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JP2018518893

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DESCRIPTION JP2018518893
Abstract One or more embodiments describe an audio processing system for a personal listening
device that includes a set of microphones, a noise reduction module, an audio ducker, and a
mixer. The set of microphones is configured to receive a first set of audio signals from the
environment. The noise reduction module is configured to detect when the signal of interest is
present in the first plurality of audio signals and to transmit a ducking control signal after
detecting the signal of interest. The audio ducker is configured to receive the ducking control
signal and to receive the second plurality of audio signals via the playback device. The voice
ducker is further configured to reduce the amplitude of the second plurality of voice signals
relative to the signal of interest based on the ducking control signal. The mixer combines the first
plurality of audio signals and the second plurality of audio signals. [Selected figure] Figure 1
Sport headphones with situational awareness
[0001]
Embodiments of the present disclosure relate generally to audio signal processing, and more
particularly to sports headphones with context awareness.
[0002]
Headphones, earphones, earbuds and other personal listening devices are usually used by
individuals who want to hear sound sources such as music, speech or movie soundtracks without
disturbing other people in the vicinity Be done.
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1
Such devices typically cover the entire ear or completely seal the ear canal in order to provide
good sound quality. Typically, these devices include an audio plug for plugging into the audio
output of the audio playback device. The audio plug is connected to a cable that carries the audio
signal from the audio playback device to a pair of headphones or earphones that covers or is
inserted into the listener's ear. As a result, headphones or earphones provide a good acoustic
seal, thereby reducing the leakage of the audio signal and improving the listener's perceived
quality, in particular with regard to the bass response.
[0003]
One problem with the above devices is that the ability of the device to hear the ambient sound of
the listener is substantially reduced in order to form a good acoustic seal with the ear. As a result,
the listener may not be able to hear certain important sounds from the environment, such as
approaching vehicles, interphone system announcements, or alarms. In one example, a cyclist
traveling in the pace line is listening to music, but may still want to hear the voices of other
cyclists in the pace line that run forward and backward. In another example, a dining customer
may be listening to music while waiting for an announcement that the customer's table is ready.
[0004]
One solution to the above problem is to acoustically or electronically mix the audio from the
environment with the audio signal received from the playback device. The listener can then hear
both the audio from the playback device and the audio from the environment. However, one
drawback of such a solution is that the listener typically hears not only the specific
environmental sounds that the listener wants to hear, but all the sound from the environment. As
a result, the perceived quality of the listener may be substantially reduced.
[0005]
As illustrated above, more effective techniques for providing both playback and ambient sound to
a personal listening device would be useful.
[0006]
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One or more embodiments describe an audio processing system for a personal listening device
that includes a set of microphones, a noise reduction module, an audio ducker, and a mixer.
The set of microphones is integrated into the personal listening device and configured to receive
the first set of audio signals from the environment. The noise reduction module is coupled to the
first plurality of microphones and configured to detect when the signal of interest is present in
the first plurality of audio signals and to transmit a ducking control signal after detecting the
signal of interest. Ru. The audio ducker is coupled to the noise reduction module and configured
to receive the ducking control signal and to receive a second plurality of audio signals through
the playback device. The voice ducker is further configured to reduce the amplitude of the
second plurality of voice signals relative to the signal of interest based on the ducking control
signal. The mixer is coupled to the audio ducker and is configured to combine the first plurality
of audio signals and the second plurality of audio signals.
[0007]
Other embodiments are limited to computer readable media comprising instructions for
performing one or more aspects of the disclosed technology, and methods for performing one or
more aspects of the disclosed technology. Not included.
[0008]
At least one advantage of the disclosed approach is that a listener using the disclosed personal
listening device listens to the sound of the audio of interest from the environment in addition to
the high quality audio signal from the playback device At the same time, other sounds from the
environment are suppressed as compared to the sound of interest.
As a result, the potential for the listener to hear only the desired audio signal is improved, leading
to a better quality audio experience for the listener.
[0009]
A more detailed description of one or more embodiments briefly summarized above will be to a
particular embodiment so that the recited features of the one or more embodiments described
above may be understood in detail. Reference may be made by reference, some of which are
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3
shown in the attached drawings. However, the attached drawings show only typical
embodiments, and therefore should not be considered as limiting the scope in any way, and the
scope of the disclosure encompasses other embodiments as well. Please note.
[0010]
7 illustrates a speech processing system configured to implement one or more aspects of various
embodiments. FIG. 2 conceptually illustrates one application of the speech processing system of
FIG. 1 according to various embodiments. Figure 5 conceptually illustrates another application of
the speech processing system of Figure 1 according to various other embodiments. FIG. 7 is a
flow diagram of method steps for processing audio signals of playback and environment
according to various embodiments. FIG. 7 is a flow diagram of method steps for processing audio
signals of playback and environment according to various embodiments.
[0011]
In the following description, numerous specific details are set forth in order to provide a more
thorough understanding of certain embodiments. However, it will be apparent to one skilled in
the art that other embodiments may be practiced without one or more of these specific details, or
with additional specific details.
[0012]
System Overview FIG. 1 shows an audio processing system 100 configured to implement one or
more aspects of various embodiments. As shown, the audio processing system 100 includes
microphone (microphone) arrays 105 (0) and 105 (1), beam formers 110 (0) and 110 (1), noise
reduction 115, equalizer 120, gate 125, limiter 130. Mixers 135 (0) and 135 (1), amplifiers 140
(0) and 140 (1), speakers 145 (0) and 145 (1), subharmonic processing 155, automatic gain
control (AGC) 160, And including, but not limited to, ducker 165.
[0013]
In various embodiments, voice processing system 100 may process state machines, central
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processing units (CPUs), digital signal processors (DSPs), microcontrollers, application specific
integrated circuits (ASICs), or data and execute software applications. May be implemented as
any device or structure configured to In some embodiments, one or more of the blocks shown in
FIG. 1 may be implemented with separate analog or digital circuits. In one example, without
limitation, left amplifier 140 (0) and right amplifier 140 (1) may be implemented with
operational amplifiers.
[0014]
Microphone arrays 105 (0) and 105 (1) receive audio from the physical environment. The
microphone array 105 (0) receives audio from the physical environment around the listener's left
ear. Correspondingly, the microphone array 105 (1) receives audio from the physical
environment around the listener's right ear. Each of the microphone arrays 105 (0) and 105 (1)
includes a plurality of microphones. The microphone arrays 105 (0) and 105 (1) are shown as
each including two microphones, but may each include more than two microphones within the
scope of the present disclosure. Because microphone arrays 105 (0) and 105 (1) include multiple
microphones, beamformers 110 (0) and 110 (1) may be configured in a directional manner as
described further herein. It is possible to filter speech spatially. Microphone arrays 105 (0) and
105 (1) transmit the received speech to beamformers 110 (0) and 110 (1), respectively.
[0015]
Beamformers 110 (0) and 110 (1) receive audio signals from microphone arrays 105 (0) and
105 (1), respectively. Beamformers 110 (0) and 110 (1) process the received speech signal
according to one of several modes. Modes include, without limitation, omnidirectional modes,
dipole modes, and cardioid modes. In various embodiments, the mode may be preprogrammed by
the manufacturer or may be a user selectable setting. Beamformers 110 (0) and 110 (1) measure
the strength of the speech received from each microphone in the corresponding microphone
arrays 105 (0) and 105 (1) to determine the direction of the incoming speech. In some
embodiments, signals received from one of the microphones in microphone arrays 105 (0) and
105 (1) are digitally delayed, and then microphone arrays 105 (0) and 105 (1) are ) Is subtracted
from the signal from another one of the microphones in.
[0016]
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Depending on the mode selected, beamformers 110 (0) and 110 (1) amplify signals originating
from one direction and attenuate signals originating from the other. For example, and not by way
of limitation, if the selected mode is an omnidirectional mode, beamformers 110 (0) and 110 (1)
will uniformly amplify signals originating from all directions. If the selected mode is a dipole
mode, also referred to herein as a "figure-of-eight" mode, beamformers 110 (0) and 110 (1) may
be voices originating from two directions, typically from the front to back. The signal may be
amplified and the audio signal originating from the other direction, typically from left to right,
may be attenuated. If the selected mode is a cardioid mode, the beamformers 110 (0) and 110
(1) amplify audio signals originating from most directions, such as from the side and from above,
down the listener's Etc. can attenuate the audio signal originating from a particular direction.
Alternatively, if the selected mode is a cardioid mode, beamformers 110 (0) and 110 (1) amplify
the audio signal originating from the listener's front, and originate from the listener's back Audio
signal to be attenuated. Beamformers 110 (0) and 110 (1) amplify and attenuate the signals
received from microphone arrays 105 (0) and 105 (1), respectively, according to the selected
mode, and 110 (1) sends the resulting audio signal to the noise reduction 115.
[0017]
The noise reduction 115 is a module that receives audio signals from the beamformers 110 (0)
and 110 (1). The noise reduction 115 analyzes the received voice signal, suppresses a signal
determined not to pay much attention, such as a steady state or a noise signal, and passes a
signal determined to be a focus signal, such as a transient signal. In some embodiments, noise
reduction 115 may analyze the received signal in the frequency domain over a period of time. In
such embodiments, noise reduction 115 may transform the received signal into the frequency
domain and divide the frequency domain into appropriate bins, each bin corresponding to a
particular frequency range. The noise reduction 115 may measure the amplitude across samples
over time to determine which frequency bins correspond to steady state signals and which
frequency bins correspond to transient signals. In general, steady state signals may correspond
to background noise including but not limited to traffic noise, hum, hiss, rain, and wind. If a
particular frequency bin is associated with an amplitude that remains relatively constant over
time, noise reduction 115 may determine that the frequency bin is associated with a steady state
signal. Noise reduction 115 may attenuate such steady state signals.
[0018]
On the other hand, the transient signal may correspond to the signal of interest, including but not
limited to human speech, car horns and sirens. If a particular frequency bin is associated with an
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amplitude that varies significantly over time, noise reduction 115 may determine that the
frequency bin is associated with a transient signal. The noise reduction 115 may pass such
transients to the equalizer 120 and optionally amplify the transients.
[0019]
In one example, without limitation, noise reduction 115 may analyze 256 frequency domain
samples, in which case the frequency domain samples will be evenly distributed over a 1 second
period. The noise reduction 115 will analyze 256 samples for each frequency bin to determine
which frequency bins are associated with the steady state signal and which bins are associated
with the transient signal. Noise reduction may analyze another 256 frequency domain samples.
Each set of 256 frequency domain samples may have a designated overlap with the preceding set
of 256 frequency domain samples and the subsequent set of 256 frequency domain samples. If
the overlap is specified to be 50%, then each set of 256 frequency domain samples will include
the last 128 samples of the immediately preceding set of samples and the first 128 samples of
the immediately following set of samples. In some embodiments, noise reduction 115 may
perform operations in the time domain without first converting to the frequency domain. In such
embodiments, the noise reduction 115 may include multiple parallel band pass filters (not
explicitly shown) corresponding to the frequency bins described herein.
[0020]
Furthermore, the noise reduction 115 produces a control signal that is identified when the noise
reduction 115 detects a signal of interest in the listener's environment. In general, the signal of
interest includes any sound from the environment that is not a low level steady state sound,
including but not limited to human speech, car horns, approaching car sounds, and alarms. This
kind of important sound emanating from the environment is characterized as an intermittent
audio signal, which typically has a high audio level compared to the average background audio
level, and acts as an interrupt. Stated differently, the signal of interest includes any intermittent
audio sounds having high audio levels as compared to the average audio signal levels received by
the microphone arrays 150 (0) and 150 (1). If the noise reduction 115 detects such a signal, the
noise reduction 115 sends a corresponding signal to the ducker 165, as further described herein.
In various embodiments, noise reduction 115 may reduce noise in the received signal by spectral
subtraction as well as other techniques including, but not limited to, speech detection,
recognition, and extraction.
03-05-2019
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[0021]
In some embodiments, the noise reduction 115 may also include active noise cancellation (ANC)
functionality (not explicitly shown). In such embodiments, the noise reduction 115 may perform
the ANC function on frequency bins associated with frequencies below a threshold frequency,
such as 200 Hz. The noise reduction 115 may perform the noise reduction function as described
herein for frequency bins associated with frequencies above a threshold frequency, such as 200
Hz.
[0022]
After performing noise reduction and optionally ANC, noise reduction 115 sends the resulting
audio signal to equalizer 120.
[0023]
The equalizer 120 receives an audio signal from the noise reduction 115.
The equalizer 120 performs frequency-based amplitude adjustment on the received audio signal
to improve the audio quality of the audio signal received from the listener's environment.
Environmental studio signals that reach the listener's ear via the microphone arrays 110 (0) and
110 (1) of the audio processing system 100 typically reach the listener's ear when the audio
processing system 100 is not used The listener sounds different compared to the sound of the
same voice. Such acoustic differences arise from acoustical changes that occur due to covering
the ear with headphones or plugging the earpiece into the ear canal. Equalizer 120 compensates
for such differences by selectively increasing, decreasing, or maintaining the volume levels of
various frequency bands within the audible range.
[0024]
In some embodiments, even though such amplification may make the audio signal less natural,
the equalizer 120 may also use a frequency band to make such audio signal more noticeable to
the user. May be amplified. In this way, the equalizer 120 may amplify certain audio signals, such
as speech or alarms, so that the listener can easily hear these certain audio signals. For example,
without limitation, equalizer 120 may amplify a signal generated in a frequency band
03-05-2019
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corresponding to human speech. As a result, the listener can easily hear human speech through
the environment, even if the resulting audio signal does not sound natural to the listener. In some
embodiments, the equalizer 120 may filter out signals in certain frequency ranges that the
listener does not focus on. In one example, without limitation, equalizer 120 may filter out
signals having frequencies lower than 120 Hz, and such signals may be associated with
background noise. The equalizer 120 transmits the equalized audio signal to the gate 125.
[0025]
The gate 125 receives the audio signal from the equalizer 120 and suppresses the audio signal
that falls below the threshold volume, ie amplitude, level. An audio signal that exceeds a
threshold volume, ie, amplitude, level passes through gate 125 to limiter 130. As a result, the
gate 125 further suppresses low-level audio signals such as hiss and hum. In some embodiments,
the threshold level may be constant over the relevant frequency band. In other embodiments, the
threshold level may vary across the relevant frequency bands. In these latter embodiments, the
gating threshold level may be higher in certain frequency bands and lower in other frequency
bands. In other words, the gating function performed by gate 125 is a function of the audio
signal frequency. Gate 125 transmits the resulting audio signal to limiter 130.
[0026]
The limiter 130 quickly detects such loud sounds before they reach the listener's ear and limits
such loud signals so that they do not exceed the maximum allowable audio level. Thus, the limiter
130 attenuates loud signals to protect the listener. In one example, without limitation, limiter
130 may have a maximum allowable audio level of 95 dB SPL. In such a case, when limiter 130
receives an audio signal above 95 dB SPL, limiter 130 will attenuate the audio signal such that
the resulting audio signal does not exceed 95 dB SPL. In some embodiments, the limiter 130 also
performs a compression function so that audio level limiting occurs gradually as the volume
increases, rather than suddenly suppressing all audio signals above the maximum allowed audio
level. You may Such dynamic range processing provides a more comfortable listening experience,
as large volume fluctuations are generally reduced. The limiter sends the resulting audio signal to
mixers 135 (0) and 135 (1).
[0027]
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Subharmonic processing 155 receives from audio feed 150 an audio signal from a playback
device (not explicitly shown). Subharmonic processing 155 receives these audio signals by any
technically feasible technology, including but not limited to wired connections, Bluetooth or
Bluetooth LE connections, and wireless Ethernet connections. . The subharmonic processing 155
synthesizes and boosts an audio signal which is a subharmonic signal of the received audio
signal. Such subharmonic synthesis mixes or combines the received speech signal with the
synthesized subharmonic signal and has a higher bass level as compared to the non-processed
speech signal, as a result. Generate an audio signal. While certain listeners may prefer
subharmonic processing 155, other listeners may not like such processing. In addition, other
listeners may prefer sub-harmonic processing 155 for certain genres, but may not prefer such
processing for other genres. In some embodiments, the listener may control whether
subharmonic processing 155 is available and may also control the level of subharmonic boost
performed by subharmonic processing 155. Subharmonic processing 155 sends the resulting
speech signal to automatic gain control 160.
[0028]
Automatic gain control 160 receives the speech signal from subharmonic processing 155. The
automatic gain control 160 amplifies the sound level of quieter sounds and reduces the level of
louder sounds to produce a more consistent output volume over time. Automatic gain control
160 is adjusted at a fixed target speech level of the received speech signal. Typically, the fixed
target voice level is a factory setting established by the manufacturer during product
development and manufacturing. In one embodiment, this fixed target sound level is -24 dB.
Automatic gain control 160 then determines that a portion of the received speech signal is
different from this fixed target speech level. Automatic gain control 160 calculates the scale
factor such that when the received speech signal is multiplied by the scale factor, the resulting
speech signal is closer to the fixed target speech level. In one example, without limitation, a song
may be mastered at different volume levels based on various factors such as the time the song
was produced and the genre of the song. When a listener selects songs with varying master
recording levels, the listener may find it difficult to listen to these songs. If the listener adjusts the
volume level to listen to a quiet song, the volume may be undesirably loud when a song with a
large amount of sound is played back. Similarly, when the listener adjusts the volume level to
listen to a loud song, the volume may be too low to hear a quiet song. Automatic gain control
160 processes the received audio signal such that the listening volume of the music is more
consistent over time.
[0029]
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Ducker 165 receives an audio signal from automatic gain control 160. The ducker also receives
control signals from the noise reduction 115. The control signal identifies when the noise
reduction 115 detects a signal of interest in the listener's environment. When such a signal is
detected, the ducker 165 temporarily reduces the volume level of the received audio signal. Thus,
when a signal of interest is received from the environment, the ducker 165 reduces or ducks the
audio from the playback device. As a result, the listener can more easily hear the signal of
interest from the environment. In other words, when the signal of interest is present on the
microphone arrays 105 (0) and 105 (1), the ducker 165 reduces or ducks the level of music so
that the signal of interest can be heard and understood. Ducker 165 transmits the resulting audio
signal to mixers 140 (0) and 140 (1).
[0030]
Mixers 135 (0) and 135 (1) receive the processed ambient audio signal from limiter 130 and
receive processed music or other audio from ducker 165. The mixer 135 (0) mixes or combines
the received audio signals for the left audio channel, and correspondingly the mixer 135 (1)
mixes the received audio signals for the right audio channel. In some embodiments, mixers 135
(0) and 135 (1) may perform simple additive or multiplicative mixing of the received audio
signal. In other embodiments, mixers 135 (0) and 135 (1) may weight each of the incoming
speech signals based on the user's volume settings. In these latter embodiments, due to the large
audio signal received from the ducker 165, such as when the listener increases the listening
volume, the audio signal received from the limiter 130 is likely compared to the audio signal
from the ducker 165 Increase by different amounts. After performing the mixing function, left
mixer 135 (0) and right mixer 135 (1) send the resulting signals to left amplifier 140 (0) and
right amplifier 140 (1). Left amplifier 140 (0) and right amplifier 140 (1) amplify the received
audio signal based on volume control (not explicitly shown), and the resulting audio signal is
output to left speaker 145 (0) and right speaker 145. Send to (1) respectively. The left speaker
145 (0) and the right speaker 145 (1) also receive audio signals from the direct feed 170. Direct
feeds represent acoustic signals received from the listener's environment. The left speaker 145
(0) and the right speaker 145 (1) may be configured as the left amplifier 140 (0) and the right
speaker 140 when the audio processing system 100 is no longer functioning, such as when the
battery power falls below a threshold voltage level. (1) Transmit the signal from the direct feed
170, not the processed audio signal received from each.
[0031]
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In some embodiments, the listener may control certain functions of the audio processing system
100 or set certain parameters by one or more capacitive touch sensors (not explicitly shown).
When a listener touches such a sensor, a change in the capacitance of the capacitive touch
sensor is detected. Such a change in capacitance causes the audio processing system 100 to
perform functions including, but not limited to, changing the beamforming mode and changing
the filter parameters. The listener may control certain functions of the audio processing system
100 or set certain parameters via multiple capacitive touch sensors that detect motion. For
example, without limitation, in the case where three or more capacitive touch sensors are
arranged in a row, the listener touches the lower capacitive touch sensor with a finger, and the
center and upper electrostatic The volume level can be increased by moving the finger vertically
to the capacitive touch sensor. Correspondingly, the listener can lower the volume level by
touching the upper capacitive touch sensor with a finger and moving the finger vertically to the
middle and lower capacitive touch sensors. In other embodiments, the listener controls certain
functions of the voice processing system 100 or certain parameters via an application executing
on a computing device, including but not limited to a smartphone, tablet computer, or laptop
computer. May be set. Such applications may communicate with speech processing system 100
in any technically feasible manner, including but not limited to Bluetooth®, BluetoothLE, and
Wireless Ethernet®.
[0032]
Operation of Speech Processing System FIG. 2 conceptually illustrates one application of the
speech processing system of FIG. 1, in accordance with various embodiments. As shown, the
riders 210 (0), 210 (1), 210 (2), 210 (3), and 210 (4) ride the bicycle in a straight line. The rider
210 (2) is equipped with a personal listening device (not explicitly shown) that exhibits a dipole,
ie, figure-of-eight pattern, as shown by the dipole patterns 220 (0) and 220 (1). The dipole
pattern 220 (0) and the dipole pattern 220 (1) correspond to the right ear and the left ear of the
rider 210 (2), respectively.
[0033]
As shown, the distances of the dipole patterns 220 (0) and dipole patterns 220 (1) from the right
and left ears of the rider 210 (2) are indicative of signal strength as a function of angle. Bicycle
riders often form a pace line, in which case the cyclists are in front of or behind each other in a
straight line. This pace line pattern reduces air resistance (since only the lead rider resists the
resistance) and is also safer when there are cars on the road. Because the rider 210 (2) is
equipped with a personal listening device having dipole patterns 220 (0) and 220 (1), the rider
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210 (2) is required to have forward riders 210 (0) and 210 (1). And the sound signals from the
rear riders 210 (3) and 210 (4) are more easily heard as compared to the sound signals from the
left and right sides of the rider 210 (2).
[0034]
FIG. 3 conceptually illustrates another application of the speech processing system of FIG. 1
according to various other embodiments. As shown, the skier 310 is wearing a personal listening
device (not explicitly shown) that represents a cardioid pattern as shown by the cardioid pattern
320. The cardioid pattern 320 corresponds to the left ear of the skier 310. For the sake of clarity,
the cardioid pattern corresponding to the right ear of the skier 310 is not shown in FIG. As
shown, the distance of the outline of the cardioid pattern 320 from the left ear of the skier 310 is
indicative of signal strength as a function of angle. Sounds from under the skier 310, such as the
sound of skis on snow and ice, are suppressed relative to sounds from other directions, including
sounds originating from the side or top of the skier 310. The application shown in FIG. 3 also
relates to other related activities, including but not limited to snowboarding, running, and
treadmill exercise.
[0035]
4A-4B show flow diagrams of method steps for processing audio signals of playback and
environment according to various embodiments. Although the method steps are described in
conjunction with the systems of FIGS. 1-3, one skilled in the art would appreciate that any system
configured to perform the method steps in any order is within the scope of the present
disclosure. You will understand that.
[0036]
As illustrated, the method 400 starts at step 402, at which the microphone arrays 105 (0) and
105 (1) associated with the audio processing system 100 receive audio signals from the listener's
environment . In step 404, microphone arrays 110 (0) and 110 (1) are used according to a
particular beamforming mode, wherein beamformers 110 (0) and 110 (1) include, but are not
limited to, omnidirectional, dipole, and cardioid patterns. Attenuate and amplify the audio signal
from In step 406, noise reduction 115 reduces the sound levels of steady state signals such as
hum, hiss and wind while amplifying the sound levels of transients such as human speech, car
03-05-2019
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horns and alarms. Let At step 408, the noise reduction 115 also performs active noise
cancellation on a portion of the received audio signal. At step 410, the equalizer compensates for
frequency imbalance, such as an imbalance associated with wearing headphones or earphones as
compared to not wearing any personal listening devices.
[0037]
At step 412, the gate 125 suppresses the audio signal below the threshold volume, ie the
amplitude level. In some embodiments, the threshold volume of gate 125 may be constant over
the relevant frequency range. In other embodiments, the threshold volume may vary as a
function of frequency. At step 414, the limiter 130 attenuates the audio signal above the
specified maximum allowable audio level. At step 416, subharmonic processing 155 synthesizes
the low frequency audio signal based on the audio signal feed received from the playback device.
At step 418, automatic gain control 160 adjusts the volume of the audio signal feed received
from the playback device. For example, without limitation, automatic gain control 160 may
increase the volume of a quiet song and may decrease the volume of a loud song. At step 420,
the ducker 165 temporarily reduces the volume of the audio signal feed received from the
playback device based on the control signal from the noise reduction 115 indicating that the
source of interest is received from the listener's environment Let
[0038]
At step 422, left mixer 135 (0) and right mixer 135 (1) mix the audio received from limiter 130
with the audio received from ducker 165 for the left and right channels, respectively. At step
424, left amplifier 140 (0) and right amplifier 140 (1) amplify the audio signal received from left
mixer 135 (0) and right mixer 135 (1), respectively. At step 426, left amplifier 140 (0) and right
amplifier 140 (1) transmit the final audio signal to left speaker 145 (0) and right speaker 145
(1), respectively. Method 400 then ends. In some embodiments, method 400 does not end, but
the components of speech processing system 100 continue to perform the steps of method 400
in a continuous loop. In these embodiments, after step 426 is performed, method 400 proceeds
to step 402 described above. The steps of method 400 continue to be performed in a continuous
loop until certain events occur, such as powering down a device that includes voice processing
system 100.
[0039]
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In summary, the disclosed technology allows a listener using a personal listening device to hear
music or a desired sound mixed with some sound of interest from the listener's environment.
Steady-state signals from the environment, such as hiss, hum and traffic noise, are removed from
the audio environment to enhance the music and the environmental sound of interest. Audio
from the listener's environment is received via the microphone array and processed by
beamformer, noise reduction, equalization, gating and limiting. Music and other audio signals
received from the playback device are processed by subharmonic processing, automatic gain
control, and ducking. The mixer mixes the ambient and playback audio and sends the resulting
signal to an amplifier, which in turn similarly transmits the audio signal to a pair of headphones,
earphones, earbuds, or other speakers in a personal listening device. Send to
[0040]
At least one advantage of the techniques described herein is that the listener using the disclosed
personal listening device adds certain quality audio of interest from the environment in addition
to high quality audio signals from the playback device. Listening to the sound is that at the same
time other sounds from the environment are suppressed compared to the sound of interest. As a
result, the potential for the listener to hear only the desired audio signal is improved, leading to a
better quality audio experience for the listener.
[0041]
The description of the various embodiments has been presented for the purpose of illustration,
but is not intended to be exhaustive or to limit the disclosed embodiments. Many modifications
and variations will be apparent to those of ordinary skill in the art without departing from the
scope and spirit of the disclosed embodiments.
[0042]
Aspects of the embodiments may be embodied as a system, method or computer program
product. Thus, aspects of the present disclosure may be complete hardware embodiments,
complete software embodiments (including firmware, resident software, microcode, etc.), or all
generally referred to herein as "circuits," "modules," Or it may take the form of an embodiment
combining software and hardware aspects, which may be referred to as a "system". Additionally,
03-05-2019
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aspects of the present disclosure may take the form of a computer program product embodied on
one or more computer readable medium (s) having computer readable program code embodied
thereon.
[0043]
Any combination of one or more computer readable medium (s) may be utilized. Computer
readable media may be computer readable signal media or computer readable storage media. A
computer readable storage medium may be, for example but not limited to, an electronic,
magnetic, optical, electromagnetic, infrared or semiconductor system, apparatus or device, or any
suitable combination of the foregoing. More specific examples (non-exhaustive list) of computer
readable storage media are: electrical connection with one or more wires, portable computer
diskette, hard disk, random access memory (RAM), read only memory (ROM) And erasable
programmable read only memory (EPROM or flash memory), optical fiber, portable compact disc
read only memory (CD-ROM), optical storage device, magnetic storage device, or any suitable
combination of the foregoing. It shall be In the context of this document, any computer readable
storage medium may contain or store programs for use by, or in conjunction with, an instruction
execution system, apparatus, or device. It may be a tangible medium of
[0044]
Aspects of the present disclosure are described above with reference to flowchart illustrations
and / or block diagrams of methods, apparatus (systems), and computer program products
according to embodiments of the present disclosure. It is to be understood that each block of the
flowchart illustrations and / or block diagrams, and combinations of blocks in the flowchart
illustrations and / or block diagrams, can be implemented by computer program instructions.
These computer program instructions may be provided to a processor of a general purpose
computer, a special purpose computer, or another programmable data processing device for
manufacturing a machine, thereby being executed by the processor of the computer or other
programmable data processing device Instructions enable the implementation of the functions /
operations specified in the blocks of the flowcharts and / or block diagrams. Such processors
may be, without limitation, general purpose processors, special purpose processors, application
specific processors, or field programmable.
[0045]
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16
The flowcharts and block diagrams in the Figures illustrate the architecture, functionality, and
operation of possible implementations of systems, methods, and computer program products
according to various embodiments of the present disclosure. In this regard, each block in the
flowchart or block diagram may also represent a module, segment, or portion of code that
includes one or more executable instructions that perform the specified logical function (s). Good.
It should also be noted that in some alternative implementations, the functions noted in the block
may occur out of the order noted in the figures. For example, two blocks shown in succession
may in fact be executed substantially simultaneously, or the blocks may be executed in the
reverse order depending on the functionality required. It may be. Each block of the block
diagrams and / or flowchart illustrations, and combinations of blocks in the block diagrams and /
or flowchart illustrations, are by dedicated hardware based systems that perform specified
functions or operations, or by a combination of dedicated hardware and computer instructions.
Note also that it can be implemented.
[0046]
While the above is directed to embodiments of the present disclosure, other and further
embodiments of the disclosure may be devised without departing from its basic scope and its
scope as determined by the following claims. It may be done.
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17
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