvantage of using a light response to acknowledge wake word detection is that the user must be looking in the direction of the networked microphone device to see the light response.

Alternatively, example networked microphone devices may provide acknowledgement of wake word detection by way of an audio response. For instance, one or more speakers may play back an audible "acknowledgement" tone shortly after a networked microphone device detects a wake word in captured audio. However, wake words typically precede a voice utterance (e.g., a voice command or query) spoken by the user. As such, an acknowledgement tone may overlap the user's voice utterance. Given this overlap, the acknowledgement tone may interfere with the networked microphone device's capturing of the voice utterance.

In an effort to avoid or lessen interference from the acknowledgement tone in the capturing of the voice utterance, a networked microphone device may use an Acoustic Echo Canceller ("AEC") to remove the sound of the acknowledgement tone from the signal captured by microphone(s) of the networked microphone device. This removal is intended to improve the signal-to-noise ratio of a voice input to other sound within the acoustic environment, which includes the sound produced by the one or more speakers in playing back the acknowledgement tone, so as to provide a less noisy signal to a voice assistant service.

In example implementations, an AEC is implemented within the audio processing pipeline of a networked microphone device. Input to an AEC may include the signal captured by the microphone(s) and a reference signal representing the analog audio expected to be output by the transducers (e.g., the acknowledgement tone). Given these inputs, the AEC attempts to find a transfer function (i.e., a 'filter') that transforms the reference signal into the captured microphone signal with minimal error. Inverting the resulting AEC output and mixing it with the microphone signal causes a redaction of the audio output signal from the signal captured by the microphone(s). Moreover, AEC is an iterative process, whereby the error during each iteration of the AEC is used to update the filter for the next iteration of the AEC. Using this process, over successive iterations, the AEC "converges" to an effective cancellation of the reference signal from the measured signal.

However, being an iterative process, an AEC may take some time to converge to an effective cancellation of the reference signal from the measured signal. For instance, example AEC processes might take <NUM> milliseconds or longer to converge, depending on the noise in the acoustic environment. If the AEC already active and stable (i.e., converged) when an acknowledgement tone is outputted - perhaps because the device is playing back other audio content, such as music - then the AEC may effectively cancel the acknowledgement tone (in addition to the other audio content). However, if instead the AEC is inactive (i.e., not active and stable) when the acknowledgement tone is outputted, then the AEC is unlikely to have enough time to converge and thereby cancel the acknowledgement tone effectively, as the reference signal might be only a few hundred milliseconds in length.

To facilitate effective cancellation of an acknowledgement tone whether the AEC is active or not, example networked microphone devices described herein implement two acoustic echo cancellations processes. If the networked microphone device is playing back audio content (e.g., music) via one or more audio drivers when a wake word is detected in captured audio, the networked microphone device runs (or continue running) a first AEC to cancel the acoustic echo of the acknowledgement tone from the captured audio. The first AEC also cancels the acoustic echo of the played back audio content. Conversely, if the one or more audio drivers of the networked microphone device are idle when the wake word is detected in the captured audio, the networked microphone device activates a second AEC to cancel the acoustic echo of the acknowledgement tone from the captured audio.

As compared with the first AEC, the second AEC is designed to converge significantly faster, thereby enabling the second AEC to cancel the acknowledgement tone effectively, even where the acknowledgement tone is only a few hundred milliseconds in length. In particular, the second AEC may converge more quickly than the first AEC by cancelling acoustic echo from only the specific frequency ranges (a. , frequency "bins") in which the acknowledgement tone has content. In contrast, the first AEC is configured to cancel acoustic echo across the entire audible frequency spectrum (e.g., <NUM> - <NUM>,<NUM>). By processing a subset of the frequency range that the first AEC processes, the second AEC may converge significantly faster (e.g., quickly enough to converge and cancel an acknowledgement tone that is only a few hundred milliseconds in length). In practice, in example implementations, such techniques have increased the rate of convergence by <NUM>% as compared with a full-spectrum acoustic echo cancellation process.

Example techniques described herein may involve selecting among different acoustic echo cancellers implemented in a networked microphone device. An example implementation may involve capturing, via the one or more microphones, first audio within an acoustic environment, determining whether the one or more speakers are (a) playing back audio content or (b) idle, determining whether the one or more speakers are (a) playing back audio content or (b) idle, and identifying a set of frequency bands of the full audible frequency spectrum in which the audible tone in acknowledgment of the detected wake word has content.

The example implementation may further involve in response to detecting the wake word for the voice service and before playing an audible tone in acknowledgement of the detected wake word on the one or more speakers, activating either (a) a first sound canceller or (b) a second sound canceller. Activating either the (a) first sound canceller or (b) the second sound canceller may involve when the one or more speakers are playing back audio content, activating the first sound canceller, the first sound canceller configured to cancel audio output from the one or more speakers in a full audible frequency spectrum, and when the one or more speakers are idle, activating the second sound canceller process, the second sound canceller configured to cancel audio output from the one or more speakers in the identified frequency bands of the full audible frequency spectrum in which the audible tone in acknowledgment of the detected wake word has content.

The example implementation may also involve in response to detecting the wake word for the voice service and after activating either (a) the first sound canceller or (b) the second sound canceller, outputting the audible tone in acknowledgement of the detected wake word via the one or more speakers, and capturing, via the one or more microphones, second audio within the acoustic environment. The second audio includes sound produced by the one or more speakers in outputting the audible tone in acknowledgement of the detected wake word. The implementation may further involve cancelling the audible tone in acknowledgement of the detected wake word from the captured second audio using the activated audio canceller.

This example implementation may be embodied as a method, a device configured to carry out the implementation, a system of devices configured to carry out the implementation, or a non-transitory computer-readable medium containing instructions that are executable by one or more processors to carry out the implementation, among other examples. It will be understood by one of ordinary skill in the art that this disclosure includes numerous other embodiments, including combinations of the example features described herein. Further, any example operation described as being performed by a given device to illustrate a technique may be performed by any suitable devices, including the devices described herein. Yet further, any device may cause another device to perform any of the operations described herein.

While some examples described herein may refer to functions performed by given actors such as "users" and/or other entities, it should be understood that this description is for purposes of explanation only. The claims should not be interpreted to require action by any such example actor unless explicitly required by the language of the claims themselves.

<FIG> illustrates an example configuration of a media playback system <NUM> in which one or more embodiments disclosed herein may be implemented. The media playback system <NUM> as shown is associated with an example home environment having several rooms and spaces, such as for example, an office, a dining room, and a living room. Within these rooms and spaces, the media playback system <NUM> includes playback devices <NUM> (identified individually as playback devices 102a-<NUM>), network microphone devices <NUM> (identified individually as "NMD(s)" 103a-<NUM>), and controller devices 104a and 104b (collectively "controller devices <NUM>"). The home environment may include other network devices, such as one or more smart illumination devices <NUM> and a smart thermostat <NUM>.

The various playback, network microphone, and controller devices <NUM>-<NUM> and/or other network devices of the media playback system <NUM> may be coupled to one another via point-to-point and/or over other connections, which may be wired and/or wireless, via a local area network (LAN) via a network router <NUM>. For example, the playback device 102j (designated as "LEFT") may have a point-to-point connection with the playback device 102a (designated as "RIGHT"). In one embodiment, the LEFT playback device 102j may communicate over the point-to-point connection with the RIGHT playback device 102a. In a related embodiment, the LEFT playback device 102j may communicate with other network devices via the point-to-point connection and/or other connections via the LAN.

The network router <NUM> may be coupled to one or more remote computing device(s) <NUM> via a wide area network (WAN) <NUM>. In some embodiments, the remote computing device(s) may be cloud servers. The remote computing device(s) <NUM> may be configured to interact with the media playback system <NUM> in various ways. For example, the remote computing device(s) may be configured to facilitate streaming and controlling playback of media content, such as audio, in the home environment. In one aspect of the technology described in greater detail below, the remote computing device(s) <NUM> are configured to provide an enhanced VAS <NUM> for the media playback system <NUM>.

In some embodiments, one or more of the playback devices <NUM> may include an on-board (e.g., integrated) network microphone device. For example, the playback devices 102a-e include corresponding NMDs 103a-e, respectively. Playback devices that include network devices may be referred to herein interchangeably as a playback device or a network microphone device unless expressly stated otherwise.

In some embodiments, one or more of the NMDs <NUM> may be a stand-alone device. For example, the NMDs 103f and <NUM> may be stand-alone network microphone devices. A stand-alone network microphone device may omit components typically included in a playback device, such as a speaker or related electronics. In such cases, a stand-alone network microphone device might not produce audio output or may produce limited audio output (e.g., relatively low-quality output relative to quality of output by a playback device).

In some embodiments, one or more network microphone devices can be assigned to a playback device or a group of playback devices. In some embodiments, a network microphone device can be assigned to a playback device that does not include an onboard network microphone device. For example, the NMD 103f may be assigned to one or more of the playback devices <NUM> in its vicinity, such as one or both of the playback devices 102i and <NUM> in the kitchen and dining room spaces, respectively. In such a case, the NMD 103f may output audio through the playback device(s) to which it is assigned. Further details regarding assignment of network microphone devices are described, for example, in <CIT>, and titled "Default Playback Device Designation," and <CIT> and titled "Default Playback Devices.

In some embodiments, a network microphone device may be configured such that it is dedicated exclusively to a particular VAS. In one example, the NMD 103a in the living room space may be dedicated exclusively to the enhanced VAS <NUM>. In such case, the NMD 102a might not invoke any other VAS except the enhanced VAS <NUM>. In a related example, other ones of the NMDs <NUM> may be configured to invoke the enhanced <NUM> VAS and one or more other VASes, such as a traditional VAS. Other examples of bonding and assigning network microphone devices to playback devices and/or VASes are possible. In some embodiments, the NMDs <NUM> might not be bonded or assigned in a particular manner.

Further aspects relating to the different components of the example media playback system <NUM> and how the different components may interact to provide a user with a media experience may be found in the following sections. For instance, the technologies described herein may be useful in other home environment configurations comprising more or fewer of any of the playback, network microphone, and/or controller devices <NUM>-<NUM>. Additionally, the technologies described herein may be useful in environments where multi-zone audio may be desired, such as, for example, a commercial setting like a restaurant, mall or airport, a vehicle like a sports utility vehicle (SUV), bus or car, a ship or boat, an airplane, and so on.

<FIG> is a functional block diagram illustrating certain aspects of a selected one of the playback devices <NUM> shown in <FIG>. As shown, such a playback device may include a processor <NUM>, software components <NUM>, memory <NUM>, audio processing components <NUM>, audio amplifier(s) <NUM>, speaker(s) <NUM>, and a network interface <NUM> including wireless interface(s) <NUM> and wired interface(s) <NUM>.

A playback device may further include a user interface <NUM>. The user interface <NUM> may facilitate user interactions independent of or in conjunction with one or more of the controller devices <NUM>. In various embodiments, the user interface <NUM> includes one or more of physical buttons and/or graphical interfaces provided on touch sensitive screen(s) and/or surface(s), among other possibilities, for a user to directly provide input. The user interface <NUM> may further include one or more of lights and the speaker(s) to provide visual and/or audio feedback to a user.

In some embodiments, the processor <NUM> may be a clock-driven computing component configured to process input data according to instructions stored in the memory <NUM>. The memory <NUM> may be a tangible computer-readable medium configured to store instructions executable by the processor <NUM>. For example, the memory <NUM> may be data storage that can be loaded with one or more of the software components <NUM> executable by the processor <NUM> to achieve certain functions. In one example, the functions may involve a playback device retrieving audio data from an audio source or another playback device. In another example, the functions may involve a playback device sending audio data to another device on a network. In yet another example, the functions may involve pairing of a playback device with one or more other playback devices to create a multi-channel audio environment.

Certain functions may involve a playback device synchronizing playback of audio content with one or more other playback devices. During synchronous playback, a listener should not perceive time-delay differences between playback of the audio content by the synchronized playback devices. <CIT>, and titled "System and method for synchronizing operations among a plurality of independently clocked digital data processing devices," provides in more detail some examples for audio playback synchronization among playback devices.

The memory <NUM> may be further configured to store data associated with a playback device. For example, the memory may store data corresponding to one or more zones and/or zone groups a playback device is a part of. One or more of the zones and/or zone groups may be named according to the room or space in which device(s) are located. For example, the playback and network microphone devices in the living room space shown in <FIG> may be referred to as a zone group named Living Room. As another example, the playback device <NUM> in the dining room space may be named as a zone "Dining Room. " The zones and/or zone groups may also have uniquely assigned names, such as "Nick's Room," as shown in <FIG>.

The memory <NUM> may be further configured to store other data. Such data may pertain to audio sources accessible by a playback device or a playback queue that the playback device (or some other playback device(s)) may be associated with. The data stored in the memory <NUM> may be stored as one or more state variables that are periodically updated and used to describe the state of the playback device. The memory <NUM> may also include the data associated with the state of the other devices of the media system, and shared from time to time among the devices so that one or more of the devices have the most recent data associated with the system. Other embodiments are also possible.

The audio processing components <NUM> may include one or more digital-to-analog converters (DAC), an audio preprocessing component, an audio enhancement component or a digital signal processor (DSP), and so on. In some embodiments, one or more of the audio processing components <NUM> may be a subcomponent of the processor <NUM>. In one example, audio content may be processed and/or intentionally altered by the audio processing components <NUM> to produce audio signals. The produced audio signals may then be provided to the audio amplifier(s) <NUM> for amplification and playback through speaker(s) <NUM>. Particularly, the audio amplifier(s) <NUM> may include devices configured to amplify audio signals to a level for driving one or more of the speakers <NUM>. The speaker(s) <NUM> may include an individual transducer (e.g., a "driver") or a complete speaker system involving an enclosure with one or more drivers. A particular driver of the speaker(s) <NUM> may include, for example, a subwoofer (e.g., for low frequencies), a mid-range driver (e.g., for middle frequencies), and/or a tweeter (e.g., for high frequencies). In some cases, each transducer in the one or more speakers <NUM> may be driven by an individual corresponding audio amplifier of the audio amplifier(s) <NUM>. In addition to producing analog signals for playback, the audio processing components <NUM> may be configured to process audio content to be sent to one or more other playback devices for playback.

Audio content to be processed and/or played back by a playback device may be received from an external source, such as via an audio line-in input connection (e.g., an auto-detecting <NUM> audio line-in connection) or the network interface <NUM>.

The network interface <NUM> may be configured to facilitate a data flow between a playback device and one or more other devices on a data network. As such, a playback device may be configured to receive audio content over the data network from one or more other playback devices in communication with a playback device, network devices within a local area network, or audio content sources over a wide area network such as the Internet. In one example, the audio content and other signals transmitted and received by a playback device may be transmitted in the form of digital packet data containing an Internet Protocol (IP)-based source address and IP-based destination addresses. In such a case, the network interface <NUM> may be configured to parse the digital packet data such that the data destined for a playback device is properly received and processed by the playback device.

As shown, the network interface <NUM> may include wireless interface(s) <NUM> and wired interface(s) <NUM>. The wireless interface(s) <NUM> may provide network interface functions for a playback device to wirelessly communicate with other devices (e.g., other playback device(s), speaker(s), receiver(s), network device(s), control device(s) within a data network the playback device is associated with) in accordance with a communication protocol (e.g., any wireless standard including IEEE <NUM>. 11a, <NUM>. 11b, <NUM>, <NUM>. 11n, <NUM>. 11ac, <NUM>, <NUM> mobile communication standard, and so on). The wired interface(s) <NUM> may provide network interface functions for a playback device to communicate over a wired connection with other devices in accordance with a communication protocol (e.g., IEEE <NUM>). While the network interface <NUM> shown in <FIG> includes both wireless interface(s) <NUM> and wired interface(s) <NUM>, the network interface <NUM> may in some embodiments include only wireless interface(s) or only wired interface(s).

In some embodiments, a playback device and one other playback device may be paired to play two separate audio components of audio content. For example, the LEFT playback device 102j in the Living Room may be configured to play a left channel audio component, while the RIGHT playback device 102a may be configured to play a right channel audio component, thereby producing or enhancing a stereo effect of the audio content. Similarly, the playback device <NUM> designated to the Dining Room may be configured to play a left channel audio component, while the playback device 102i designated to the Kitchen may be configured to play a right channel audio component. Paired playback devices may further play audio content in synchrony with other playback devices. Paired playback device may also be referred to as "bonded playback devices.

In some embodiments, one or more of the playback devices may be sonically consolidated with one or more other playback devices to form a single, consolidated playback device. A consolidated playback device may include separate playback devices each having additional or different speaker drivers through which audio content may be rendered. For example, a playback device designed to render low frequency range audio content (e.g., the playback device <NUM> designated as a subwoofer or "SUB") may be consolidated with a full-frequency playback device (e.g., the playback device 102b designated as "FRONT") to render the lower frequency range of the consolidated device. In such a case, the full frequency playback device, when consolidated with the low frequency playback device, may be configured to render only the mid and high frequency components of audio content, while the low-frequency playback device renders the low frequency component of the audio content. The consolidated playback device may be paired or consolidated with one or more other playback devices. For example, <FIG> shows the SUB playback device <NUM> consolidated with the FRONT playback device 102b to form subwoofer and center channels, and further consolidated with the RIGHT playback device 102a and the LEFT playback device 102j.

As discussed above, a playback device may include a network microphone device, such as one of the NMDs <NUM>, as show in <FIG>. A network microphone device may share some or all the components of a playback device, such as the processor <NUM>, the memory <NUM>, the microphone(s) <NUM>, etc. In other examples, a network microphone device includes components that are dedicated exclusively to operational aspects of the network microphone device. For example, a network microphone device may include far-field microphones and/or voice processing components, which in some instances a playback device may not include. In another example, a network microphone device may include a touch-sensitive button for enabling/disabling a microphone. In yet another example, a network microphone device can be a stand-alone device, as discussed above.

By way of illustration, SONOS, Inc. presently offers (or has offered) for sale certain playback devices including a "PLAY:<NUM>," "PLAY:<NUM>," "PLAY:<NUM>," "PLAYBAR," "CONNECT:AMP," "CONNECT," and "SUB. " Any other past, present, and/or future playback devices may additionally or alternatively be used to implement the playback devices of example embodiments disclosed herein. Additionally, it is understood that a playback device is not limited to the example illustrated in <FIG> or to the SONOS product offerings. For example, a playback device may include a wired or wireless headphone. In another example, a playback device may include or interact with a docking station for personal mobile media playback devices. In yet another example, a playback device may be integral to another device or component such as a television, a lighting fixture, or some other device for indoor or outdoor use.

Referring back to the media playback system <NUM> of <FIG>, the media playback system <NUM> may be established with one or more playback zones, after which one or more of the playback and/or network devices <NUM>-<NUM> may be added or removed to arrive at the example configuration shown in <FIG>. As discussed above, zones and zone groups may be given a unique name and/or a name corresponding to the space in which device(s) are located.

In one example, one or more playback zones in the environment of <FIG> may each be playing different audio content. For instance, the user may be grilling in the Balcony zone and listening to hip hop music being played by the playback device 102c while another user is preparing food in the Kitchen zone and listening to classical music being played by the playback device 102i. In another example, a playback zone may play the same audio content in synchrony with another playback zone. For instance, the user may be in the Office zone where the playback device 102d is playing the same hip-hop music that is being playing by playback device 102c in the Balcony zone. In such a case, playback devices 102c and 102d may be playing the hip-hop in synchrony such that the user may seamlessly (or at least substantially seamlessly) enjoy the audio content that is being played out-loud while moving between different playback zones. Synchronization among playback zones may be achieved in a manner similar to that of synchronization among playback devices, as described in previously referenced <CIT>.

A network microphone device may receive voice inputs from a user in its vicinity. A network microphone device may capture a voice input upon detection of the user speaking the input. For instance, in the example shown in <FIG>, the NMD 103a may capture the voice input of a user in the vicinity of the Living Room, Dining Room, and/or Kitchen zones. In some instances, other network microphone devices in the home environment, such as the NMD 104f in the Kitchen and/or the other NMD 104b in the Living Room may capture the same voice input. In such instances, network devices that detect the voice input may be configured to arbitrate between one another so that fewer or only the most proximate one of the NMDs <NUM> process the user's voice input. Other examples for selecting network microphone devices for processing voice input can be found, for example, in <CIT>, and titled "Dynamic Player Selection for Audio Signal Processing" and <CIT>, and titled "Voice Detection by Multiple Devices. " A network microphone device may control selected playback and/or network microphone devices <NUM>-<NUM> in response to voice inputs, as described in greater detail below.

As suggested above, the zone configurations of the media playback system <NUM> may be dynamically modified. As such, the media playback system <NUM> may support numerous configurations. For example, if a user physically moves one or more playback devices to or from a zone, the media playback system <NUM> may be reconfigured to accommodate the change(s). For instance, if the user physically moves the playback device 102c from the Balcony zone to the Office zone, the Office zone may now include both the playback devices 102c and 102d. In some cases, the use may pair or group the moved playback device 102c with the Office zone and/or rename the players in the Office zone using, e.g., one of the controller devices <NUM> and/or voice input. As another example, if one or more playback devices <NUM> are moved to a particular area in the home environment that is not already a playback zone, the moved playback device(s) may be renamed or associated with a playback zone for the particular area.

Further, different playback zones of the media playback system <NUM> may be dynamically combined into zone groups or split up into individual playback zones. For example, the Dining Room zone and the Kitchen zone may be combined into a zone group for a dinner party such that playback devices 102i and <NUM> may render audio content in synchrony. As another example, playback devices <NUM> consolidated in the Living Room zone for the previously described consolidated TV arrangement may be split into (i) a television zone and (ii) a separate listening zone. The television zone may include the FRONT playback device 102b. The listening zone may include the RIGHT, LEFT, and SUB playback devices 102a, 102j, and <NUM>, which may be grouped, paired, or consolidated, as described above. Splitting the Living Room zone in such a manner may allow one user to listen to music in the listening zone in one area of the living room space, and another user to watch the television in another area of the living room space. In a related example, a user may implement either of the NMD 103a or 103b to control the Living Room zone before it is separated into the television zone and the listening zone. Once separated, the listening zone may be controlled by a user in the vicinity of the NMD 103a, and the television zone may be controlled by a user in the vicinity of the NMD 103b. As described above, however, any of the NMDs <NUM> may be configured to control the various playback and other devices of the media playback system <NUM>.

<FIG> is a functional block diagram illustrating certain aspects of a selected one of the controller devices <NUM> of the media playback system <NUM> of <FIG>. Such controller devices may also be referred to as a controller. The controller device shown in <FIG> may include components that are generally similar to certain components of the network devices described above, such as a processor <NUM>, memory <NUM>, microphone(s) <NUM>, and a network interface <NUM>. In one example, a controller device may be a dedicated controller for the media playback system <NUM>. In another example, a controller device may be a network device on which media playback system controller application software may be installed, such as for example, an iPhone™™, iPad™ or any other smart phone, tablet or network device (e.g., a networked computer such as a PC or Mac™).

The memory <NUM> of a controller device may be configured to store controller application software and other data associated with the media playback system <NUM> and a user of the system <NUM>. The memory <NUM> may be loaded with one or more software components <NUM> executable by the processor <NUM> to achieve certain functions, such as facilitating user access, control, and configuration of the media playback system <NUM>. A controller device communicates with other network devices over the network interface <NUM>, such as a wireless interface, as described above.

In one example, data and information (e.g., such as a state variable) may be communicated between a controller device and other devices via the network interface <NUM>. For instance, playback zone and zone group configurations in the media playback system <NUM> may be received by a controller device from a playback device, a network microphone device, or another network device, or transmitted by the controller device to another playback device or network device via the network interface <NUM>. In some cases, the other network device may be another controller device.

Playback device control commands such as volume control and audio playback control may also be communicated from a controller device to a playback device via the network interface <NUM>. As suggested above, changes to configurations of the media playback system <NUM> may also be performed by a user using the controller device. The configuration changes may include adding/removing one or more playback devices to/from a zone, adding/removing one or more zones to/from a zone group, forming a bonded or consolidated player, separating one or more playback devices from a bonded or consolidated player, among others.

The user interface(s) <NUM> of a controller device may be configured to facilitate user access and control of the media playback system <NUM>, by providing controller interface(s) such as the controller interfaces 400a and 400b (collectively "controller interface <NUM>") shown in <FIG>, respectively. Referring to <FIG> together, the controller interface <NUM> includes a playback control region <NUM>, a playback zone region <NUM>, a playback status region <NUM>, a playback queue region <NUM>, and a sources region <NUM>. The user interface <NUM> as shown is just one example of a user interface that may be provided on a network device such as the controller device shown in <FIG> and accessed by users to control a media playback system such as the media playback system <NUM>. Other user interfaces of varying formats, styles, and interactive sequences may alternatively be implemented on one or more network devices to provide comparable control access to a media playback system.

The playback control region <NUM> (<FIG>) may include selectable (e.g., by way of touch or by using a cursor) icons to cause playback devices in a selected playback zone or zone group to 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.

The playback zone region <NUM> (<FIG>) may include representations of playback zones within the media playback system <NUM>.

For example, as shown, a "group" icon may be provided within each of the graphical representations of playback zones. The "group" icon provided within a graphical representation of a particular zone may be selectable to bring up options to select one or more other zones in the media playback system to be grouped with the particular zone. Once grouped, playback devices in the zones that have been grouped with the particular zone will be configured to play audio content in synchrony with the playback device(s) in the particular zone. Analogously, a "group" icon may be provided within a graphical representation of a zone group. In this case, the "group" icon may be selectable to bring up options to deselect one or more zones in the zone group to be removed from the zone group. Other interactions and implementations for grouping and ungrouping zones via a user interface such as the user interface <NUM> are also possible. The representations of playback zones in the playback zone region <NUM> (<FIG>) may be dynamically updated as playback zone or zone group configurations are modified.

The playback status region <NUM> (<FIG>) may include graphical representations of audio content that is presently being played, previously played, or scheduled to play next in the selected playback zone or zone group. The selected playback zone or zone group may be visually distinguished on the user interface, such as within the playback zone region <NUM> and/or the playback status region <NUM>. The graphical representations may include track title, artist name, album name, album year, track length, and other relevant information that may be useful for the user to know when controlling the media playback system via the user interface <NUM>.

With reference still to <FIG>, the graphical representations of audio content in the playback queue region <NUM> (<FIG>) may include track titles, artist names, track lengths, and other relevant information associated with the audio content in the playback queue. In one example, graphical representations of audio content may be selectable to bring up additional selectable icons to manage and/or manipulate the playback queue and/or audio content represented in the playback queue. For instance, a represented audio content may be removed from the playback queue, moved to a different position within the playback queue, or selected to be played immediately, or after any currently playing audio content, among other possibilities. A playback queue associated with a playback zone or zone group may be stored in a memory on one or more playback devices in the playback zone or zone group, on a playback device that is not in the playback zone or zone group, and/or some other designated device. Playback of such a playback queue may involve one or more playback devices playing back media items of the queue, perhaps in sequential or random order.

The sources region <NUM> may include graphical representations of selectable audio content sources and selectable voice assistants associated with a corresponding VAS. The VASes may be selectively assigned. In some examples, multiple VASes, such as AMAZON's ALEXA® and another voice service, may be invokable by the same network microphone device. In some embodiments, a user may assign a VAS exclusively to one or more network microphone devices, as discussed above. For example, a user may assign first VAS to one or both of the NMDs 102a and 102b in the living room space shown in <FIG>, and a second VAS to the NMD 103f in the kitchen space. Other examples are possible.

The audio sources in the sources region <NUM> may be audio content sources from which audio content may be retrieved and played by the selected playback zone or zone group. One or more playback devices in a zone or zone group may be configured to retrieve for playback audio content (e.g., according to a corresponding URI or URL for the audio content) from a variety of available audio content sources. In one example, audio content may be retrieved by a playback device directly from a corresponding audio content source (e.g., a line-in connection). In another example, audio content may be provided to a playback device over a network via one or more other playback devices or network devices.

Example audio content sources may include a memory of one or more playback devices in a media playback system such as the media playback system <NUM> of <FIG>, local music libraries on one or more network devices (such as a controller device, a network-enabled personal computer, or a networked-attached storage (NAS), for example), streaming audio services providing audio content via the Internet (e.g., the cloud), or audio sources connected to the media playback system via a line-in input connection on a playback device or network devise, among other possibilities.

<FIG> is a functional block diagram showing additional features of one or more of the NMDs <NUM> in accordance with aspects of the disclosure. The network microphone device shown in <FIG> may include components that are generally similar to certain components of network microphone devices described above, such as the processor <NUM> (<FIG>), network interface <NUM> (<FIG>), microphone(s) <NUM>, and the memory <NUM>. Although not shown for purposes of clarity, a network microphone device may include other components, such as speakers, amplifiers, signal processors, as discussed above.

The microphone(s) <NUM> may be a plurality of microphones arranged to detect sound in the environment of the network microphone device. In one example, the microphone(s) <NUM> may be arranged to detect audio from one or more directions relative to the network microphone device. The microphone(s) <NUM> may be sensitive to a portion of a frequency range. In one example, a first subset of the microphone(s) <NUM> may be sensitive to a first frequency range, while a second subset of the microphone(<NUM>) <NUM> may be sensitive to a second frequency range. The microphone(s) <NUM> may further be arranged to capture location information of an audio source (e.g., voice, audible sound) and/or to assist in filtering background noise. Notably, in some embodiments the microphone(s) <NUM> may have a single microphone rather than a plurality of microphones.

A network microphone device may further include wake-word detector <NUM>, beam former <NUM>, acoustic echo canceller (AEC) <NUM>, and speech/text conversion <NUM> (e.g., voice-to-text and text-to-voice). In various embodiments, one or more of the wake-word detector <NUM>, beam former <NUM>, AEC <NUM>, and speech/text conversion <NUM> may be a subcomponent of the processor <NUM>, or implemented in software stored in memory <NUM> which is executable by the processor <NUM>.

The wake-word detector <NUM> is configured to monitor and analyze received audio to determine if any wake words are present in the audio. The wake-word detector <NUM> may analyze the received audio using a wake word detection algorithm. If the wake-word detector <NUM> detects a wake word, a network microphone device may process voice input contained in the received audio. Example wake word detection algorithms accept audio as input and provide an indication of whether a wake word is present in the audio. Many first- and third-party wake word detection algorithms are known and commercially available. For instance, operators of a voice service may make their algorithm available for use in third-party devices. Alternatively, an algorithm may be trained to detect certain wake-words.

In some embodiments, the wake-word detector <NUM> runs multiple wake word detections algorithms on the received audio simultaneously (or substantially simultaneously). As noted above, different voice services (e.g. AMAZON's ALEXA®, APPLE's SIRI®, or MICROSOFT's CORTANA®) each use a different wake word for invoking their respective voice service. To support multiple services, the wake word detector <NUM> may run the received audio through the wake word detection algorithm for each supported voice service in parallel.

The beam former <NUM> and AEC <NUM> are configured to detect an audio signal and determine aspects of voice input within the detect audio, such as the direction, amplitude, frequency spectrum, etc. For example, the beam former <NUM> and AEC <NUM> may be used in a process to determine an approximate distance between a network microphone device and a user speaking to the network microphone device. In another example, a network microphone device may detective a relative proximity of a user to another network microphone device in a media playback system.

<FIG> is a diagram of an example voice input in accordance with aspects of the disclosure. The voice input may be captured by a network microphone device, such as by one or more of the NMDs <NUM> shown in <FIG>. The voice input may include a wake word portion 557a and a voice utterance portion 557b (collectively "voice input <NUM>"). In some embodiments, the wake word 557a can be a known wake word, such as "Alexa," which is associated with AMAZON's ALEXA®).

In some embodiments, a network microphone device may output an audible and/or visible response upon detection of the wake word portion 557a. In addition or alternately, a network microphone device may output an audible and/or visible response after processing a voice input and/or a series of voice inputs (e.g., in the case of a multi-turn request).

The voice utterance portion 557b may include, for example, one or more spoken commands <NUM> (identified individually as a first command 558a and a second command 558b) and one or more spoken keywords <NUM> (identified individually as a first keyword 559a and a second keyword 559b). In one example, the first command 557a can be a command to play music, such as a specific song, album, playlist, etc. In this example, the keywords <NUM> may be one or words identifying one or more zones in which the music is to be played, such as the Living Room and the Dining Room shown in <FIG>. In some examples, the voice utterance portion 557b can include other information, such as detected pauses (e.g., periods of non-speech) between words spoken by a user, as shown in <FIG>. The pauses may demarcate the locations of separate commands, keywords, or other information spoke by the user within the voice utterance portion 557b.

In some embodiments, the media playback system <NUM> is configured to temporarily reduce the volume of audio content that it is playing while detecting the wake word portion 557a. The media playback system <NUM> may restore the volume after processing the voice input <NUM>, as shown in <FIG>. Such a process can be referred to as ducking, examples of which are disclosed in <CIT> and titled "Audio Playback Settings for Voice Interaction".

<FIG> is a functional block diagram showing additional details of the remote computing device(s) <NUM> in <FIG>. In various embodiments, the remote computing device(s) <NUM> may receive voice inputs from one or more of the NMDs <NUM> over the WAN <NUM> shown in <FIG>. For purposes of illustration, selected communication paths of the voice input <NUM> (<FIG>) are represented by arrows in <FIG>. In one embodiment, the voice input <NUM> processed by the remote computing device(s) <NUM> may include the voice utterance portion 557b (<FIG>). In another embodiment, the processed voice input <NUM> may include both the voice utterance portion 557b and the wake word 557a (<FIG>).

The remote computing device(s) <NUM> include a system controller <NUM> comprising one or more processors, an intent engine <NUM>, and a memory <NUM>. The memory <NUM> may be a tangible computer-readable medium configured to store instructions executable by the system controller <NUM> and/or one or more of the playback, network microphone, and/or controller devices <NUM>-<NUM>.

The intent engine <NUM> is configured to process a voice input and determine an intent of the input. In some embodiments, the intent engine <NUM> may be a subcomponent of the system controller <NUM>. The intent engine <NUM> may interact with one or more database(s), such as one or more VAS database(s) <NUM>, to process voice inputs. The VAS database(s) <NUM> may reside in the memory <NUM> or elsewhere, such as in memory of one or more of the playback, network microphone, and/or controller devices <NUM>-<NUM>. In some embodiments, the VAS database(s) <NUM> may be updated for adaptive learning and feedback based on the voice input processing. The VAS database(s) <NUM> may store various user data, analytics, catalogs, and other information for NLU-related and/or other processing.

The remote computing device(s) <NUM> may exchange various feedback, information, instructions, and/or related data with the various playback, network microphone, and/or controller devices <NUM>-<NUM> of the media playback system <NUM>. Such exchanges may be related to or independent of transmitted messages containing voice inputs. In some embodiments, the remote computing device(s) <NUM> and the media playback system <NUM> may exchange data via communication paths as described herein and/or using a metadata exchange channel as described in <CIT>, and titled "Metadata exchange involving a networked playback system and a networked microphone system.

Processing of a voice input by devices of the media playback system <NUM> may be carried out at least partially in parallel with processing of the voice input by the remote computing device(s) <NUM>. Additionally, the speech/text conversion components <NUM> of a network microphone device may convert responses from the remote computing device(s) <NUM> to speech for audible output via one or more speakers.

<FIG> illustrates an example configuration of a media playback system <NUM> in which one or more embodiments disclosed herein may be practiced or implemented. The media playback system <NUM> as shown is associated with an example home environment having several rooms and spaces, such as for example, a master bedroom, an office, a dining room, and a living room. For example, the media playback system <NUM> includes playback devices <NUM> (identified individually as playback devices 102a-<NUM>), controller devices 103a and 103b (collectively "controller devices <NUM>"), and network microphone devices <NUM> ("NMDs"); identified individually as NMDs 104a-<NUM>), and a wired or wireless network router <NUM>.

In some examples, one or more individual playback devices <NUM> can have an on-board (e.g., integrated) NMD, such as one of the playback devices 102a-e, which include corresponding NMDs 104a-e, respectively. In some instances, an NMD can be a stand-alone device, such as the NMD 104f or the NMD 10fg. A stand-alone NMD may omit components, such as a speaker or related electronics, in which case it might not produce audio output or may produce limited audio output (e.g., relatively low quality output relative to the quality of output by a playback device). For instance, a playback device might have more transducers and/or larger transducers (e.g., a woofer) and/or more powerful amplifiers as compared with a stand-alone NMD so as to produce a higher quality output than the stand-alone NMD.

In some examples, one or more NMDs can be assigned to a playback device, a group, and/or or a bonded-set of playback devices. For instance, the NMD 104f may be assigned to the playback device 102a in the living room and/or the playback device 102i in the kitchen. In such implementations, the NMD may be assigned to a single voice assistant service, such as AMAZON® Alexa ® or another voice assistant service. Further details regarding assignment of playback devices and NMDs are described, for example, in: App. No. <CIT>, titled "Default Playback Device Designation;" App. No. <CIT>, titled "Default Playback Devices;" App. No. <CIT>, titled "Audio Response Playback;" and App. No. <CIT>, titled "Determining Direction of Networked Microphone Device Relative to Audio Playback Device.

As discussed above, some embodiments described herein may involve acoustic echo cancellation. <FIG> is a functional block diagram of an acoustic echo cancellation pipeline 800a configured to be implemented within a playback device that includes a NMD, such as NMDs 103a-e. By way of example, the acoustic echo cancellation pipeline 800a is described as being implemented within playback device <NUM> of <FIG>. However, in other implementations, the acoustic echo cancellation pipeline 800a may be implemented in an NMD that is not necessarily a playback device (e.g., a device that doesn't include speakers, or includes relatively low-output speakers configured to provide audio feedback to voice inputs), such as NMDs 103f-g.

In operation, acoustic echo cancellation pipeline 800a is activated when playback device <NUM> is playing back audio content. As noted above, acoustic echo cancellation can be used to remove acoustic echo (i.e., the sound of the audio playback and reflections and/or other acoustic artifacts from the acoustic environment) from the signal captured by microphone(s) of the networked microphone device. When effective, acoustic echo cancellation improves the signal-to-noise ratio of a voice input with respect to other sound within the acoustic environment. In some implementations, when audio playback is paused or otherwise idle, the acoustic echo cancellation pipeline 800a is bypassed or otherwise disabled.

As shown in <FIG>, the microphone array <NUM> is configured to capture a "measured signal," which is an input to the acoustic echo cancellation pipeline 800a. As described above in reference to <FIG> and <FIG>, the microphone array <NUM> can be configured to capture audio within an acoustic environment in an attempt to detect voice inputs (e.g., wake-words and/or utterances) from one or more users. When the playback device <NUM> plays back audio content via speakers <NUM> (<FIG>), the microphone array <NUM> can capture audio that also includes audio signals representing sound produced by speakers <NUM> in playing back the audio content, as well as other sound being produced within the acoustic environment.

At block 870a, the measured signal is pre-processed in advance of acoustic echo cancellation. Pre-processing of the measured signal may involve analog-to-digital conversion of the microphone array signals. Other pre-processing may include sample rate conversion, de-jittering, de-interleaving, or filtering, among other examples. The term "measured signal" is generally used to refer to the signal captured by the microphone array <NUM> before and after any pre-processing.

As shown in <FIG>, another input to the acoustic echo cancellation pipeline 800a is a "reference signal. " The reference signal can represent the audio content being played back by the speakers <NUM> (<FIG>). As shown, the reference signal is routed form the audio processing components <NUM>. In an effort to more closely represent the audio content being played back by the speakers <NUM>, the reference signal is sourced from a point in an audio processing pipeline of the audio processing components <NUM> that closely represents the expected analog audio output of the speakers <NUM>. Since each stage of an audio processing pipeline may introduce artifacts, the point in the audio processing pipeline of the audio processing components <NUM> that closely represents the expected analog audio output of the speakers <NUM> is typically near the end of the pipeline.

As noted above, although the acoustic echo cancellation pipeline 800a is shown by way of example as being illustrated within the playback device <NUM>, the acoustic echo cancellation pipeline 800a may alternatively be implemented within a dedicated NMD such as NMD 103f-g of <FIG>. In such examples, the reference signal may sent from the playback device(s) that are playing back audio content to the NMD, perhaps via a network interface or other communications interface, such as a line-in interface.

At block 870b, the reference signal is pre-processed in advance of acoustic echo cancellation. Pre-processing of the reference signal may involve sample rate conversion, de-jittering, de-interleaving, time-delay, or filtering, among other examples. The term "measured signal" is generally used to refer to the signal captured by the microphone array <NUM> before and after any pre-processing.

Pre-processing the measured signal and the reference signals readies the signals for mixing during acoustic echo cancellation. For instance, since audio content is output by the speakers <NUM> before the microphone array <NUM> captures a representation of that same content, time-delay is introduced to the reference signal to time-align the measured and reference signals. Similarly, since the respective sample rates of analog-to-digital conversation of the analog microphone signals and the reference signal from the audio processing components <NUM> may be different, sample rate conversation of one or both of the signals may convert the signal(s) into the same or otherwise compatible sample rates. In some examples, other similar pre-processing is performed in blocks 870a and 870b to render the measured signals and reference signals compatible.

At block 871a, the measured and reference signals are converted into the short-time Fourier transform domain. Acoustic echo cancellation in the STFT domain may lessen the processing requirements of acoustic echo cancellation as compared with acoustic echo cancellation in other domains, such as the Frequency-Dependent Adaptive Filter ("FDAF") domain. As such, by processing in the STFT domain, additional techniques for acoustic echo cancellation may become practical. However, while acoustic echo cancellation is shown in the STFT domain by way of example, AEC in other domains (e.g., the FDAF domain) can be implemented in alternative examples.

As those of ordinary skill in the art will appreciate, a STFT is a transform used to determine the sinusoidal frequency and phase content of local sections (referred to as "frames" or "blocks") of a signal as it changes over time. To compute STFTs of the measured and reference signals, each signal is divided into a plurality of frames. In an example implementation, each frame is <NUM> milliseconds (ms) long. The number of samples in a <NUM> frame may vary based on the sample rate of the measured and reference signals.

Given a signal x(n), the signal is transformed to the STFT domain by: <MAT> where k is the frequency index, m is the frame index, N is the frame size, R is the frame shift size, wA[n] is an analysis window of size N, and <MAT>.

Referring now to AEC <NUM> (<FIG>), after being converted into the STFT domain, the measured and reference signals are provided as input to the AEC <NUM>, as shown in <FIG>. The acoustic echo cancellation performed by the AEC <NUM> on the measured signal is an iterative process. Each iteration of the AEC processes a respective frame of the measured signal using a respective frame of the reference signal. Such processing includes passing a frame of the reference signal through the adaptive filter <NUM> to yield a frame of a model signal. The adaptive filter <NUM> is intended to transform the reference signal into the measured signal with minimal error. In other words, the model signal is an estimate of the acoustic echo.

To cancel the acoustic echo from the measured signal, the measured signal and the model signal are provided to a redaction function <NUM>. The redaction function <NUM> redacts the model signal from the measured signal, thereby cancelling the estimated acoustic echo from the measured signal yielding an output signal. In some examples, the redaction function <NUM> redacts the model signal from the measured signal by inverting the model signal via inverter <NUM> and mixing the inverted model signal with a frame of the measured signal with mixer <NUM>. In effect, this mixing removes the audio playback (the reference signal) from the measured signal, thereby cancelling the echo (i.e., the audio playback and associated artifacts) from the measured signal. Alternate implementations may use other techniques for redaction.

At block 871b, the output signal of AEC <NUM> is transformed back by applying the inverse STFT. The inverse STFT is applied by: <MAT> where w, [n] is a synthesis window.

After block 871b, the output signal is provided to a voice input processing pipeline at block <NUM>. Voice input processing may involve wake word detection, voice/speech conversion, and/or sending one or more voice utterances to a voice assistant service, among other examples.

Turning now in more detail to internal aspects of the AEC <NUM>, at block <NUM>, the reference signal in the STFT domain is passed through the adaptive filter <NUM>. As noted above, the adaptive filter <NUM> is a transfer function that adapts during each iteration of the AEC 554in an attempt to transform the reference signal into the measured signal with diminishing error. Passing a frame of the reference signal through adaptive filter <NUM> yields a frame of a model signal. The model signal is an estimate of the acoustic echo of the reference signal (i.e., the audio that is being cancelled).

Within examples, adaptive filter <NUM> implements multi-delay adaptive filtering. To illustrate example multi-delay adaptive filtering, let N be the multi-delay filter (MDF) block size, K be the number of blocks and F<NUM>N denote the <NUM>N × <NUM>N Fourier transform matrix, and the frequency-domain signals for frame m are: <MAT> <MAT> <MAT> where d(m) is the modeled signal, e(m) is the modeling error, and Xk(m) is the measured signal. The MDF algorithm then becomes: <MAT> <MAT> with model update: <MAT> and <MAT> where G<NUM> and G<NUM> are matrices which select certain time-domain parts of the signal in the frequency domain, <MAT> and <MAT> The matrix <MAT> is a diagonal approximation of the input power spectral density matrix. To reduce the variance of the power spectrum estimate, the instantaneous power estimate can be substituted by its smoothed version, <MAT> where β is the smoothing term. This example also assumes a fixed step-size (how much the filter is adapted during each iteration) for each partition µ(m) = µ<NUM>I, however the step size may be varied in some implementations.

Example implementations of adaptive filter <NUM> implement cross-band filtering. To illustrate such filtering, let y[n] be the near-end measured signal, which includes the near-end speech and/or noise v[n] mixed with the acoustic echo d[n] = h[n] * x[n], where h[n] is the impulse response of the system, x[n] is the far-end reference signal, and * is the convolution operator. Let x[m] = [x[mR],. x[mR + N - <NUM>]]T be the mth reference signal vector, wA = [WA[<NUM>],. , wA[N - <NUM>]]Tbe the analysis window vector, (F)k+<NUM>,n+<NUM> = <MAT> be the N×N discrete Fourier transform matrix, and x[m] = F(wA∘x[m]) = [X<NUM>[m],. , XN-<NUM>[m]]T be the DFT of the windowed reference signal vector, where ∘ is the Hadamard (element-wise) product operator and {. }T is the transpose operator.

Given a transfer function H, the acoustic echo can be represented in the STFT domain as <MAT> where d[m] is the DFT of the mth frame echo signal, Hi is the ith impulse response matrix (i.e., the filter for the mth iteration of AEC <NUM>), x[m] is the DFT of the mth frame reference siganl, and M is the filter length in the STFT domain.

Given the foregoing, acoustic echo cancellation by AEC <NUM> can be expressed in the STFT domain as: <MAT> where x[m] is the reference signal, <MAT> where y[m] is the measured signal, and <MAT> where e[m] is the output signal. As noted above, the redaction function <NUM> redacts the model signal d̂[m] from the measured signal.

At block <NUM>, an update filter is determined. As noted above, ultimately, the update filter is multiplied by the filter used in the current iteration of the AEC <NUM> to yield the filter for the next iteration of the AEC <NUM>. Generally, during the first iterations of the AEC <NUM>, some error exists in the cancellation of the echo from the measured signal. However, over successive iterations of the AEC <NUM>, this error is diminished. In particular, during each iteration of the AEC554, the adaptive filter <NUM> is updated for the next iteration based on error from the current iteration. In this way, during successive iterations of the AEC <NUM>, the AEC <NUM> mathematically converges to a cancellation of the audio playback by the speakers <NUM> (<FIG>).

In the first iteration of the AEC <NUM>, an initial filter is utilized, as no adaptation has yet occurred. In some implementations, the initial filter is a transfer function representing the acoustic coupling between the speakers <NUM> and the microphones <NUM> in an anechoic chamber. In some embodiments, the initial filter comprises a transfer function generated, for example using measurements performed in an anechoic chamber. The generated transfer function can represent an acoustic coupling between the speakers <NUM> and the microphones <NUM> without any room effect. Such an initial filter could be used in any acoustic environment. Alternatively, in an effort to start the adaptive filter in a state that more closely matches the actual acoustic environment in which the playback device is located, a transfer function representing an acoustic coupling between the speakers <NUM> and the microphones <NUM> may be determined during a calibration procedure that involves microphones <NUM> recording audio output by speakers <NUM> in a quiet room (e.g., with minimal noise). Other initial filters may be used as well, although a filter that poorly represents the acoustic coupling between the speakers <NUM> and the microphones <NUM> may provide a less-optimal starting point for AEC <NUM> and result in to additional iterations of AEC <NUM> before convergence occurs.

In subsequent iterations of the AEC <NUM>, the adaptive filter <NUM> can continue to adapt. During each nth iteration of the AEC , an n+<NUM>th instance of the adaptive filter <NUM> is determined for the next iteration of the AEC <NUM>. In particular, during the nth iteration of the AEC <NUM>, the nth instance of the adaptive filter is multiplied by an nth update filter to yield the n+<NUM>th instance of the adaptive filter. The nth update filter is based on the modelling error of the filter during the nth iteration.

To illustrate, let Ĥ be an adaptive filter matrix. As noted above, the model signal (i.e., the estimated acoustic echo) can be written as <MAT> and the adaptive filter matrix can be updated from iteration to iteration using <MAT> where ΔĤi[m] is an update matrix for the filter coefficients matrix and <MAT> is a matrix that selects the <NUM> + <NUM> diagonal bands. P is a permutation matrix defined as <MAT> For a filter having K blocks, to improve the modeling accuracy, <NUM> cross-terms, or <NUM> off-diagonal bands are added around the main diagonal terms of H without increasing the computational complexity to an impractical extent. In this example, Ĥ has <NUM> + <NUM> diagonal bands. The matrix G limits the number of crossband filters that are useful for system identification in the STFT domain since increasing the number of crossband filters does not necessarily lead to a lower steady-state error.

As noted above, the nth update filter is based on the modelling error of the filter during the nth iteration. Using a least mean squares algorithm, the update filter is given by <MAT> where e[m] = y[m] - d̂[m] is the error signal vector in the STFT domain, µ > <NUM> is a step-size, and {·}H is the Hermitian transpose operator.

As an alternative to the least mean squares, the AEC <NUM> may implement a normalized least mean squares algorithm to improve noise-robustness. Under an NMLS algorithm, the update filter is given by: <MAT> where the reference signal is normalized by its signal power before being multiplied by the error signal. As noted above, during an nth iteration, the update filter is multiplied by the adaptive filter for the nth iteration to yield the adaptive filter for the n+<NUM> iteration. Given the example above, the adaptive filter is represented as: <MAT>.

In example implementations, acoustic echo cancellation pipeline 800a may be integrated into an audio processing pipeline that includes additional audio processing of microphone-captured audio such as beam forming, blind source separation, and frequency gating before the microphone-captured audio is processed as a voice input to a voice service. Second Example Acoustic Echo Cancellation Pipeline.

<FIG> is a functional block diagram of an example acoustic echo cancellation pipeline 800b that includes two acoustic echo cancellers (as encompassed by the claimed invention). In particular, the audio processing pipeline 800b includes the AEC <NUM> and a tone interference canceller (TIC) <NUM>. In operation, audio processing pipeline 800b runs either the AEC <NUM> or the TIC <NUM> when cancelling acoustic echo. Like the acoustic echo cancellation pipeline 800a, the acoustic echo cancellation pipeline 800b is configured to be implemented within playback device <NUM> of <FIG>.

As shown in <FIG>, the acoustic echo cancellation pipeline 800b utilizes a de-multiplexer (de-mux) 881a, a de-mux 881b, and a multiplexer (mux) <NUM> to switch between the AEC <NUM> and TIC <NUM>. In particular, the de-mux 881a and the de-mux 881b route the measured signal (from microphone array <NUM> of <FIG>) and the reference signal (from audio processing components <NUM> of <FIG>), respectively, to either the AEC <NUM> or the TIC <NUM> based on control signal(s) from a AEC/TIC Control <NUM>. Similarly, the mux <NUM> routes output to the voice input processing <NUM> (<FIG>) from either the AEC <NUM> or the TIC <NUM> based on control signal(s) from the AEC/TIC Control <NUM>. In this manner, either the AEC <NUM> or the TIC <NUM> can be activated to cancel acoustic echo.

The AEC <NUM> is configured to cancel audio output from speakers <NUM> (<FIG>) in a full audible frequency spectrum. In some examples, the full audible frequency spectrum includes frequencies generally considered audible to human ears (e.g., <NUM> to <NUM>,<NUM>). Alternatively, some implementations of the AEC <NUM> are configured to filter a frequency spectrum that includes frequencies generally considered within the range of human speech (e.g., <NUM> to <NUM>). Minor adjustments to these frequency ranges are possible as well. Acoustic echo having content across such frequency ranges is referred to as "full-range" acoustic echo.

Like the AEC <NUM>, the TIC <NUM> is an acoustic echo canceller and may include generally similar components and have similar functionality to the AEC <NUM>. However, in contrast to the AEC <NUM>, the TIC <NUM> is configured to cancel audio output from the speakers <NUM> in the frequency bands of the full audible frequency spectrum in which the acknowledgment tone has content. Example acknowledgment tones, being tones, may have content in relatively few frequency bins. Further, as compared with full range audio content that is user-selectable, the frequency bands of the full audible frequency spectrum in which a given acknowledgment tone has content are known (e.g., pre-determined), perhaps by playback device <NUM> or during manufacturing. Alternatively, the frequency bands of the full audible frequency spectrum in which a given acknowledgment tone has content may be determined by the playback device, perhaps in advance of using the TIC <NUM> to cancel the acknowledgment tone.

To illustrate, as described above with respect to AEC <NUM>, example filters (e.g., adaptive filter <NUM>) may filter in the STFT domain. When filtering certain frequency bands (frequency "bins"), the filter is shorter than when filtering the full audible frequency spectrum. For instance, referring to the example above, the transfer function H can be shorter (i.e., include fewer elements with transfer functions). This reduces the complexity of the second sound cancellation process, allowing the TIC <NUM> to converge significantly faster than the AEC <NUM>, which has a longer filter so as to be able to cancel acoustic echo across a significantly larger frequency range (i.e., the full audible frequency spectrum). Note that attempting to cancel full range acoustic echo with the TIC <NUM> will typically not result in effective acoustic echo cancellation, as the TIC <NUM> is configured to cancel acoustic echo in a subset of the full range by way of its filter.

As noted above, switching between the AEC <NUM> and the TIC <NUM> is performed using the de-mux 881a, the de-mux 881b, and the mux <NUM> based on control signal(s) from the AEC/TIC Control <NUM>. This switching mechanism is shown by way of example. In some examples, equivalent switching is implemented programmatically, such as in implementations where the AEC <NUM> and the TIC <NUM> are implemented in a processor (e.g., a digital signal processor ("DSP") of playback device <NUM> (<FIG>).

In <FIG>, the AEC/TIC Control <NUM> operates the de-mux 881a, the de-mux 881b, and mux <NUM> to switch between AEC <NUM> and TIC <NUM> based on whether the speakers <NUM> are playing audio content. In particular, when playback device <NUM> begins playing audio content via speakers <NUM>, the AEC/TIC Control <NUM> activates the AEC <NUM> to cancel acoustic echo from the playback of the audio content. Conversely, if the speakers <NUM> are inactive (i.e., idle), then the AEC <NUM> is inactive. Accordingly, when the speakers <NUM> are inactive and the playback device <NUM> detects a wake word (e.g., via a wake word detector <NUM> of <FIG>), the AEC/TIC Control <NUM> activates the TIC <NUM> to cancel acoustic echo from the playback of an audible tone played back in acknowledgment of the detected wake word.

As noted above, being an iterative process, the AEC <NUM> takes some time to converge to an effective cancellation of acoustic echo from an inactive state (e.g., ~<NUM> or more, depending on the processing capabilities and algorithm implemented). As such, if the AEC <NUM> were activated instead of the TIC <NUM> when the speakers <NUM> are inactive and the playback device <NUM> detects a wake word, the AEC <NUM> is unlikely to converge in time to effectively cancel acoustic echo of an audible tone coming shortly after a wake word (in acknowledgment of detecting the wake word). However, as described above, the TIC <NUM> is designed to converge more quickly than AEC <NUM>, and as such will typically be able to converge in time to cancel the acoustic echo of the audible tone in acknowledgment of the wake word.

Under certain reset conditions, the AEC/TIC Reset <NUM> will reset the input states of the AEC/TIC Control <NUM>. Input states may include event detection (i.e., the detection of a wake word) and the presence or absence of audio playback via the speakers <NUM>. This allows the AEC/TIC Control <NUM> to select either (a) the AEC <NUM> or (b) the TIC <NUM> under new input conditions, such as another wake word or a change in playback status of the playback device <NUM> (<FIG>).

In certain conditions, both the AEC <NUM> and the TIC <NUM> may be bypassed. Namely, when the speakers <NUM> are inactive, the AEC <NUM> may be inactive as well, as there is no full-range acoustic echo of the playback device <NUM> to cancel. Further, the TIC <NUM> may be inactive as well until activated by the detection of a wake word. In such conditions, the speakers <NUM> are expected to be remain idle (until a wake word is detected or playback of audio content is started) and both the AEC <NUM> and the TIC <NUM> can be bypassed, as there is no acoustic echo to cancel (either from playback of the acknowledgment tone or from playback of other audio content). Note that if the speakers <NUM> return to an idle state after the TIC <NUM> cancels the acoustic echo of an acknowledgment tone, then the TIC <NUM> can be bypassed.

In some instances, the AEC/TIC Control <NUM> and the AEC/TIC Reset <NUM> are configured as a state machine. <FIG> is a functional block diagram of an example state machine <NUM>, which is configured to be implemented by the playback device <NUM> (<FIG>), to select between AEC <NUM> and TIC <NUM>. As shown in <FIG>, the state machine <NUM> starts in an initial condition at block <NUM>. In the initial condition, the speakers <NUM> (<FIG>) are inactive and there is no wake word acknowledgement tone to be cancelled. These states may be represented as variables, such, "driversIdle == True" to represent that the speakers <NUM> are idle and "ackTone == False" to represent that there no wake word acknowledgement tone to be cancelled. In such states, both the AEC <NUM> and the TIC <NUM> may be bypassed.

However, during operation, the states may change. In particular, at block <NUM>, the state machine <NUM> determines whether the speakers <NUM> are active or inactive. In some examples, determines whether the speakers <NUM> are active or inactive involves determining whether an audio signal is passing through an audio playback pipeline (e.g., an audio playback pipeline implemented by the audio processing components <NUM> and/or the audio amplifiers <NUM>, perhaps in a DSP). Alternatively, determining whether speakers <NUM> are active or inactive involves referencing a state variable (e.g., "driversIdle") that is maintained in the memory <NUM> (<FIG>) by the playback device <NUM> to indicate the current state of the speakers <NUM>. Other examples are possible as well.

If the speakers <NUM> are inactive, the state machine <NUM> proceeds to block <NUM>, where the state machine <NUM> determines whether an acknowledgement tone is about to be played. Determining whether determines an acknowledgement tone is about to be played may involve referencing a state variable (e.g., "ackTone") that is maintained in the memory <NUM> by the playback device <NUM>. A wake word detector (e.g., the wake word detector <NUM>) may set ackTone to "true" in response to detecting a wake word in captured audio.

However, if the speakers <NUM> are active, the state machine <NUM> proceeds to block 908a, where the AEC <NUM> is run to cancel the acoustic echo of audio content being played back by the speakers <NUM>. As noted above, the AEC <NUM> is configured to cancel full-range acoustic echo. If the wake word detector <NUM> detects a wake word (and the playback device <NUM> responsively outputs an audible tone in acknowledgment) while the speakers <NUM> are already active playing other audio content, then the AEC <NUM> cancels the acoustic echo of the audible tone (perhaps in addition to the acoustic echo of the audio content).

Referring back to block <NUM>, if an acknowledgement tone is about to be played (and the speakers <NUM> are idle), the state machine <NUM> proceeds to block 908b, where the TIC <NUM> is run to cancel the acoustic echo of the acknowledgement tone when the tone is played back by the speakers <NUM>. To effectively cancel the acknowledgement tone, the TIC <NUM> is run prior to the acknowledgement tone being played back by the speakers <NUM>. For instance, the TIC <NUM> is activated at least one frame prior to the acknowledgement tone being played back by speakers <NUM>, where the TIC <NUM> implements an acoustic echo cancellation algorithm that processes input signals on a frame-by-frame basis, as described with reference to the AEC <NUM> in <FIG>. In some implementations, the TIC <NUM> is activated multiple frames prior to the acknowledgement tone being played back by the speakers <NUM>, to provide more iterations for the TIC <NUM> to converge.

If no acknowledgement tone is about to be played (and the speakers <NUM> are idle), the state machine <NUM> proceeds to block <NUM> and bypasses both the AEC <NUM> and the TIC <NUM>. State machine <NUM> then returns to the initial condition at block <NUM>. The state machine <NUM> may loop through blocks <NUM>, <NUM>, <NUM>, and <NUM>, thereby bypassing the AEC <NUM> and the TIC <NUM> while the speakers <NUM> remain idle and no wake word is detected (e.g., while "driversIdle" == true and "ackTone" == false).

At block 912a, the state machine <NUM> may determine whether a reset condition for AEC <NUM> has occurred. Example reset conditions of block 912a include the speakers <NUM> becoming inactive (e.g., "driversIdle" being set to true) or the expiration of a timer. If a reset condition is detected, the state machine <NUM> returns to block <NUM>. However, if no reset condition is detected, then the state machine <NUM> returns to block 908a to continue running the AEC <NUM> (e.g., if audio content playback is on-going).

Similarly, at block 912b, the state machine <NUM> may determine whether a reset condition for the TIC <NUM> has occurred. For instance, an example reset condition for the TIC <NUM> is completion of the process of cancelling the acoustic echo of the acknowledgment tone using the TIC <NUM>. In particular, when the TIC <NUM> completes the process of cancelling the acoustic echo of the acknowledgment tone, the TIC <NUM> is reset in block 912b so that the TIC <NUM> can return to an idle state (and be bypassed) if appropriate.

Other reset conditions are related to audio playback. For example, a reset condition may be that the speakers <NUM> becoming active playing audio content (e.g., "driversIdle" being set to false). In such circumstances, the state machine <NUM> should return to the initial condition, so that AEC <NUM> can be run to cancel full-range acoustic echo. Alternatively, the TIC <NUM> may be reset upon the expiration of a timer. A timer may limit the length of the TIC <NUM> to allow the AEC <NUM> to be run instead of the TIC <NUM> if audio playback (other than the acknowledgment tone) starts during playback of the acknowledgment tone.

Some reset conditions are related to user input. For instance, playback of the acknowledgment tone may be deactivated via a voice command, user input via a control device (e.g., control device <NUM> of <FIG>), or user input on the playback device <NUM>, among other examples. In such cases, the TIC <NUM> need not be run. As another example, the voice input (containing the wake word) may be cancelled via a control device, or user input on the playback device <NUM>, among other examples. In such cases, an acknowledgment tone is unnecessary. In a further example, microphones <NUM> of playback device <NUM> may be muted, which implies a cancellation of voice inputs. Other examples are possible as well.

As discussed above, embodiments described herein involve acoustic echo cancellation. <FIG> is a flow diagram of an example implementation <NUM> by which a system (e.g., the claimed playback device <NUM>) performs acoustic echo cancellation using either a first sound canceller or a second sound canceller. In some embodiments, the implementation <NUM> can comprise instructions stored on a memory (e.g., the memory <NUM> and/or the memory <NUM>) and executable by one or more processors (e.g., the processor <NUM> and/or the processor <NUM>).

At block <NUM>, the implementation <NUM> captures first audio within an acoustic environment. For instance, the implementation <NUM> can be configured to capture audio within an acoustic environment via a playback device that includes an NMD (e.g., the playback device 102a-e of <FIG>) having one or more microphones (e.g., the microphones <NUM> of <FIG>). Alternatively, the implementation <NUM> can be configured to capture audio within an acoustic environment via a NMD (e.g., NMDs 102f-g of <FIG>) that has a network connection with a playback device.

Capturing audio may involve recording audio within an acoustic environment, as well as processing of the recorded audio, such as analog-to-digital conversation. The implemetnation may capture audio in an effort to detect voice inputs, such as the voice input <NUM> of <FIG>. As described in <FIG>, the example voice input <NUM> may include a wake word portion 557a and a voice utterance portion 557b. In examples, the first audio captured by the system might include the wake word portion 557a of the voice input <NUM>. The acoustic environment may be defined as sound within detectable range of the microphone.

In some instances, the implementation <NUM> may capture audio within an acoustic environment while one or more playback devices are also playing back audio content within the acoustic environment. In such instances, the captured first audio includes audio signals representing the acoustic echo caused by playback of the audio content in the acoustic environment. The captured audio may also include other noise present in the acoustic environment.

At block <NUM>, the implementation <NUM> determines whether one or more speakers are playing back audio content or idle. For instance, the implementation <NUM> is configured to determine whether the speakers <NUM> are playing back audio content or idle via the playback device <NUM>. Determining whether the speakers <NUM> are playing back audio content may involve determining whether an audio signal is passing through the audio playback pipeline of the playback device <NUM> (e.g., through a DSP of the playback device <NUM>), as described above with reference to block <NUM> of <FIG>.

Alternatively, determining whether the speakers <NUM> are playing back audio content may involve the implementation <NUM> referencing a state variable, as also described above with reference to block <NUM> of <FIG>. In some examples, the playback device <NUM> may reference such a state variable from the memory <NUM>. Alternatively, one or more of the NMDs 103a-f may reference such a state variable from the memory <NUM> of one of the playback device 102a-<NUM> (e.g., using the network interface <NUM>). Yet further, one or more of the NMDs 103a-f may maintain representations of state variables indicating states of the playback device(s) 102a-<NUM> in the memory <NUM>, and may reference the state variable from the memory <NUM>. Other examples are possible as well.

In <FIG>, at block <NUM>, the implementation <NUM> detects a wake word. For instance, the system detects a wake word within the captured first audio content. By way of example, the implementation <NUM> may detect the wake word within the captured first audio content using the wake word detector <NUM> of the NMD <NUM> (<FIG>). In some examples, detecting the wake word triggers an event detector, which ultimately causes the implementation <NUM> to perform one or more responsive functions, such as outputting an acknowledgment tone and/or activating a sound canceller.

At block <NUM>, the implementation <NUM> activates either (A) a first sound canceller or (B) a second sound canceller. For instance, when the one or more speakers <NUM> are playing back audio content, the implementation <NUM> activates a first sound canceller (e.g., the AEC <NUM>) configured to cancel audio output from the one or more speakers <NUM> in a full audible frequency spectrum. Alternatively, when the one or more speakers are idle, the implementation <NUM> activates a second sound canceller (e.g., the TIC <NUM>) in response to detecting the wake word. The second sound canceller is configured to cancel audio output from the one or more speakers in the frequency bands of the full audible frequency spectrum in which the audible tone in acknowledgment of the detected wake word has content.

In some instances, the implementation <NUM> includes an acoustic echo cancellation pipeline, such as acoustic echo cancellation pipeline 800b of <FIG>. In such instances, the implementation <NUM> may switch between the first sound canceller (e.g., the AEC <NUM>) or the second sound canceller (e.g., TIC <NUM>) based on control signals from the AEC/TIC Control <NUM>. Further, the implementation <NUM> may implement one or more state machines (e.g., the state machine <NUM> of <FIG>) to determine when to activate either the first sound canceller, the second sound canceller, or to bypass both sound cancellers.

At block <NUM>, the implementation <NUM> outputs the acknowledgment tone. For instance, the implementation <NUM> outputs an acknowledgment tone via the speakers <NUM> of the playback device <NUM> in response to detecting the wake word in block <NUM>, thereby acknowledging detection of the wake word. In some examples, a digital representation (e.g., a file) of the acknowledgment tone is stored in memory <NUM>. Alternatively, the acknowledgment tone is streamed from a remote computing system, such as a server of a streaming content service, or the control device <NUM> (<FIG>).

At block <NUM>, the implementation <NUM> captures second audio within the acoustic environment. For example, the implementation <NUM> can be configured to capture second audio within an acoustic environment via a playback device or an NMD, as described above with respect to block <NUM>. Capturing audio may involve recording audio within an acoustic environment, as well as processing of the recorded audio, such as analog-to-digital conversation.

As described in <FIG>, the example voice input <NUM> may include a wake word portion 557a and a voice utterance portion 557b. Given that the implementation <NUM> outputs the acknowledgment tone in response to detecting the wake word in block <NUM>, the acknowledgment tone may overlap with a voice utterance portion 557b following a wake word portion 557a. Such overlap may interfere with capturing and/or processing of the voice utterance portion 557b. As such, the implementation <NUM> has enabled the AEC <NUM> or the TIC <NUM> to cancel the acoustic echo of the acknowledgment tone from the captured audio.

At block <NUM>, the implementation <NUM> cancels the acknowledgment tone from the captured second audio using the activated sound canceller. In particular, the implementation <NUM> cancels the acknowledgment tone using the sound canceller activated in block <NUM>. In an example, the implementation <NUM> provides the captured second audio and the acknowledgment tone as measured and reference signals, respectively, to the TIC <NUM>, which then provides an output signal with the acoustic echo of the acknowledgment tone removed. In another example, the implementation <NUM> provides the captured second audio and a compound audio signal (including the acknowledgment tone and audio content being played back by playback <NUM>) as measured and reference signals, respectively, to AEC <NUM>, which then provides an output signal with the acoustic echo of the acknowledgment tone and the acoustic echo of the audio content being played back by playback <NUM> removed.

In some examples, the implementation <NUM> may perform additional functions. Some examples functions are provided to illustrate examples. Such examples should not be considered limiting.

In some examples, the implementation <NUM> identifies a set of frequency bands of the full audible frequency spectrum in which an audible tone in acknowledgment of the detected wake-word has content.

If the acknowledgment tone is known and static, the second sound canceller (e.g., the TIC <NUM>) can be pre-configured to process only the set of frequency bins in which the audible tone has content. Other examples are possible as well.

As noted above, in some examples, detecting the wake word triggers an event detector, which ultimately causes the implementation <NUM> to perform one or more responsive functions. In some examples, this event detector is reset upon detecting one or more reset events, which allows the implementation <NUM> to select a different sound canceller if different conditions arise. Detecting the reset event may include one of: (i) cancelling the audible tone from the captured second audio using the activated sound canceller; (ii) expiration of a timer on the activated sound canceller; (iii) initiation of audio content playback via the one or more speakers; (iv) cancellation of a voice input corresponding to the wake-word detected within the captured first audio content via a control interface; (v) muting of the one or more microphones; and (iv) de-activation of the audible tone via the control interface. Other example reset events are described in connection with blocks 912a and 912b of <FIG>.

The specification is presented largely in terms of illustrative environments, systems, procedures, steps, logic blocks, processing, and other symbolic representations that directly or indirectly resemble the operations of data processing devices coupled to networks. These process descriptions and representations are typically used by those skilled in the art to most effectively convey the substance of their work to others skilled in the art. Numerous specific details are set forth to provide a thorough understanding of the present disclosure. However, it is understood to those skilled in the art that certain embodiments of the present disclosure can be practiced without certain, specific details. In other instances, well known methods, procedures, components, and circuitry have not been described in detail to avoid unnecessarily obscuring aspects of the embodiments. Accordingly, the scope of the present disclosure is defined by the appended claims rather than the forgoing description of embodiments.

Claim 1:
A method for a playback device comprising one or more speakers (<NUM>), the method comprising:
capturing, via one or more microphones (<NUM>), a first portion of a first voice input;
detecting, within the first portion of the first voice input, a wake word for a voice assistant service; the method being characterised by further comprising:
when one or more speakers of the playback device are currently idle, activating a tone interference canceller (<NUM>) in response to detection of the wake word within the first portion of the first voice input, wherein if one or more speakers of the playback device are currently active in playing audio, the playback device is configured for activating an echo canceller (<NUM>) configured to cancel audio output from the one or more speakers in a full-range of a frequency spectrum;
in response to detection of the wake word within the first portion of the first voice input and after activating the tone interference canceller (<NUM>), outputting, via the one or more speakers, an audible tone in acknowledgement of the detected wake word, wherein the audible tone has content in one or more known frequency bands in the full-range of the frequency spectrum;
capturing, via the one or more microphones (<NUM>), a second portion of the first voice input; and
cancelling, via the activated tone interference canceller (<NUM>), the one or more known frequency bands in the audible tone from the captured second portion of the first voice input.