Patent Grant
9747920
In audio systems, automatic echo cancellation (AEC) refers to techniques that are used to recognize when a system has recaptured sound via a microphone after some delay that the system previously output via a speaker. Systems that provide AEC subtract a delayed version of the original audio signal from the captured audio, producing a version of the captured audio that ideally eliminates the “echo” of the original audio signal, leaving only new audio information. For example, if someone were singing karaoke into a microphone while prerecorded music is output by a loudspeaker, AEC can be used to remove any of the recorded music from the audio captured by the microphone, allowing the singer's voice to be amplified and output without also reproducing a delayed “echo” the original music. As another example, a media player that accepts voice commands via a microphone can use AEC to remove reproduced sounds corresponding to output media that are captured by the microphone, making it easier to process input voice commands.
For a more complete understanding of the present disclosure, reference is now made to the following description taken in conjunction with the accompanying drawings.
Typically, a conventional Acoustic Echo Cancellation (AEC) system may remove audio output by a loudspeaker from audio captured by the system's microphone(s) by subtracting a delayed version of the originally transmitted audio. However, in stereo and multi-channel audio systems that include wireless or network-connected loudspeakers and/or microphones, a major cause of problems is when there are differences between the signal sent to a loudspeaker and a signal played at the loudspeaker. As the signal sent to the loudspeaker is not the same as the signal played at the loudspeaker, the signal sent to the loudspeaker is not a true reference signal for the AEC system. For example, when the AEC system attempts to remove the audio output by the loudspeaker from audio captured by the system's microphone(s) by subtracting a delayed version of the originally transmitted audio, the audio captured by the microphone is subtly different than the audio that had been sent to the loudspeaker.
There may be a difference between the signal sent to the loudspeaker and the signal played at the loudspeaker for one or more reasons. A first cause is a difference in clock synchronization (e.g., clock offset) between loudspeakers and microphones. For example, in a wireless “surround sound” 5.1 system comprising six wireless loudspeakers that each receive an audio signal from a surround-sound receiver, the receiver and each loudspeaker has its own crystal oscillator which provides the respective component with an independent “clock” signal. Among other things that the clock signals are used for is converting analog audio signals into digital audio signals (“A/D conversion”) and converting digital audio signals into analog audio signals (“D/A conversion”). Such conversions are commonplace in audio systems, such as when a surround-sound receiver performs A/D conversion prior to transmitting audio to a wireless loudspeaker, and when the loudspeaker performs D/A conversion on the received signal to recreate an analog signal. The loudspeaker produces audible sound by driving a “voice coil” with an amplified version of the analog signal.
A second cause is that the signal sent to the loudspeaker may be modified based on compression/decompression during wireless communication, resulting in a different signal being received by the loudspeaker than was sent to the loudspeaker. A third case is non-linear post-processing performed on the received signal by the loudspeaker prior to playing the received signal. A fourth cause is buffering performed by the loudspeaker, which could create unknown latency, additional samples, fewer samples or the like that subtly change the signal played by the loudspeaker.
To perform Acoustic Echo Cancellation (AEC) without knowing the signal played by the loudspeaker, devices, systems and methods may perform audio beamforming on a signal received by the microphones and may determine a reference signal and a target signal based on the audio beamforming. For example, the system may receive audio input and separate the audio input into multiple directions. The system may detect a strong signal associated with a speaker and may set the strong signal as a reference signal, selecting another direction as a target signal. In some examples, the system may determine a speech position (e.g., near end talk position) and may set the direction associated with the speech position as a target signal and an opposite direction as a reference signal. If the system cannot detect a strong signal or determine a speech position, the system may create pairwise combinations of opposite directions, with an individual direction being used as a target signal and a reference signal. The system may remove the reference signal (e.g., audio output by the loudspeaker) to isolate speech included in the target signal.
To isolate the additional sounds from the reproduced sounds, the device 102 may include an adaptive beamformer 104 that may perform audio beamforming on the echo signals 120 to determine a target signal 122 and a reference signal 124. For example, the adaptive beamformer 104 may include a fixed beamformer (FBF) 105, a multiple input canceler (MC) 106 and/or a blocking matrix (BM) 107. The FBF 105 may be configured to form a beam in a specific direction so that a target signal is passed and all other signals are attenuated, enabling the adaptive beamformer 104 to select a particular direction. In contrast, the BM 107 may be configured to form a null in a specific direction so that the target signal is attenuated and all other signals are passed. The adaptive beamformer 104 may generate fixed beamforms (e.g., outputs of the FBF 105) or may generate adaptive beamforms using a Linearly Constrained Minimum Variance (LCMV) beamformer, a Minimum Variance Distortionless Response (MVDR) beamformer or other beamforming techniques. For example, the adaptive beamformer 104 may receive audio input, determine six beamforming directions and output six fixed beamform outputs and six adaptive beamform outputs. In some examples, the adaptive beamformer 104 may generate six fixed beamform outputs, six LCMV beamform outputs and six MVDR beamform outputs, although the disclosure is not limited thereto. Using the adaptive beamformer 104 and techniques discussed below, the device 102 may determine the target signal 122 and the reference signal 124 to pass to an acoustic echo cancellation (AEC) 108. The AEC 108 may remove the reference signal (e.g., reproduced sounds) from the target signal (e.g., reproduced sounds and additional sounds) to remove the reproduced sounds and isolate the additional sounds (e.g., speech) as audio output 126.
To illustrate, in some examples the device 102 may use outputs of the FBF 105 as the target signal 122. For example, the outputs of the FBF 105 may be shown in equation (1):
Target=s+z+noise (1)
where s is speech (e.g., the additional sounds), z is an echo from the signal sent to the loudspeaker (e.g., the reproduced sounds) and noise is additional noise that is not associated with the speech or the echo. In order to attenuate the echo (z), the device 102 may use outputs of the BM 107 as the reference signal 124, which may be shown in equation 2:
Reference=z+noise (2)
By removing the reference signal 124 from the target signal 122, the device 102 may remove the echo and generate the audio output 126 including only the speech and some noise. The device 102 may use the audio output 126 to perform speech recognition processing on the speech to determine a command and may execute the command. For example, the device 102 may determine that the speech corresponds to a command to play music and the device 102 may play music in response to receiving the speech.
In some examples, the device 102 may associate specific directions with the reproduced sounds and/or speech based on features of the signal sent to the loudspeaker. Examples of features includes power spectrum density, peak levels, pause intervals or the like that may be used to identify the signal sent to the loudspeaker and/or propagation delay between different signals. For example, the adaptive beamformer 104 may compare the signal sent to the loudspeaker with a signal associated with a first direction to determine if the signal associated with the first direction includes reproduced sounds from the loudspeaker. When the signal associated with the first direction matches the signal sent to the loudspeaker, the device 102 may associate the first direction with a wireless speaker. When the signal associated with the first direction does not match the signal sent to the loudspeaker, the device 102 may associate the first direction with speech, a speech position, a person or the like.
As illustrated in
The device 102 may determine the target signal and the reference signal using multiple techniques, which are discussed in greater detail below. For example, the device 102 may use a first technique when the device 102 detects a clearly defined speaker signal, a second technique when the device 102 doesn't detect a clearly defined speaker signal but does identify a speech position and/or a third technique when the device 102 doesn't detect a clearly defined speaker signal or a speech position. Using the first technique, the device 102 may associate the clearly defined speaker signal with the reference signal and may select any or all of the other directions as the target signal. For example, the device 102 may generate a single target signal using all of the remaining directions for a single loudspeaker or may generate multiple target signals using portions of remaining directions for multiple loudspeakers. Using the second technique, the device 102 may associate the speech position with the target signal and may select an opposite direction as the reference signal. Using the third technique, the device 102 may select multiple combinations of opposing directions to generate multiple target signals and multiple reference signals.
The device 102 may remove (140) an echo from the target signal by removing the reference signal to isolate speech or additional sounds and may output (142) audio data including the speech or additional sounds. For example, the device 102 may remove music (e.g., reproduced sounds) played over the loudspeakers 114 to isolate a voice command input to the microphones 118.
The device 102 may include a microphone array having multiple microphones 118 that are laterally spaced from each other so that they can be used by audio beamforming components to produce directional audio signals. The microphones 118 may, in some instances, be dispersed around a perimeter of the device 102 in order to apply beampatterns to audio signals based on sound captured by the microphone(s) 118. For example, the microphones 118 may be positioned at spaced intervals along a perimeter of the device 102, although the present disclosure is not limited thereto. In some examples, the microphone(s) 118 may be spacedon a substantially vertical surface of the device 102 and/or a top surface of the device 102. Each of the microphones 118 is omnidirectional, and beamforming technology is used to produce directional audio signals based on signals from the microphones 118. In other embodiments, the microphones may have directional audio reception, which may remove the need for subsequent beamforming.
In various embodiments, the microphone array may include greater or less than the number of microphones 118 shown. Speaker(s) (not illustrated) may be located at the bottom of the device 102, and may be configured to emit sound omnidirectionally, in a 360 degree pattern around the device 102. For example, the speaker(s) may comprise a round speaker element directed downwardly in the lower part of the device 102.
Using the plurality of microphones 118 the device 102 may employ beamforming techniques to isolate desired sounds for purposes of converting those sounds into audio signals for speech processing by the system. Beamforming is the process of applying a set of beamformer coefficients to audio signal data to create beampatterns, or effective directions of gain or attenuation. In some implementations, these volumes may be considered to result from constructive and destructive interference between signals from individual microphones in a microphone array.
The device 102 may include an adaptive beamformer 104 that may include one or more audio beamformers or beamforming components that are configured to generate an audio signal that is focused in a direction from which user speech has been detected. More specifically, the beamforming components may be responsive to spatially separated microphone elements of the microphone array to produce directional audio signals that emphasize sounds originating from different directions relative to the device 102, and to select and output one of the audio signals that is most likely to contain user speech.
Audio beamforming, also referred to as audio array processing, uses a microphone array having multiple microphones that are spaced from each other at known distances. Sound originating from a source is received by each of the microphones. However, because each microphone is potentially at a different distance from the sound source, a propagating sound wave arrives at each of the microphones at slightly different times. This difference in arrival time results in phase differences between audio signals produced by the microphones. The phase differences can be exploited to enhance sounds originating from chosen directions relative to the microphone array.
Beamforming uses signal processing techniques to combine signals from the different microphones so that sound signals originating from a particular direction are emphasized while sound signals from other directions are deemphasized. More specifically, signals from the different microphones are combined in such a way that signals from a particular direction experience constructive interference, while signals from other directions experience destructive interference. The parameters used in beamforming may be varied to dynamically select different directions, even when using a fixed-configuration microphone array.
A given beampattern may be used to selectively gather signals from a particular spatial location where a signal source is present. The selected beampattern may be configured to provide gain or attenuation for the signal source. For example, the beampattern may be focused on a particular user's head allowing for the recovery of the user's speech while attenuating noise from an operating air conditioner that is across the room and in a different direction than the user relative to a device that captures the audio signals.
Such spatial selectivity by using beamforming allows for the rejection or attenuation of undesired signals outside of the beampattern. The increased selectivity of the beampattern improves signal-to-noise ratio for the audio signal. By improving the signal-to-noise ratio, the accuracy of speaker recognition performed on the audio signal is improved.
The processed data from the beamformer module may then undergo additional filtering or be used directly by other modules. For example, a filter may be applied to processed data which is acquiring speech from a user to remove residual audio noise from a machine running in the environment.
The beampattern 202 may exhibit a plurality of lobes, or regions of gain, with gain predominating in a particular direction designated the beampattern direction 204. A main lobe 206 is shown here extending along the beampattern direction 204. A main lobe beam-width 208 is shown, indicating a maximum width of the main lobe 206. In this example, the beampattern 202 also includes side lobes 210, 212, 214, and 216. Opposite the main lobe 206 along the beampattern direction 204 is the back lobe 218. Disposed around the beampattern 202 are null regions 220. These null regions are areas of attenuation to signals. In the example, the person 10 resides within the main lobe 206 and benefits from the gain provided by the beampattern 202 and exhibits an improved SNR ratio compared to a signal acquired with non-beamforming. In contrast, if the person 10 were to speak from a null region, the resulting audio signal may be significantly reduced. As shown in this illustration, the use of the beampattern provides for gain in signal acquisition compared to non-beamforming. Beamforming also allows for spatial selectivity, effectively allowing the system to “turn a deaf ear” on a signal which is not of interest. Beamforming may result in directional audio signal(s) that may then be processed by other components of the device 102 and/or system 100.
While beamforming alone may increase a signal-to-noise (SNR) ratio of an audio signal, combining known acoustic characteristics of an environment (e.g., a room impulse response (RIR)) and heuristic knowledge of previous beampattern lobe selection may provide an even better indication of a speaking user's likely location within the environment. In some instances, a device includes multiple microphones that capture audio signals that include user speech. As is known and as used herein, “capturing” an audio signal includes a microphone transducing audio waves of captured sound to an electrical signal and a codec digitizing the signal. The device may also include functionality for applying different beampatterns to the captured audio signals, with each beampattern having multiple lobes. By identifying lobes most likely to contain user speech using the combination discussed above, the techniques enable devotion of additional processing resources of the portion of an audio signal most likely to contain user speech to provide better echo canceling and thus a cleaner SNR ratio in the resulting processed audio signal.
To determine a value of an acoustic characteristic of an environment (e.g., an RIR of the environment), the device 102 may emit sounds at known frequencies (e.g., chirps, text-to-speech audio, music or spoken word content playback, etc.) to measure a reverberant signature of the environment to generate an RIR of the environment. Measured over time in an ongoing fashion, the device may be able to generate a consistent picture of the RIR and the reverberant qualities of the environment, thus better enabling the device to determine or approximate where it is located in relation to walls or corners of the environment (assuming the device is stationary). Further, if the device is moved, the device may be able to determine this change by noticing a change in the RIR pattern. In conjunction with this information, by tracking which lobe of a beampattern the device most often selects as having the strongest spoken signal path over time, the device may begin to notice patterns in which lobes are selected. If a certain set of lobes (or microphones) is selected, the device can heuristically determine the user's typical speaking location in the environment. The device may devote more CPU resources to digital signal processing (DSP) techniques for that lobe or set of lobes. For example, the device may run acoustic echo cancelation (AEC) at full strength across the three most commonly targeted lobes, instead of picking a single lobe to run AEC at full strength. The techniques may thus improve subsequent automatic speech recognition (ASR) and/or speaker recognition results as long as the device is not rotated or moved. And, if the device is moved, the techniques may help the device to determine this change by comparing current RIR results to historical ones to recognize differences that are significant enough to cause the device to begin processing the signal coming from all lobes approximately equally, rather than focusing only on the most commonly targeted lobes.
By focusing processing resources on a portion of an audio signal most likely to include user speech, the SNR of that portion may be increased as compared to the SNR if processing resources were spread out equally to the entire audio signal. This higher SNR for the most pertinent portion of the audio signal may increase the efficacy of the device 102 when performing speaker recognition on the resulting audio signal.
Using the beamforming and directional based techniques above, the system may determine a direction of detected audio relative to the audio capture components. Such direction information may be used to link speech / a recognized speaker identity to video data as described below.
The number of portions/sections generated using beamforming does not depend on the number of microphones in the microphone array. For example, the device 102 may include twelve microphones in the microphone array but may determine three portions, six portions or twelve portions of the audio data without departing from the disclosure. As discussed above, the adaptive beamformer 104 may generate fixed beamforms (e.g., outputs of the FBF 105) or may generate adaptive beamforms using a Linearly Constrained Minimum Variance (LCMV) beamformer, a Minimum Variance Distortionless Response (MVDR) beamformer or other beamforming techniques. For example, the adaptive beamformer 104 may receive the audio input, may determine six beamforming directions and output six fixed beamform outputs and six adaptive beamform outputs corresponding to the six beamforming directions. In some examples, the adaptive beamformer 104 may generate six fixed beamform outputs, six LCMV beamform outputs and six MVDR beamform outputs, although the disclosure is not limited thereto.
The device 102 may determine a number of wireless loudspeakers and/or directions associated with the wireless loudspeakers using the fixed beamform outputs. For example, the device 102 may localize energy in the frequency domain and clearly identify much higher energy in two directions associated with two wireless loudspeakers (e.g., a first direction associated with a first speaker and a second direction associated with a second speaker). In some examples, the device 102 may determine an existence and/or location associated with the wireless loudspeakers using a frequency range (e.g., 1 kHz to 3 kHz), although the disclosure is not limited thereto. In some examples, the device 102 may determine an existence and location of the wireless speaker(s) using the fixed beamform outputs, may select a portion of the fixed beamform outputs as the target signal(s) and may select a portion of adaptive beamform outputs corresponding to the wireless speaker(s) as the reference signal(s).
To perform echo cancellation, the device 102 may determine a target signal and a reference signal and may remove the reference signal from the target signal to generate an output signal. For example, the loudspeaker may output audible sound associated with a first direction and a person may generate speech associated with a second direction. To remove the audible sound output from the loudspeaker, the device 102 may select a first portion of audio data corresponding to the first direction as the reference signal and may select a second portion of the audio data corresponding to the second direction as the target signal. However, the disclosure is not limited to a single portion being associated with the reference signal and/or target signal and the device 102 may select multiple portions of the audio data corresponding to multiple directions as the reference signal/target signal without departing from the disclosure. For example, the device 102 may select a first portion and a second portion as the reference signal and may select a third portion and a fourth portion as the target signal.
Additionally or alternatively, the device 102 may determine more than one reference signal and/or target signal. For example, the device 102 may identify a first wireless speaker and a second wireless speaker and may determine a first reference signal associated with the first wireless speaker and determine a second reference signal associated with the second wireless speaker. The device 102 may generate a first output by removing the first reference signal from the target signal and may generate a second output by removing the second reference signal from the target signal. Similarly, the device 102 may select a first portion of the audio data as a first target signal and may select a second portion of the audio data as a second target signal. The device 102 may therefore generate a first output by removing the reference signal from the first target signal and may generate a second output by removing the reference signal from the second target signal.
The device 102 may determine reference signals, target signals and/or output signals using any combination of portions of the audio data without departing from the disclosure. For example, the device 102 may select first and second portions of the audio data as a first reference signal, may select a third portion of the audio data as a second reference signal and may select remaining portions of the audio data as a target signal. In some examples, the device 102 may include the first portion in a first reference signal and a second reference signal or may include the second portion in a first target signal and a second target signal. If the device 102 selects multiple target signals and/or reference signals, the device 102 may remove each reference signal from each of the target signals individually (e.g., remove reference signal 1 from target signal 1, remove reference signal 1 from target signal 2, remove reference signal 2 from target signal 1, etc.), may collectively remove the reference signals from each individual target signal (e.g., remove reference signals 1-2 from target signal 1, remove reference signals 1-2 from target signal 2, etc.), remove individual reference signals from the target signals collectively (e.g., remove reference signal 1 from target signals 1-2, remove reference signal 2 from target signals 1-2, etc.) or any combination thereof without departing from the disclosure.
The device 102 may select fixed beamform outputs or adaptive beamform outputs as the target signal(s) and/or the reference signal(s) without departing from the disclosure. In a first example, the device 102 may select a first fixed beamform output (e.g., first portion of the audio data determined using fixed beamforming techniques) as a reference signal and a second fixed beamform output as a target signal. In a second example, the device 102 may select a first adaptive beamfrom output (e.g., first portion of the audio data determined using adaptive beamforming techniques) as a reference signal and a second adaptive beamform output as a target signal. In a third example, the device 102 may select the first fixed beamform output as the reference signal and the second adaptive beamform output as the target signal. In a fourth example, the device 102 may select the first adaptive beamform output as the reference signal and the second fixed beamform output as the target signal. However, the disclosure is not limited thereto and further combinations thereof may be selected without departing from the disclosure.
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After determining that there is a single wireless speaker 502 in the configuration 510, the device 102 may set the first section S1 as a reference signal 522 and may identify one or more other sections (e.g., sections S2-S8) as target signals 520a-520g. By removing the reference signal 522 from the target signals 520a-520g, the device 102 may remove an echo caused by receiving audible sound from the wireless speaker 502. Therefore, when the device 102 detects a single wireless speaker 502, the device 102 may associate the wireless speaker 502 (or the section receiving audio from the wireless speaker) with the reference signal and remove the reference signal from the other sections.
While the configuration 510 includes a single wireless speaker 502, the disclosure is not limited thereto and there may be multiple wireless speakers.
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In some examples, the device 102 may not detect a clearly defined speaker signal or determine a speech position. In order to remove an echo, the device 102 may determine pairwise combinations of opposing sections.
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If the device 102 does not detect a strong speaker signal, the device 102 may determine (918) if there is a speech position in the audio data or associated with the audio data. For example, the device 102 may identify a person speaking and/or a position associated with the person using audio data (e.g., audio beamforming), associated video data (e.g., facial recognition) and/or other inputs known to one of skill in the art. In some examples, the device 102 may determine that speech is associated with a section and may determine a speech position using the section. In other examples, the device 102 may receive video data associated with the audio data and may use facial recognition or other techniques to determine a position associated with a face recognized in the video data. If the device 102 detects a speech position, the device 102 may determine (920) the speech position to be a target signal and may determine (922) an opposite direction to be reference signal(s). For example, a first section S1 may be associated with the target signal and the device 102 may determine that a fifth section S5 is opposite the first section S1 and may use the fifth section S5 as the reference signal. The device 102 may determine more than one section to be reference signals without departing from the disclosure. The device 102 may then remove (140) an echo from the target signal using the reference signal(s) and may output (142) speech, as discussed above with regard to
If the device 102 does not detect a speech position, the device 102 may determine (924) a number of combinations based on the audio beamforming. For example, the device 102 may determine a number of combinations of opposing sections and/or microphones, as illustrated in
In some examples, the speech position may be in proximity to a wireless speaker (e.g., a distance between the speech position and the wireless speaker is below a threshold). Therefore, the device 102 may group speech generated by a person with audio output by the wireless speaker, removing both the echo (e.g., audio output by the wireless speaker) and the speech from the audio data. If the device 102 detects more than one wireless speaker, the device 102 may perform a fourth technique to remove the echo while retaining the speech.
As illustrated in
In some examples, the device 102 may use techniques known to one of skill in the art to match first audio output by the first wireless speaker 1004a to second audio output by the second wireless speaker 1004b. For example, the device 102 may determine a propagation delay between the first audio output and the second audio output and may remove the reference signal 1022 from the target signal 1020 based on the propagation delay.
The system 100 may include one or more audio capture device(s), such as a microphone or an array of microphones 118. The audio capture device(s) may be integrated into the device 102 or may be separate.
The system 100 may also include an audio output device for producing sound, such as speaker(s) 116. The audio output device may be integrated into the device 102 or may be separate.
The device 102 may include an address/data bus 1224 for conveying data among components of the device 102. Each component within the device 102 may also be directly connected to other components in addition to (or instead of) being connected to other components across the bus 1224.
The device 102 may include one or more controllers/processors 1204, that may each include a central processing unit (CPU) for processing data and computer-readable instructions, and a memory 1206 for storing data and instructions. The memory 1206 may include volatile random access memory (RAM), non-volatile read only memory (ROM), non-volatile magnetoresistive (MRAM) and/or other types of memory. The device 102 may also include a data storage component 1208, for storing data and controller/processor-executable instructions (e.g., instructions to perform the algorithms illustrated in
Computer instructions for operating the device 102 and its various components may be executed by the controller(s)/processor(s) 1204, using the memory 1206 as temporary “working” storage at runtime. The computer instructions may be stored in a non-transitory manner in non-volatile memory 1206, storage 1208, or an external device. Alternatively, some or all of the executable instructions may be embedded in hardware or firmware in addition to or instead of software.
The device 102 includes input/output device interfaces 1202. A variety of components may be connected through the input/output device interfaces 1202, such as the speaker(s) 116, the microphones 118, and a media source such as a digital media player (not illustrated). The input/output interfaces 1202 may include A/D converters for converting the output of microphone 118 into signals y 120, if the microphones 118 are integrated with or hardwired directly to device 102. If the microphones 118 are independent, the A/D converters will be included with the microphones, and may be clocked independent of the clocking of the device 102. Likewise, the input/output interfaces 1202 may include D/A converters for converting the reference signals x 112 into an analog current to drive the speakers 114, if the speakers 114 are integrated with or hardwired to the device 102. However, if the speakers are independent, the D/A converters will be included with the speakers, and may be clocked independent of the clocking of the device 102 (e.g., conventional Bluetooth speakers).
The input/output device interfaces 1202 may also include an interface for an external peripheral device connection such as universal serial bus (USB), FireWire, Thunderbolt or other connection protocol. The input/output device interfaces 1202 may also include a connection to one or more networks 1299 via an Ethernet port, a wireless local area network (WLAN) (such as WiFi) radio, Bluetooth, and/or wireless network radio, such as a radio capable of communication with a wireless communication network such as a Long Term Evolution (LTE) network, WiMAX network, 3G network, etc. Through the network 1299, the system 100 may be distributed across a networked environment.
The device 102 further includes an adaptive beamformer 104, which includes a fixed beamformer (FBF) 105, a multiple input canceler (MC) 106 and a blocking matrix (BM) 107, and an acoustic echo cancellation (AEC) 108.
Multiple devices 102 may be employed in a single system 100. In such a multi-device system, each of the devices 102 may include different components for performing different aspects of the AEC process. The multiple devices may include overlapping components. The components of device 102 as illustrated in
The concepts disclosed herein may be applied within a number of different devices and computer systems, including, for example, general-purpose computing systems, multimedia set-top boxes, televisions, stereos, radios, server-client computing systems, telephone computing systems, laptop computers, cellular phones, personal digital assistants (PDAs), tablet computers, wearable computing devices (watches, glasses, etc.), other mobile devices, etc.
The above aspects of the present disclosure are meant to be illustrative. They were chosen to explain the principles and application of the disclosure and are not intended to be exhaustive or to limit the disclosure. Many modifications and variations of the disclosed aspects may be apparent to those of skill in the art. Persons having ordinary skill in the field of digital signal processing and echo cancellation should recognize that components and process steps described herein may be interchangeable with other components or steps, or combinations of components or steps, and still achieve the benefits and advantages of the present disclosure. Moreover, it should be apparent to one skilled in the art, that the disclosure may be practiced without some or all of the specific details and steps disclosed herein.
Aspects of the disclosed system may be implemented as a computer method or as an article of manufacture such as a memory device or non-transitory computer readable storage medium. The computer readable storage medium may be readable by a computer and may comprise instructions for causing a computer or other device to perform processes described in the present disclosure. The computer readable storage medium may be implemented by a volatile computer memory, non-volatile computer memory, hard drive, solid-state memory, flash drive, removable disk and/or other media. Some or all of the STFT AEC module 1230 may be implemented by a digital signal processor (DSP).
As used in this disclosure, the term “a” or “one” may include one or more items unless specifically stated otherwise. Further, the phrase “based on” is intended to mean “based at least in part on” unless specifically stated otherwise.
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