close

Вход

Забыли?

вход по аккаунту

?

DESCRIPTION JP2008250270

код для вставкиСкачать
Patent Translate
Powered by EPO and Google
Notice
This translation is machine-generated. It cannot be guaranteed that it is intelligible, accurate,
complete, reliable or fit for specific purposes. Critical decisions, such as commercially relevant or
financial decisions, should not be based on machine-translation output.
DESCRIPTION JP2008250270
An object of the present invention is to commercialize a noise canceling system with a digital
configuration. SOLUTION: A second signal processing system which picks up an external sound
by a microphone and interpolates a signal of 64 fs to 8 fs to generate a second audio signal for
noise cancellation, and further interleaves up to 1 fs The first signal processing system generates
a second noise cancellation audio signal after interpolation, interpolates to 8 fs, and the driver
synthesizes the noise cancellation audio signal from the driver. Output. Then, the second signal
processing system cancels at least the high frequency band, and the first signal processing
system cancels the lower frequency band. [Selected figure] Figure 6
Signal processing apparatus, signal processing method
[0001]
The present invention relates to a signal processing apparatus and method for performing signal
processing according to a predetermined purpose for an audio signal.
[0002]
There is known a so-called noise canceling system compatible with a headphone device, which is
designed to actively cancel external noises that are heard when playing back audio of content
such as music by using the headphone device. It has become
And as such a noise canceling system, two systems of a feedback system and a feedforward
10-04-2019
1
system are known roughly.
[0003]
For example, Patent Document 1 generates an audio signal obtained by inverting the phase of
noise inside a sound tube collected by a microphone unit provided in the vicinity of an earphone
unit in a sound tube attached to the user's ear, The configuration in which external noise is
reduced by outputting this as sound from the earphone unit 3, that is, the configuration of the
noise canceling system corresponding to the feedback method is described. Further, Patent
Document 2 has, as a basic configuration thereof, a configuration in which characteristics based
on a required transfer function are given to an audio signal obtained by collecting sound by a
microphone attached to a headphone device outer casing, and output from the headphone
device. That is, the configuration of the noise canceling system corresponding to the feed forward
method is described.
[0004]
JP-A-3-214892 JP-A-3-96199
[0005]
By the way, although it can be said about any of the above-mentioned feedback system and
feedforward system, what is put into practical use as noise canceling system of the headphone
apparatus in consumer equipment at present is what was comprised by the analog circuit. It has
become.
In order to effectively obtain the noise cancellation effect of the noise canceling system, for
example, the difference between the external unnecessary sound picked up by the microphone
and the sound outputted from the driver for the cancellation of the unnecessary sound It is
necessary to keep the difference within a certain level. In other words, in the noise canceling
system, it is required that the speed (response speed) from when the external unnecessary sound
is input until the cancellation sound corresponding to this is output is within a certain value.
However, when the noise canceling system is to be configured by digital circuits, the input and
the output are provided with an A / D converter and a D / A converter. In the processing time of
A / D converter and D / A converter widely used at present, the delay is considerably large when
considering adoption as a noise canceling system, and it is difficult to obtain an effective noise
10-04-2019
2
cancellation effect. For example, in fields such as military and industrial, there are A / D
converters and D / A converters with high sampling frequencies and low delays, but these are
extremely expensive and are not suitable for use in consumer products. It is not realistic. This is
the reason why the noise canceling system is currently configured with an analog circuit instead
of a digital circuit.
[0006]
However, by replacing the analog circuit with a digital circuit, it is easier to change or switch the
characteristics or operation mode without changing the physical component element constants
or changing them. Also, if the system is related to sound such as a noise canceling system, there
are many advantages such as further improvement of the sound quality. Therefore, an object of
the present invention is, for example, to form a noise canceling system of a headphone device for
consumer use with a digital circuit, but to obtain a practically sufficient noise canceling effect.
[0007]
Therefore, in consideration of the problems described above, the present invention is configured
as a signal processing device as follows. That is, a digital signal of the first format with a
predetermined number of quantization bits of 1 bit or more obtained by ΔΣ modulation
processing is input, and it is represented by n × fs (n is a natural number), where fs is a
predetermined reference sampling frequency. A first decimation processing means for generating
and outputting a second type digital signal to be converted into a pulse code modulation signal
according to a sampling frequency, and a second type digital signal output from the first
decimation processing means To generate and output a third type digital signal having a format
as a pulse code modulation signal according to a sampling frequency represented by m × fs (m
is a natural number and m <n) Decimation processing means and the third type digital signal
output from the second decimation processing means to input a predetermined cancellation
target sound. A first function corresponding signal processing means for giving a predetermined
cancellation signal characteristic for cell and outputting in the same third format, and a third
function output from the first function corresponding signal processing means An interpolation
processing means adapted to convert a signal of one format into a second format and outputting
the signal, and a digital signal of the second format output from the first decimation means are
input to a predetermined format. A second function corresponding signal processing means for
giving a predetermined cancellation signal characteristic for canceling a cancellation target
sound and outputting in the same second format, and at least from the second function
corresponding signal processing means At least the digital signal of the second type output and
10-04-2019
3
the digital signal of the second type output from the interpolation processing means are
combined, It comprises the combining means for outputting to the input stage for log conversion
processing.
[0008]
In the above configuration, with respect to the digital signal processing system of the system for
canceling (reducing or attenuating) the predetermined cancellation target sound, signal
characteristics for canceling the predetermined cancellation target sound to a signal of sampling
frequency of m × fs (cancel signal characteristic And a system for giving cancellation signal
characteristics to a signal of sampling frequency n × fs higher than this. Then, in the latter
system in which the cancellation signal characteristic is given to the signal of sampling frequency
n × fs, the synthesizer is not subjected to the decimation by the second decimation processing
means and the interpolation processing by the interpolation processing means. , The former
cancellation signal characteristic is synthesized with the signal of sampling frequency n × fs
given. In this way, the delay of the signal of the latter system is greatly reduced by omitting
decimation processing and interpolation processing.
[0009]
As described above, by providing a system for providing a plurality of cancellation signal
characteristics, the signal delay (signal propagation time) in at least one of the systems is made
short, and for example, a headphone device as a whole system. It is possible to satisfy the
response speed requirements of the signal processing system of the noise canceling system of
the present invention. That is, it is possible to easily realize the noise canceling system by the
digital circuit method. And, by realizing the noise canceling system by the digital circuit, the
implementation of the function which was difficult by the analog circuit or the high sound quality
etc. will be realized, and the use for the user The value increases. Further, as in the present
invention, by providing a plurality of systems giving cancellation signal characteristics, it is
possible to assign specific signal processing functions to each system and share them, for
example, noise cancellation The system performance can be improved, and the design freedom
can be improved.
[0010]
10-04-2019
4
As a best mode for carrying out the present invention (hereinafter referred to as an embodiment),
a headphone device equipped with a noise canceling system will be exemplified. Therefore, prior
to describing the configuration as the present embodiment, the basic concept of the noise
canceling system corresponding to the headphone device will be described.
[0011]
As a basic method of such a noise canceling system compatible with a headphone device, a
method of performing servo control by a feedback method and a feed forward method are
known. First, the feedback method will be described with reference to FIG.
[0012]
Fig. 1 (a) schematically shows a model example of a noise canceling system by the feedback
method on the side of the right ear (R channel in 2-channel stereo by L (left) and R (right) of the
headphone wearer (user). Is shown. In the structure on the R channel side of the headphone
device here, first, the driver 202 is provided in a position corresponding to the right ear of the
user 500 wearing the headphone device in the housing unit 201 corresponding to the right ear. .
The driver 202 has the same meaning as a so-called speaker, and is driven by the amplified
output of the audio signal to emit the audio into space so as to be output.
[0013]
Then, as a feedback method, the microphone 203 is provided at a position near the right ear of
the user 500 in the housing portion 201. Depending on the microphone 203 provided in this
manner, the sound output from the driver 202 and the sound that is going to intrude into the
housing portion 201 from the external noise source 301 and reach the right ear, that is, the right
ear The external noise in the housing 302, which is an external sound to be generated, is picked
up. The noise source 301 leaks out as a sound pressure from, for example, a gap of an ear pad of
the housing as a cause of the generation of the noise 302 in the housing, or the housing of the
headphone device vibrates due to the sound pressure of the noise sound source 301. It can be
mentioned that this is transmitted into the housing part. In order to cancel (attenuate, reduce) the
noise 302 in the housing, such as a signal having an inverse characteristic to the sound signal
component of the external sound, from the sound signal obtained by collecting the sound by the
microphone 203 Signal (audio signal for cancellation) is generated, and this signal is fed back so
10-04-2019
5
as to be synthesized with an audio signal (audio source) of a necessary sound for driving the
driver 202. Thus, at the noise cancellation point 400 set at a position corresponding to the right
ear in the housing unit 201, the external sound is canceled by synthesizing the output sound
from the driver 201 and the component of the external sound. A sound is obtained and the user's
right ear will hear this sound. Then, by giving such a configuration also to the L channel (left ear)
side, a noise canceling system as a headphone device corresponding to normal L, R 2 channel
stereo can be obtained.
[0014]
The block diagram of FIG. 1 (b) shows an example of a basic model configuration of the noise
canceling system by the feedback method. In FIG. 1 (b), as in FIG. 1 (a), the configuration
corresponding to only the R channel (right ear) side is shown, and the L channel (left The same
system configuration is provided for the side of the ear. Also, the block shown in this figure
indicates one specific transfer function corresponding to a specific circuit part, circuit system,
etc. in a system of a noise canceling system by a feedback method, and in this case, the transfer
function block Do. The character shown in each transfer function block represents the transfer
function of the transfer function block, and the voice signal (or voice) is transferred to the
transfer function indicated there each time it passes through the transfer function block. Is to be
given. First, the sound collected by the microphone 203 provided in the housing unit 201 is a
transfer function block corresponding to the microphone 203 and a microphone amplifier that
amplifies the electric signal obtained by the microphone 203 and outputs the sound signal. It will
be obtained as an audio signal via 101 (transfer function M). The voice signal having passed
through the transfer function block 101 is input to the synthesizer 103 via a transfer function
block 102 (transfer function -β) corresponding to an FB (FeedBack) filter circuit. The FB filter
circuit is a filter circuit in which the characteristic for generating the above-described canceling
audio signal is set from the audio signal obtained by collecting the sound by the microphone
203, and the transfer function thereof is represented as -β. It is
[0015]
Also, here, the audio signal S of the audio sound source, which is the content such as music, is
assumed to be subjected to equalization by the equalizer, and the synthesizer via the transfer
function block 107 (transfer function E) corresponding to this equalizer It is supposed to be
input to 13.
[0016]
10-04-2019
6
In the synthesizer 103 here, the above two signals are synthesized by addition.
The audio signal synthesized in this manner is amplified by the power amplifier and output as a
drive signal to the driver 202, whereby the driver 202 is output as audio. That is, the voice signal
from the synthesizer 103 passes through the transfer function block 104 (transfer function A)
corresponding to the power amplifier, and further passes through the transfer function block
105 (transfer function D) corresponding to the driver 202 as voice. It is released into space. The
transfer function D of the driver 202 is determined by, for example, the structure of the driver
202.
[0017]
Then, the voice output by the driver 202 passes through the transfer function block 106
(transfer function H) corresponding to the spatial path (spatial transfer function) from the driver
202 to the noise cancel point 400, and the noise cancel point 400 , And will be combined with
the in-housing noise 302 in the space here. Then, as the sound pressure P of the output sound
that is to reach, for example, the right ear from the noise cancellation point 400, the sound of the
noise sound source 301 entering from the outside of the housing unit 201 is canceled.
[0018]
In the noise canceling system model shown in FIG. 1 (b), the sound pressure P of the output
sound is N for the noise 302 in the housing and S for the audio signal of the audio source. The
transfer function shown in the transfer function block, M, −β, E, A, D, H, is represented as
follows. In the equation (1), focusing on the noise 302 in the housing, it can be seen that N is
attenuated by a coefficient represented by 1 / (1 + ADHMβ). However, in order for the system of
the equation shown in (Equation 1) to operate stably without oscillating in the noise reduction
target frequency band, it is necessary that
[0019]
Generally speaking, the absolute value of the product of each transfer function in the feedback
type noise canceling system is expressed by 1 << | ADHMβ |, and combined with the stability
10-04-2019
7
determination of Nyquist in classical control theory. The equation (2) can be interpreted as
follows. Here, in the system of the noise canceling system shown in FIG. 1B, consider the system
represented by (−ADHMβ) obtained by cutting one portion of the loop portion related to N
which is the noise 302 in the housing . We will call this system "open loop" here. As an example,
if the transfer function block 101 corresponding to the microphone and the microphone
amplifier and the transfer function block 102 corresponding to the FB filter circuit are portions
to be disconnected, the above-described open loop can be formed.
[0020]
The above-mentioned open loop is assumed to have, for example, the characteristics shown by
the Bode diagram of FIG. In this Bode diagram, the frequency is shown on the horizontal axis, the
gain is shown on the lower half, and the phase is shown on the upper half. When this open loop
is targeted, the following two conditions need to be satisfied in order to satisfy Equation (2)
based on the stability determination of Nyquist. Condition 1: Phase 0 deg. When passing through
the point (0 degrees), the gain must be less than 0 dB. Condition 2: When the gain is 0 dB or
more, the phase 0 deg. Do not include the points of
[0021]
If the above two conditions 1 and 2 are not satisfied, the loop is subjected to positive feedback,
which causes oscillation (howling). In FIG. 2, phase margins Pa and Pb corresponding to the
above-mentioned condition 1 and gain margins Ga and Gb corresponding to the condition 2 are
shown. If the margin is small, the possibility of oscillation will increase due to various individual
differences among users who use the headphone device to which the noise canceling system is
applied, and variations in the state when the headphone device is worn. For example, in FIG. The
gain when passing through the point is smaller than 0 db, and according to this, gain margins Ga
and Gb are obtained. However, for example, temporarily phase 0 deg. The gain when passing
through the point is 0 dB or more and the gain margins Ga and Gb disappear, or the phase 0 deg.
If the gain when passing through the point is less than 0 dB, the gain margins Ga and Gb become
close to 0 dB, and oscillation will occur or the possibility of oscillation will increase. Similarly, in
FIG. 2, when the gain is 0 dB or more, the phase 0 deg. The phase margin Pa, Pb is obtained.
However, for example, when the gain is 0 dB or more, the phase 0 deg. It has passed the point of
Alternatively, phase 0 deg. When the phase margins Pa and Pb become smaller, the oscillation or
the possibility of oscillation increases.
10-04-2019
8
[0022]
Next, in the configuration of the noise canceling system of the feedback type shown in FIG. 1 (b),
in addition to the above-mentioned external voice (noise) canceling (reducing) function,
necessary sound (necessary sound) is The case of reproduction output will be described. Here, as
the required sound, for example, an audio signal S of an audio sound source as content such as
music is shown. In addition, as this audio | voice signal S, it is considered besides the thing of
such a musical thing or the content according to this. For example, when the noise canceling
system is applied to a hearing aid or the like, a microphone provided outside the housing to pick
up the surrounding necessary sound (different from the microphone 203 provided in the system
for noise cancellation) It becomes an audio signal obtained by sound collection. When applied to
what is called a so-called headset, it becomes an audio signal such as the other party's speaking
voice received by communication such as telephone communication. That is, the audio signal S
corresponds to general audio that needs to be reproduced and output according to the
application of the headphone device.
[0023]
First, in (Equation 1), the audio signal S of the audio source is focused. Then, it is assumed that
the transfer function E corresponding to the equalizer is set as having a characteristic according
to the equation represented by The transfer characteristic E is substantially the reverse
characteristic (1 + open loop characteristic) with respect to the open loop when viewed on the
frequency axis. Then, substituting the equation of the transfer function E shown by this equation
3 into the equation 1, the sound pressure P of the output sound in the model of the noise
canceling system shown in FIG. Can be represented. Of the transfer functions A, D, H shown in
the term of ADHS in (Eq. 4), first, the transfer function A corresponds to the power amplifier, the
transfer function D corresponds to the driver 202, and the transfer function H is noise from the
driver 202 Since it corresponds to the space transfer function of the path up to the cancellation
point 400, if the position of the microphone 203 in the housing unit 201 is in a position close to
the ear, the sound signal S has a noise cancellation function. It can be seen that the same
characteristics as a normal headphone can be obtained.
[0024]
Next, the noise canceling system by the feedforward method will be described. FIG. 3A shows a
configuration of the side corresponding to the R channel as an example of a noise canceling
10-04-2019
9
system according to the feedforward method, as in FIG. 1A. In the feed-forward method, the
microphone 203 is provided outside the housing portion 201 in such a manner that voices
arriving from the noise source 301 can be picked up. Then, an external voice collected by the
microphone 203, that is, a voice that has arrived from the noise source 301 is collected to obtain
a voice signal, the voice signal is subjected to an appropriate filtering process, and the canceling
audio signal is obtained. Will be generated. Then, this canceling audio signal is synthesized with
the audio signal of the required sound. That is, the cancel audio signal that electrically simulates
the acoustic characteristic from the position of the microphone 203 to the position of the driver
202 is synthesized with the audio signal of the necessary sound. Then, by causing the driver 202
to output an audio signal in which the canceling audio signal and the audio signal of the required
sound are synthesized in this manner, the noise source 301 to the housing portion 201 can be
obtained as the sound obtained at the noise cancellation point 400. It is made to be able to hear
what the sound which has invaded into the inside was canceled.
[0025]
FIG. 3B shows the configuration of the side corresponding to one channel (R channel) as a basic
model configuration example of the noise canceling system by the feedforward method. First, the
sound collected by the microphone 203 provided outside the housing portion 201 is obtained as
an audio signal via the transfer function block 101 having the transfer function M corresponding
to the microphone 203 and the microphone amplifier. Next, the voice signal that has passed
through the transfer function block 101 is input to the synthesizer 103 via a transfer function
block 102 (transfer function −α) corresponding to an FF (Feed Forward) filter circuit. The FF
filter circuit 102 is a filter circuit in which the characteristic for generating the above-described
canceling audio signal is set from the audio signal obtained by collecting the sound by the
microphone 203, and the transfer function thereof is expressed as −α. It is
[0026]
Also, the audio signal S of the audio source here is directly input to the synthesizer 103. The
audio signal synthesized by the synthesizer 103 is amplified by the power amplifier and output
to the driver 202 as a drive signal, so that the driver 202 is output as audio. That is, also in this
case, the audio signal from the synthesizer 103 passes through the transfer function block 104
(transfer function A) corresponding to the power amplifier, and further the transfer function
block 105 (transfer function D) corresponding to the driver 202. It is emitted into space as voice
via. Then, the voice output by the driver 202 passes through the transfer function block 106
(transfer function H) corresponding to the spatial path (spatial transfer function) from the driver
10-04-2019
10
202 to the noise cancel point 400, and the noise cancel point 400 , Where it will be synthesized
in space with the noise 302 in the housing.
[0027]
Also, as shown by a transfer function block 110, the path from the noise source 301 to the noise
cancellation point 400 until the sound emitted from the noise source 301 intrudes into the
housing portion 201 and reaches the noise cancellation point 400. A transfer function (space
transfer function F) corresponding to is given. On the other hand, the microphone 203 picks up a
voice that is supposed to arrive from the noise source 301 which is an external voice, but at this
time, the sound (noise) emitted from the noise source 301 is the microphone 203. By the time it
reaches, as shown as the transfer function block 111, the transfer function (spatial transfer
function G) corresponding to the path from the noise source 301 to the microphone 203 will be
given. As an FF filter circuit corresponding to the transfer function block 102, a transfer function
-α is set in consideration of the above space transfer functions F and G. As a result, the sound
pressure of the noise source 301 entering from the outside of the housing portion 201 is
canceled as the sound pressure P of the output sound that is to reach the right ear, for example,
from the noise cancellation point 400.
[0028]
In the system of the noise canceling system according to the feedforward method shown in FIG.
3B, the sound pressure P of the output sound is N for the noise emitted from the noise source
301 and the audio signal of the audio source. After assuming S, the transfer functions
represented in each transfer function block, M, -α, E, A, D, H, are used to be represented as
follows. Also, ideally, the transfer function F of the path from the noise source 301 to the
cancellation point 400 can be expressed as Next, when the equation shown in (Equation 6) is
substituted into (Equation 5), the first term and the second term on the right side will be offset.
From this result, the sound pressure P of the output sound can be expressed as In this way, it is
indicated that the sound arriving from the noise source 301 is canceled, and only the audio
signal of the audio source is obtained as voice. That is, in theory, in the right ear of the user, the
noise-cancelled voice can be heard. However, in reality, it is difficult to configure a complete FF
filter circuit that can provide a transfer function such that equation (6) completely holds. In
addition, the shape of the ear by a person, the individual difference in how to wear a headphone
device is relatively large, and the change in the relationship between the noise generation
position and the microphone position is a noise particularly in the middle and high frequency
band. It is known to affect the reduction effect. For this reason, with regard to the middle and
10-04-2019
11
high frequencies, active noise reduction processing is avoided, and passive sound insulation is
often performed mainly depending on the structure of the housing of the headphone device and
the like. Moreover, to describe for confirmation, (Equation 6) means imitating the transfer
function of the path from the noise source 301 to the ear with the electric circuit including the
transfer function -α.
[0029]
Further, in the noise canceling system of the feedforward method shown in FIG. 3A, since the
microphone 203 is provided on the outside of the housing, the noise canceling system of the
feedback method of FIG. However, it can be arbitrarily set in the housing portion 201 in
accordance with the listener's ear position. However, normally, the transfer function -α is fixed,
and at the design stage, it is determined that some target characteristic is targeted. On the other
hand, the shape of the ear differs depending on the listener. For this reason, there is also a
possibility that a phenomenon such as a sufficient noise cancellation effect can not be obtained
or noise components are added in a non-reversed phase to cause abnormal noise. From these
facts, it is generally considered that it is difficult to obtain a sufficient amount of noise
attenuation (cancellation amount) although the feedforward method has low possibility of
oscillation and high stability. On the other hand, it is said that the feedback method needs
attention to the stability of the system instead of expecting a large noise attenuation amount. As
described above, the feedback method and the feed forward method each have features.
[0030]
By the way, as a present condition, the noise canceling system of the headphone device which is
practically used in the consumer and is put to practical use is an analog system adopting an
analog circuit. However, if the signal processing system of the noise canceling system is digital
signal processing, it is easy to provide various functions such as changing the characteristics of
the noise canceling system and the operation mode, switching, etc. It is possible, and high sound
quality can be achieved. Thus, the merit of digitizing the noise canceling system is great.
[0031]
Therefore, FIG. 4 shows one configuration example that can be considered properly when it is
assumed that a noise canceling system of a headphone device is constructed using digital devices
10-04-2019
12
known at present. The noise canceling system shown in this figure is configured based on the
feed forward method shown in FIG. Also, although the headphone device (hereinafter simply
referred to as a headphone) 1 shown here corresponds to a two-channel stereo by L (left) and R
(right), the system configuration of this figure is L It corresponds to either channel or R channel.
Also, in this figure, in order to make the explanation simple and easy to understand, the signal
system of the audio sound source to be originally heard is omitted, and only the system for
canceling the external sound (noise source) is shown. .
[0032]
In FIG. 4, first, the microphone 2F is for picking up an external sound including an external sound
(external noise) around the headphone 1 to be canceled. In the case of the feed-forward system,
in general, the microphone 2F is generally provided outside the housing (headphone unit)
corresponding to each of the L and R single-side channels of the headphone 1 . Note that, in this
figure, a microphone 2F provided on the headphone unit 1c corresponding to one of L and R
channels among the headphone units 1c and 1d is shown. A signal obtained by collecting an
external sound by the microphone 2F is amplified by the amplifier 3 and is input to the A / D
converter 50 as an analog audio signal. In the following description, it is assumed that the
reference sampling frequency indicated by fs (1 fs) corresponds to the sampling frequency of the
digital audio source that is originally intended to be heard by the headphone 1. As a specific
example of the digital audio source here, a digital audio signal recorded on a CD (compact disc),
etc., which has fs = 44.1 kHz and the number of quantization bits = 16 can be mentioned. . Of
course, other types of digital audio sources may be adopted, such as the one with fs = 48 kHz.
[0033]
The A / D converter 50 in this case is, for example, one component or device, and the input
analog signal is digitalized in the form of a pulse code modulation (PCM) signal according to a
predetermined sampling frequency and the number of quantization bits. Convert to a signal and
output. For this purpose, for example, as shown in the drawing, the ΔΣ modulator 4 and the
decimation filter 5 are provided. The ΔΣ (delta sigma) modulator 4 converts the input analog
audio signal into, for example, a 1-bit digital signal with a sampling frequency of 64 fs. This
digital signal is reduced in sampling frequency to, for example, 1 fs by the decimation filter 5,
and the number of quantization bits is set to a predetermined multi-bit (here, 16 bits)
corresponding to the digital audio source. Is converted to a PCM signal, and output as a digital
signal from the A / D converter 50. Further, in the device as such an A / D converter 50, the
decimation filter 5 described above is generally formed by a linear phase FIR (Finite Impulse
10-04-2019
13
Response) system (linear phase FIR) having linear phase characteristics. doing. The digital signal
to be processed in this noise canceling system is an audio signal. Therefore, on the premise of
faithful sound reproduction, it is ideally found that no distortion of the waveform occurs. If linear
phase characteristics are given by FIR, the above waveform distortion does not occur. In addition,
in the case of an FIR system, it is possible to easily obtain an accurate linear phase characteristic,
as is well known. For such reasons, the digital filter as the decimation filter 5 is configured by the
linear phase FIR. In order to make the digital filter of the FIR system linear phase type, for
example, with respect to tap coefficients, peak values of coefficients are set at the centers of the
number of taps (orders) to be symmetrical. It can be realized by setting.
[0034]
The digital signal output from the A / D converter 50 is input to the DSP 60. In this case, the DSP
60 is a part that performs required signal processing for generating an audio signal of at least
sound to be output from the driver 1a of the headphone 1 by digital signal processing, and
provides the necessary function by programming. It is made to be able. As understood from the
following description, the audio signal to be output from the driver 1a of the headphone 1 is an
audio signal of a digital audio source and an audio for allowing the external sound collected by
the microphone 2 to be canceled and canceled. The signal (audio signal for cancellation) is
synthesized. Further, the DSP 60 is provided as, for example, one chip or device, and corresponds
to a predetermined PCM signal format (here, sampling frequency = 1 fs (= 44.1 kHz), number of
quantization bits = 16 bits) It is configured to perform digital signal processing. The PCM signal
format supported by the DSP is set on the premise of being adapted to the format of a digital
audio source to be synthesized with the noise canceling audio signal in the noise canceling
system.
[0035]
In this figure, the noise cancellation signal processing unit 6 is shown as a signal processing
functional block implemented in the DSP 60. The noise cancellation signal processing unit 6 is
configured by a digital filter that inputs / outputs data corresponding to the above-mentioned
PCM signal format. The noise cancellation signal processing unit 6 corresponds to the FF filter
circuit of FIG. 3 and is a digital signal output from the A / D converter 50, that is, a digital audio
signal corresponding to the external sound collected by the microphone 2F. Enter Then, using
this input signal, as a sound to be output from the driver 1a, an audio signal of sound having a
function of canceling external sound that reaches the ear of the headphone wearer
corresponding to the driver 1a and is heard (audio for cancellation Signal). As the simplest audio
10-04-2019
14
signal for such cancellation, for example, the reverse characteristic, reverse to the audio signal
input to the noise cancellation signal processing unit 6, that is, the audio signal obtained by
picking up the external sound It is a signal that becomes a phase. In addition, in practice, a
characteristic (corresponding to the transfer characteristic −α in FIG. 3) in consideration of the
transfer characteristic such as a circuit or space in the system of the noise canceling system is
given.
[0036]
In this case, the digital signal from the noise cancellation signal processing unit 6 which is output
from the DSP 60 is synthesized by the synthesizer 12 with the signal of the digital audio source
of the PCM signal format with sampling frequency = 1 fs and the number of quantization bits =
16 bits. Then, they are input to the D / A converter 70. The D / A converter 70 is also, for
example, one chip part, which receives the PCM format digital signal converted by the A / D
converter 50 described above and converts it into an analog signal. For example, as shown in the
drawing, the interpolation filter 7, the noise shaper 8, the PWM circuit 9, and the power drive
circuit 10 are provided.
[0037]
The digital signal input to the D / A converter 70 is first input to the interpolation filter 7. The
interpolation (oversampling) filter 7 converts and outputs the input digital signal so as to raise
the sampling frequency to a sampling frequency obtained by multiplying the sampling frequency
by a factor represented by a power of 2 by a predetermined factor. . In this case, the sampling
frequency is increased to 8 fs. In addition, with regard to the number of quantization bits of the
output signal, in this case, conversion is performed such that the number of bits is smaller than
16 bits at the time of input. The interpolation filter 7 is also formed by a linear phase FIR system
for the same reason as the decimation filter 5 described above.
[0038]
The digital signal output from the interpolation filter 7 is subjected to processing called noise
shaping by the noise shaper 8. The signal after this noise shaping is, for example, a sampling
frequency (here, 16 fs) obtained by multiplying the sampling frequency at the input by a factor
represented by a power of 2 (in this case, 16 fs). Converted to bit number format. As is well
10-04-2019
15
known, noise shaping is obtained as a result of the ΔΣ modulation process, and hence the noise
shaper 8 can be realized by the ΔΣ modulator. That is, the digital noise canceling system shown
in this figure adopts a configuration to which ΔΣ modulation is applied for A / D conversion
and D / A conversion.
[0039]
The output of the noise shaper 8 is subjected to PWM modulation by a PWM (Pulse Width
Modulation) circuit 9 and converted into a 1-bit string signal, and then input to the power drive
circuit 10 in the subsequent stage. The power drive circuit 10 is formed of, for example, a
switching drive circuit that switches and amplifies a 1-bit string signal at high voltage, and a low
pass filter (LC low pass filter) for converting the amplified output into an audio signal waveform.
An amplified output as an audio signal is obtained. Here, the output of the power drive unit 64 is
taken as the output of the D / A converter 70. The amplified output from the D / A converter 70
is supplied as a drive signal to the driver 1a via the DC insulating capacitor C1 after a
predetermined unnecessary band component is removed by the filter 11, for example. .
[0040]
The sound output from the driver 1a driven in this manner is a combination of the sound
component of the digital audio source and the sound component of the noise cancellation audio
signal, but the sound of the noise cancellation audio signal is Depending on the component, an
external sound reaching the ear corresponding to the driver 1a from the outside is canceled
(canceled). As a result, ideally, the external sound is canceled and the sound of the digital audio
source is relatively emphasized as the sound that the headphone wearing person listens to by the
ear corresponding to the driver 1a.
[0041]
The configuration shown in FIG. 4 uses, for example, an A / D converter, a DSP, a D / A converter,
etc. which are easily available for consumer use, and in the present situation, for example, as a
digital noise canceling system In the case of trying to create an audio source such as a CD, the
configuration can be properly considered first.
[0042]
10-04-2019
16
However, in the above configuration, it has been found that it is difficult to obtain a sufficient
noise cancellation effect in reality.
This is because the signal processing time (propagation time) possessed by the actual devices as
the A / D converter 50 and the D / A converter 70, that is, the delay between the input and the
output is considerably large. Essentially, these devices are supposed to handle audio signals as
audio sources such as ordinary music pieces in a single way, so even if the signal processing
causes delays, this is not a problem It is However, when such a device is used as it is for a noise
cancellation ring system, the delay becomes so large that it can not be ignored. That is, in the
entire system of the noise canceling system configured using these devices, there is a large delay
in the time (response speed) from the external sound picked up by the microphone 2 to the
sound output by the driver. It will occur. Due to this delay, for example, it becomes difficult to
cancel the external sound due to the sound component for noise cancellation output from the
driver. For example, even if only the A / D converter 50 is taken, if the delay at a sampling
frequency of 44.1 kHz is for 40 samples, the phase rotation of a signal of about 550 Hz or more
will be 180 ° or more. When the delay is increased to such an extent, not only it is difficult to
obtain the noise cancellation effect, but also there may be a phenomenon that the external sound
is emphasized. As described above, in the configuration of the digital noise canceling system as
illustrated in FIG. 1, the allowable noise canceling effect is limited to the range of the frequency
band lower than about 550 Hz, for example, the audible band. As compared with the standard
case of 20 Hz to 20 kHz, the noise cancellation effect can be obtained only in a very narrow
range of the lower frequency band. That is, it is difficult to obtain a noise cancellation effect that
is sufficient for practical use. This is the reason that most of the noise canceling systems of
headphone devices currently put into practical use are analog systems.
[0043]
However, as described above, the advantages obtained by digitizing the noise canceling system
are significant. Therefore, as the present embodiment, as will be described later, with regard to
the noise canceling system of the headphone device, a configuration for solving the abovementioned problem of delay while using digital system is proposed. It is
[0044]
First, the process by which the inventor of the present application has come to configure the
10-04-2019
17
noise canceling system of the present embodiment will be described with reference to FIG. In FIG.
5, the same parts as in FIG. 4 are assigned the same reference numerals and explanation thereof
is omitted. FIG. 5A shows a system of noise cancellation signals including the decimation filter 5,
the noise cancellation signal processing unit 6 (DSP 60), and the interpolation filter 7 in the noise
canceling system having the configuration shown in FIG. It extracts and shows. Although the
decimation filter 5 is shown as a single block in the A / D converter 50 in FIG. 4, the inventor of
the present application has made it possible to use the decimation filter 5 as shown in FIG. Of the
decimation filters 5A and 5B and connect them in series. The decimation filter 5 converts a signal
of sampling frequency = 64 fs to a signal of 1 fs and outputs it, as is understood from the
explanation of FIG. 4, that is, down-sampling the sampling frequency to 1/64. Therefore, in the
configuration in FIG. 5A, decimation filters 5A and 5B that perform 1/8 downsampling are
respectively configured for decimation filters 5 that perform 1/64 downsampling, and
decimation filters 5A The decimation filter 5B is connected in series in the latter stage. According
to this configuration, the signal of sampling frequency = 64 fs input to the decimation filter 5 is
first converted to a signal of sampling frequency = 8 fs by the decimation filter 5A and output.
Subsequently, the signal of sampling frequency = 8 fs is input to the decimation filter 5B,
whereby the signal is converted into a signal of sampling frequency = 1 fs in the PCM format. In
this manner, depending on the series connection of the decimation filters 5A-5B, 1/64
downsampling is performed in total as represented by 1/8 × 1/8. As described for confirmation,
also in FIG. 5A, the processing of the signal after passing through the decimation filter 5
(decimation filter 5B) is the same as that in FIG. That is, the signal (PCM signal) of sampling
frequency = 1 fs output from the decimation filter 5 is input to the noise cancellation signal
processing unit 6.
The noise cancellation signal processing unit 6 generates and outputs a cancellation audio signal
by giving predetermined characteristics to the input signal as signal processing corresponding to
a signal of PCM format at sampling frequency = 1 fs. The cancellation audio signal output from
the cancellation signal processing unit 6 is in PCM format with sampling frequency = 1 fs, but the
interpolation filter 7 inputs this cancellation audio signal and up-samples (interpolation) By
performing the sampling frequency of 8 fs.
[0045]
Here, as for the system consisting of the decimation filter 5B, the noise cancellation signal
processing unit 6, and the interpolation filter 7 shown by the dashed dotted line in FIG. 5 (a), this
system is composed of an input signal and an output signal. The sampling frequency of is the
same at 8 fs. Note that, in the following, the system indicated by the one-dot and dash line is also
referred to as an 8 fs input / output signal processing system. If this 8 fs input / output signal
10-04-2019
18
processing system is considered as one black box, a PCM signal of sampling frequency = 8 fs is
input, and an audio signal for noise cancellation by the PCM format of the same sampling
frequency = 8 fs is generated and output. It can be viewed as a part that executes digital signal
processing (noise cancellation signal processing).
[0046]
Then, based on the fact that the 8 fs input / output signal processing system is regarded as a part
having the above-described function, it can be considered that the configuration shown in FIG.
That is, only the noise cancellation signal processing unit 6A is provided as the 8 fs input /
output signal processing system. Then, a signal of sampling frequency = 8 fs is directly input by
this noise cancellation signal processing unit 6A, and an audio signal for noise cancellation with
sampling frequency = 8 fs is generated by digital signal processing corresponding to a PCM
signal format of 8 fs. Output.
[0047]
When comparing the configuration of FIG. 5B with the configuration of FIG. 5A, in FIG. 5B, the
decimation filter 5 first performs sampling frequency conversion of 1/8 times. The decimation
filter (5B) for performing is omitted, and furthermore, the interpolation filter 7 for performing
sampling frequency conversion of 8 times is omitted. As described above, in the configuration
shown in FIG. 4, the delay in A / D converter 50 and D / A converter 70 is large. However, as a
factor of these delays, the decimation filter in A / D converter 50 It is known that the delay by 5
is dominant, and in the D / A converter 70, the delay by the interpolation filter 7 is dominant.
From this, in FIG. 5 (b), since the signal input / output to / from the noise cancellation signal
processor 6A does not pass through the decimation filter 5B and the interpolation filter 7, 8 fs
shown in FIG. 5 (a) is input. As compared with the output signal processing system, that is, the
configuration of FIG. 4, the signal delay is greatly shortened. And, as the signal delay in the noise
cancellation signal processing system is shortened in this way, the frequency band of the voice
where noise cancellation is considered to work effectively can be derived as described above. It is
possible to expand it to the range of That is, by adopting the configuration of FIG. 5B, the
problems of the noise canceling system shown in FIG. 4 are eliminated.
[0048]
10-04-2019
19
Here, in the case where the noise canceling system is actually configured according to the model
shown in FIG. 5 (b), the configuration of the noise cancellation signal processing unit 6A will be
considered. First, the noise cancellation signal processing unit 6 shown in FIG. 5A is actually
realized by first programming the DSP as described in FIG. Moreover, as a form of a digital filter,
it is common to use FIR. Therefore, even in the case of configuring the noise canceling system
based on FIG. 5 (b), the noise canceling signal processing unit 6A may first be configured as a
digital filter (FIR filter) of an FIR included in the DSP. It is a matter of course to be considered.
[0049]
However, the sampling frequency of the signal processed by the noise cancellation signal
processing unit 6A is 8 fs, which is considerably higher than that of the noise cancellation signal
processing unit 6 of FIG. . Then, due to the relationship with the clock, the noise cancellation
signal processing unit 6A is smaller than the noise cancellation signal processing unit 6 as the
number of operations (number of processing steps) that can be performed per one cycle of the
sampling frequency. Specifically, assuming that the clock is 1024 fs, the noise cancellation signal
processing unit 6A whose corresponding sampling frequency is 8 fs means that the number of
operations per sample period is 1024/8 = 128. On the other hand, the noise cancellation signal
processing unit 6 with a corresponding sampling frequency of 1 fs is 1024/1 = 1024 times. This
means that if the noise cancellation signal processing unit 6A is configured to use a DSP, it can
not obtain the arithmetic processing capability of a DSP that executes digital signal processing
corresponding to sampling frequency = 1 fs. means. From this point of view, it is preferable to
configure the noise cancellation signal processing unit 6A using hardware. In addition, since the
characteristic as the canceling audio signal is correspondingly complicated, after the noise
canceling signal processing unit 6A is configured by the FIR filter, it is possible to execute the
signal processing for the noise cancellation target of the wide audio frequency band as much as
possible. In the case of attempting to configure, a huge number of orders (number of taps) are
required, and the resources for processing become very large. Therefore, the inventor of the
present application examined that the noise cancellation signal processing unit 6A in the case
where the model shown in FIG. 5 (b) is actually configured is constituted by a digital filter (IIR
filter) of Infinite Impulse Response (IIR). It has been confirmed that the IIR filter can also provide
the necessary and sufficient characteristics as an audio signal for noise cancellation. That is, it
has been confirmed that an IIR filter that can be formed with an order smaller than that of an FIR
filter and resources smaller than that of an FIR filter can be sufficiently used to provide an equal
signal characteristic as a noise canceling audio signal. Thus, one conclusion was obtained that it
is appropriate to configure the noise cancellation signal processing unit 6A in the configuration
shown in FIG. 5 (b) using a hardware IIR filter.
10-04-2019
20
[0050]
As described above, by adopting the configuration of FIG. 5B, the decimation filter 5B and the
interpolation filter 7 are omitted from the noise cancellation signal processing system, and the
signal delay due to this is not generated. The frequency band in which the effective noise
cancellation effect is obtained will be expanded to a higher frequency. That is, it is possible to
obtain noise cancellation performance that is sufficient for practical use, though digital signal
processing. However, when actually trying to configure a noise canceling system, it is pure,
including freedom in filter characteristics and design that is an advantage of being digital, cost
reduction, size and weight reduction, etc. It is necessary to satisfy several conditions other than
the noise cancellation performance. When the noise canceling system based on FIG. 5B is to be
actually configured, the execution portion (noise canceling signal processing unit 6A) of the noise
canceling signal processing is configured by only dedicated hardware, for example. However, in
this case, for example, the setting of the filter characteristic becomes fixed, and the change
setting of the filter characteristic according to the switching operation, the adaptive control, or
the like tends to be restricted. Incidentally, with regard to the degree of freedom such as changes
in filter characteristics and design, a DSP designed to execute digital signal processing in
accordance with a program is advantageous. Further, since the noise cancellation signal
processing is inherently complicated, even if an IIR filter by hardware is adopted for the noise
cancellation signal processing unit 6A, a corresponding resource is required. For this reason,
depending on the conditions, the noise cancellation signal processing unit 6A, which is hardware,
may cost more than an allowance, or may have a circuit size or mounting area larger than an
allowance. It is believed that there is. In view of this, as shown in FIG. 5B, it is not so realistic to
actually obtain a noise canceling system that executes digital signal processing as noise
cancellation signal processing using only hardware. It means that it is not a goal.
[0051]
Therefore, as shown in FIG. 5C, the inventor of the present invention has two systems of the 8 fs
input / output signal processing system: a system including the noise cancellation signal
processing unit 6A; and a system including the noise cancellation signal processing unit 6 It is
intended to consider a configuration provided in parallel. As described above, as the signal delay
as the noise canceling voice increases in the noise canceling system, it becomes more difficult to
obtain the noise canceling effect for the high frequency band. This means that on the low band
side, even if there is a considerable signal delay, it is easy to obtain the noise cancellation effect.
Based on this, in the configuration shown in FIG. 5C, the noise cancellation signal processing unit
6 generates a noise cancellation signal for performing noise cancellation on low frequencies in all
audio frequency bands to be subjected to noise cancellation. It was decided to be configured as
10-04-2019
21
follows. On the other hand, in the noise cancellation signal processing unit 6A, the noise
cancellation signal for performing the noise cancellation on the middle and high frequencies
considered to be higher than the low frequency in all the audio frequency bands to be subjected
to noise cancellation is used. It is to be configured to generate. In such a configuration, the noise
cancellation signal processing unit 6A in charge of the middle and high frequencies in the entire
audio frequency band to be subjected to noise cancellation executes noise cancellation signal
processing as main processing, and one noise cancellation signal processing unit 6 As a subprocess, it can be viewed as a part that executes the noise cancellation signal processing unit for
the low band.
[0052]
With such a configuration, first, for the noise cancellation signal processing unit 6A configured
by the IIR filter of hardware, a noise cancellation audio signal whose noise cancellation target is a
frequency band on only the middle high band side excluding the low band. As compared with the
case where the entire voice frequency band including the low band is subjected to the noise
cancellation, the reduction of the necessary resource amount is also promoted accordingly.
Further, by reducing the hardware resources in this manner, the power consumption of the noise
cancellation signal processing unit 6A is also reduced. This leads to the reduction of the power
consumption of the noise canceling system. For example, when the noise canceling system is
driven by a battery, it is expected to extend the life of the battery. Further, as described above,
the noise cancellation signal processing unit 6 that executes digital signal processing
corresponding to the sampling frequency = 1 fs has the number of operations compared to the
noise cancellation signal processing unit 6A corresponding to the sampling frequency = 8 fs.
Since the processing capability is high in point, there is no problem in configuring with DSP.
Therefore, if the noise cancellation signal processing unit 6 is configured as one function of the
DSP, for example, it is easily possible to dynamically change and set the filter characteristics.
That is, the degree of freedom in signal processing is improved.
[0053]
Thus, in the configuration of FIG. 5C, the problem of the deterioration of the noise cancellation
performance due to the delay of the noise cancellation audio signal is first solved. Further, with
the noise cancellation signal processing unit 6A configured by hardware logic and corresponding
to the sampling frequency = 8 fs, the resources can be further reduced, and at the same time, a
high degree of freedom can be obtained regarding the noise cancellation signal processing. The
inventor of the present application has concluded that, as a noise canceling system, the model
10-04-2019
22
configuration shown in FIG. 5 (c) would be the best at present based on the fact that the above
advantages can be obtained. It is. That is, the noise canceling system as the embodiment based on
the present invention is configured including the system of the noise canceling audio signal
based on the model form shown in FIG. 5C.
[0054]
Incidentally, in FIG. 5C, the system on the side of the noise cancellation signal processing unit 6A
is main and performs noise cancellation signal processing for the middle and high frequencies,
and the system on the side of the noise cancellation signal processing unit 6 is It has been
described that noise cancellation signal processing for the low frequency band is additionally
performed. As described above, in consideration of, for example, the cost, the board mounting
area, etc., the noise cancellation signal processing unit 6A configured by hardware reduces
resources as much as possible and reduces the size of the circuit. It can be said that it is required
to
[0055]
Therefore, the inventor of the present application assumes that resources should be reduced as
much as possible for the noise cancellation signal processing unit 6A, for example, because it is
desirable to give priority to cost as a noise canceling system or reduction in size and weight. I
examined it. As a result, as shown in FIG. 5D, under the same model mode as FIG. 5C, the noise
cancellation signal processing unit 6 is made to take charge of main noise cancellation signal
processing, and the noise cancellation signal processing unit We came up with a configuration in
which 6A was in charge of the noise cancellation signal processing as a sub. In this configuration,
first, for the noise cancellation signal processing unit 6, for example, of all the audio frequency
bands to be subjected to noise cancellation, an audio frequency of a predetermined high
frequency band or more where effective noise cancellation effect is difficult to be obtained.
Except for the band, noise cancellation is configured to be performed for the voice frequency
band as lower middle and lower bands. On the other hand, the noise cancellation signal
processing unit 6A is configured as, for example, a gain adjustment circuit that performs gain
adjustment on an input signal, or is configured to obtain a moving average based on values of
several samples. The signal processing operation of the noise cancellation signal processing unit
6A is, for example, to compensate for the high frequency noise cancellation signal processing
that is insufficient on the noise cancellation signal processing unit 6 side (generation of the high
frequency noise cancellation audio signal). Equivalent to.
10-04-2019
23
[0056]
And if it is the structure corresponding to the said FIG. 5 (d), as noise cancellation signal
processing part 6A, it can implement | achieve, for example by a FIR filter of about several taps.
That is, the resources can be very small, and when actually configured as hardware, they can be
made compact at low cost.
[0057]
Thus, in the present embodiment, as described with reference to FIGS. 5C and 5D, digital signal
processing corresponding to different sampling frequencies is performed on a system that
executes noise cancellation signal processing 2 By providing two systems, it is possible to obtain
a noise cancellation effect that is sufficient for practical use despite digital signal processing, and
to suppress hardware resources or circuit scale to a certain level or less, and a noise cancellation
signal. It is the setting freedom degree about processing.
[0058]
By the way, the fundamental difference when comparing FIG. 5 (a) (b) with FIG. 5 (c) (d) which is
the basis of the present embodiment is firstly FIG. 5 (a) (b) The noise cancellation signal
processing (generation of the noise cancellation audio signal) is performed by the digital signal
processing of only one system corresponding to the sampling frequency = 1 fs or the sampling
frequency = 8 fs. In the configuration of (d), noise cancellation signal processing is
simultaneously performed by one digital signal processing corresponding to sampling frequency
= 1 fs and one digital signal processing corresponding to sampling frequency = 8 fs. It can be said
that it is what it is doing.
That is, in the configuration of FIGS. 5 (a) and 5 (b), noise cancellation signal processing is
executed by digital signal processing corresponding to a specific single sampling frequency,
while in FIG. 5 (c) (d) In the above configuration, noise cancellation signal processing is
performed by digital signal processing of each system corresponding to two different sampling
frequencies. It should be noted that the configuration of FIG. 4 described above is equivalent to
that of FIG. 5A if stated for confirmation, and therefore is included in the category of the former
configuration. In the latter case, the output of the lower sampling frequency (1 fs) system is
upsampled (interpolated) up to the higher sampling frequency (8 fs), and the upsampled signal
and the sampling frequency Is added and output with the output of the higher system.
10-04-2019
24
[0059]
Then, based on the difference in the configuration described above, the former configuration
corresponding to the above FIGS. 5 (a) and (b) (and FIG. 4) is hereinafter also referred to as
“single pass” for the noise cancellation signal processing system. The latter configuration
corresponding to 5 (c) (d) is also referred to as "dual path".
[0060]
Hereinafter, a more specific configuration example as the noise canceling system of the present
embodiment based on the model configuration shown in FIGS. 5 (c) and 5 (d) will be described.
First, FIG. 6 is a block diagram showing a configuration example of a noise canceling system
according to the first embodiment. In this figure, the parts that are the same as in FIG. 4 are
given the same reference numerals, and descriptions of the same contents as in FIG. 4 are
omitted. Further, the noise canceling system shown in FIG. 6 also adopts a configuration based
on the feed forward method in the same manner as in FIG. 4 and corresponds to either one of L
and R stereo channels. is there. Also in the following embodiments, it is assumed that the
reference sampling frequency fs is 44.1 kHz corresponding to a digital audio source such as a
CD, for example.
[0061]
First, in the noise canceling system according to this embodiment, portions corresponding to the
A / D converter 50, the DSP 60, and the D / A converter 70 shown in FIG. 4 are one LSI (Large
Scale Integration) 100. It is configured to be contained in a physical structural unit as an
integrated circuit component. In addition, the LSI 100 is internally provided with two signal
processing units of an analog block 700 and a digital block 800. Analog block 700 corresponds
to the input / output of an analog signal, and ΔD modulator 4 which is the first stage in A / D
converter 50 and the power drive circuit 10 which is the last stage in D / A converter 70. Is
formed. Further, in this figure, the analog block 700 also includes the power supply unit 22 and
the oscillator 21. The power supply unit 22 supplies direct current power with a predetermined
voltage value to the circuit in the LSI 100. The oscillator 21 is configured to output a clock (CLK)
for a circuit in the LSI 100 (analog block 700, digital block 800) using, for example, a signal from
a crystal oscillator provided outside the LSI 100. In this embodiment, this clock frequency is
10-04-2019
25
1024 fs. The digital block 800 is a part forming a function corresponding to the A / D converter
50, the DSP 60, and the D / A converter 70, including input and output by digital signals
including parts other than the .DELTA..SIGMA. It will be formed including the site to be
performed. Also, here, the analog block 700 and the digital block 800 are chips that are
manufactured by different processes. That is, the LSI 600 in this embodiment is configured by
packaging at least a chip corresponding to the analog block 700 and a chip corresponding to the
digital block 800. Incidentally, since manufacturing of an analog circuit and a digital circuit as
one chip is also currently performed, it is possible to manufacture the analog block 700 and the
digital block 800 as one chip according to this. . That is, in the present embodiment, in
consideration of, for example, manufacturing efficiency and other conditions, it is determined
whether to configure the analog block 700 and the digital block 800 with separate chips or with
a single chip. Just do it.
[0062]
The functional block configuration of the noise canceling system shown in FIG. 6 is as follows.
First, the microphone 2F is attached to the outer casing of the headphone unit 1c in accordance
with the feed forward method. A signal obtained by collecting the sound by the microphone 2F is
amplified by the amplifier 3 and becomes an analog sound signal. This analog audio signal is
input to the LSI 100, and is first input to the Δ 変 調 modulator 4 in the analog block 700,
where, for example, the sampling frequency is 64 fs and the number of quantization bits is 1 bit
(64 fs, 1 bit) Converted to a digital signal of the form In this case, the digital signal as the output
of the ΔΣ modulator 4 is input to one input terminal of the switch SW1.
[0063]
The noise canceling system of the present embodiment takes into consideration that the
microphone input stage can also cope with the input from the digital microphone (digital
microphone) in consideration of the extensibility. Digital audio signals are enabled. For example,
the digital microphone is configured by integrating at least a microphone and a ΔΣ modulator
that converts a signal obtained by collecting the sound by the microphone into a 1-bit digital
audio signal. An input signal from the digital microphone is input to the other output terminal of
the switch SW1.
[0064]
10-04-2019
26
The switch SW1 is switched in such a manner that one of the two input terminals is selected and
connected to the output terminal. The output terminal is connected to the input of the decimation
filter 5A of the digital block 800. In any case, the output of the switch SW1 is a digital audio
signal based on the sound collected outside the headphone case in accordance with the
feedforward method. The digital audio signal output from the switch SW1 is input to the
decimation filter 5.
[0065]
The decimation filter 5A corresponds to the decimation filter 5 of FIG. 4 by series connection
with the decimation filter 5B of the latter stage. Since the decimation filters 5A and 5B are
respectively formed so as to decimate to the sampling frequency of 1/8, (1/8) × (1/8) =, by
serial connection of the decimation filters 5A and 5B. As represented by 1/64, the sampling
frequency will be decimated to 1/64. That is, similarly to the decimation filter 5, the 64 fs input
signal is converted to 1 fs. Here, the decimation filter 5A has fixed filter characteristics, whereas
the decimation filter 5B can be configured to vary the filter characteristics as described later.
[0066]
First, the decimation filter 5A converts the input 64 fs, 1-bit signal into a 8 fs, 24-bit signal by
executing so-called decimation processing to decimate data according to a predetermined
decimation pattern corresponding to the sampling period. Output. That is, the decimation filter
5A performs 1/8 decimation (downsampling) with respect to the processing of the sampling
frequency. This output is input to the decimation filter 5B and branched to be input also to the
noise cancellation signal processing unit 6A.
[0067]
The noise cancellation signal processing unit 6A is formed by a digital filter, and generates an
audio signal for noise cancellation with 8 fs and 24 bits and inputs it to the synthesizer 12 as
described later. In the configuration of the noise canceling system according to the present
embodiment, the noise canceling audio signal is generated also by the noise canceling signal
processing unit 6 in the DSP 60 as described later. Therefore, hereinafter, the one generated by
the noise cancellation signal processing unit 6 is referred to as a first noise cancellation audio
10-04-2019
27
signal, and the one generated by the noise cancellation signal processing unit 6A is referred to as
a second noise cancellation audio signal. Let's distinguish between the two.
[0068]
The decimation filter 5B performs 1/8 downsampling in the same manner as the above-described
decimation filter 5A. That is, the input 8-fs 24-bit signal is converted into, for example, a 1-fs 16bit PCM (Pulse Code Modulation) signal and output to the DSP 60.
[0069]
The DSP 60 is provided to receive a digital audio signal obtained based on the collected sound of
the microphone 2F and an audio signal as a digital audio source, and perform required signal
processing for each. Further, the DSP 60 in this case is configured to be capable of signal
processing corresponding to the format of a PCM signal of, for example, 1 fs, 16 bits. The signal
processing function executed by the DSP 60 can be obtained by programming. The program is
stored and held as data of an instruction, for example, in the flash memory 16. The DSP 60
appropriately executes signal processing by reading out a necessary instruction from this and
executing it.
[0070]
In the DSP 60 as the present embodiment, first, the noise cancellation signal processing unit 6
generates the first noise cancellation audio signal by using the signal input from the decimation
filter 5B. The noise cancellation signal processing unit 6 is formed by a digital filter. In addition,
the acoustic analysis processing unit 62 takes in the signal input from the decimation filter 5B,
executes a predetermined acoustic analysis process, and adapts to the analysis result to be a
predetermined functional part in the digital block 800. It is also possible to change and set the
characteristics of the digital filter. The acoustic analysis processing unit 62 can variably set the
filter characteristics for the digital filter as the noise cancellation signal processing unit 6 in the
same DSP 60 first. Further, it is possible to variably set the filter characteristics for the digital
filter as the noise cancellation signal processing unit 6A. In addition, it is possible to variably set
the filter characteristic for the digital filter as the decimation filter 5B. In addition, it is possible to
variably set the filter characteristic for the anti-imaging filter 7b in the interpolation filter 7.
Then, in order to vary the filter characteristics for the above-mentioned digital filter, first, the
10-04-2019
28
filter characteristics table is stored in advance in the flash memory 16, and then the filter
characteristics corresponding to the analysis result are read out. Ru. Then, parameters such as
the number of taps and coefficients as the read filter characteristics are set to obtain the
configuration of the digital filter having the desired characteristics. Further, for example, after
securing an area for holding the filter characteristic table also in the RAM 15, the acoustic
analysis processing unit 62 newly executes an operation or the like based on the analysis result
and the like to newly generate a filter characteristic. May be stored in the filter characteristic
table of the RAM 15. In this way, if the acoustic analysis processing unit 62 can generate the
filter characteristic adaptively according to the analysis result, the degree of freedom and
adaptability for the characteristic to be set to the digital filter can be further enhanced, and the
noise can be further improved. We can expect cancellation effect.
[0071]
In addition, it is possible to perform adjustment, correction, and the like regarding the sound
including the sound quality adjustment and the like on the signal of the digital audio source input
as described later by the equalizer 61 and output it.
[0072]
The first noise canceling audio signal (1 fs, 16 bits) output from the noise cancellation signal
processing unit 6 in the DSP 60 is input to the interpolation filter 7.
The interpolation filter 7 converts the input 1 fs, 16-bit signal into a signal of 8 fs, 24-bit by
executing processing to increase the sampling frequency by 8 and outputs the converted signal
to the synthesizer 12 . Also, here, the interpolation filter 7 is shown as being composed of an
oversampling circuit 7a and an anti-imaging filter 7b. That is, in the interpolation filter 7, the
oversampling circuit 7a first converts the input 1 fs, 16-bit signal into 8 fs, 24-bit format, and
then, for example, sampling as an image frequency component by the anti-imaging filter 7 b.
Signal processing is performed so as to remove frequency components higher than 1/2 of the
frequency 8 fs.
[0073]
Also, an audio signal as a digital audio source in this case is input to the DSP 60 in the form of 1
fs, 16 bits via the PCM interface 13. Further, in this case, it is branched and supplied to one input
10-04-2019
29
terminal of the switch SW2. In the DSP 60, the input digital audio source signal is subjected to
predetermined equalization processing and the like by the equalizer 61, and then input to the
other input terminal of the switch SW2.
[0074]
The switch SW2 is switched to connect either of the two input terminals to the output terminal.
The output terminal of the switch SW2 is connected to the input of the interpolation filter 14.
Therefore, in response to the switching of the switch SW2, a path for inputting the signal of the
digital audio source output from the PCM interface 13 to the interpolation filter 7 without
passing through the DSP 60, the digital output from the PCM interface 13 The switching is
performed in the path to which the signal of the audio source is input to the interpolation filter 7
through the DSP 60.
[0075]
As described above, to the interpolation filter 14, a digital audio signal of 1 fs, 16 bits as a digital
audio source is input. The interpolation filter 14 executes processing to multiply the sampling
frequency by 8 with respect to this input signal, converts it into a signal of a format of 8 fs, 24
bits, and outputs the converted signal to the synthesizer 12.
[0076]
In this case, in the synthesizer 12, an audio signal of a digital audio source in the form of 8 fs and
24 bits, and a first noise canceling audio output from the noise canceling signal processing unit 6
via the interpolation filter 7. The signal and the second noise cancellation audio signal output
from the noise cancellation signal processing unit 6A are input and synthesized. As an output of
the synthesizer 12, an audio signal obtained by further combining an audio signal as a digital
audio source, an audio signal as a digital audio source, and a composite noise canceling audio
signal formed by combining components of the first and second noise canceling audio signals. It
will be. The audio signal is first subjected to noise shaping by the noise shaper 8 to be a 16 fs, 4bit digital signal, and further to a PWM modulation by the PWM circuit 9 to be converted to a
512 fs, 1-bit digital signal . Then, the digital signal of this 1-bit string is input to the power drive
circuit 10 provided on the analog block 700 side, and converted into an amplified analog signal
here. The amplified analog signal is supplied to the driver 1a via the filter 11 and the capacitor
10-04-2019
30
C1 outside the LSI 100. Further, the input signal of the power drive circuit 10 is branched and
can be output to the outside (external 1-bit output).
[0077]
Here, the following can be said when the configuration of the noise canceling system according
to the present embodiment shown in FIG. 6 is compared with the configuration shown in FIG. In
the configuration of FIG. 6, the signal system for noise cancellation corresponding to FIG. 4 is
ΔΣ modulator 4 → (switch SW1) → decimation filter 5A → decimation filter 5B → DSP 60
(noise cancel signal processing unit 6) → interpolation A signal system is formed of filter 7 →
synthesizer 12 → noise shaper 8 → PWM circuit 9 → power drive circuit 10 → filter 11 →
capacitor C 1 → driver 1 a. This is a signal system for generating a first noise canceling audio
signal and outputting it as sound from the driver 1a. Then, in FIG. 6, a noise cancellation signal
processing unit 6A is provided. In other words, another signal system for noise cancellation is
provided in which a second noise cancellation audio signal is generated from the output signal of
the decimation filter 5A and output to the synthesizer 12. Thus, in the present embodiment, two
systems are provided as a system for forming a noise canceling audio signal based on a signal
obtained by collecting the sound by the microphone 2F. That is, in the system (first noise
cancellation signal processing system) including the noise cancellation signal processing unit 6 in
the DSP 60 to generate the first noise cancellation audio signal, the first noise cancellation signal
processing system is: Signals are propagated in the order of the decimation filter 5A, the
decimation filter 5B, the noise cancellation signal processing unit 6, the interpolation filter 7, and
the synthesizer 12. On the other hand, in a system (second noise cancellation signal processing
system) that includes the noise cancellation signal processing unit 6A and generates a second
noise cancellation audio signal, the decimation filter 5A, the noise cancellation signal processing
unit 6A, and the combination The signal is propagated in the order of the unit 12. That is, the
first noise cancellation signal processing system has the A / D conversion side decimation filter
(5A, 5B) and the D / A conversion side as in the configuration of the noise cancellation system
shown in FIG. While the signal passes through the sampling filter (7), in the second noise
cancellation signal processing system, although passing through the decimation filter 5A, it
passes through the decimation filter 5B in the subsequent stage, and further interpolates. It also
passes the relationship filter 6A, and passes only the noise cancellation signal processor 6A to
which a signal of sampling frequency = 8 fs is input / output.
Then, the signals obtained by the first and second noise cancellation signal processing systems
are synthesized by the synthesizer 12 to obtain a comprehensive noise cancellation audio signal.
And, this configuration is nothing but the configuration of the noise cancellation signal
processing system as the “dual path” described above with reference to FIGS. 5 (c) and 5 (d).
10-04-2019
31
[0078]
Here, in the dual path configuration of the present embodiment provided with the first and
second noise cancellation signal processing systems as described above, the respective model
configurations in FIG. Correspondingly, two basic embodiments can be taken based on the
function and role to be given to each of the first and second noise cancellation signal processing
systems. Therefore, the two functional mode examples will be described first.
[0079]
7 shows a decimation filter 5A, a decimation filter 5B, a noise cancellation signal processing unit
6A, a noise cancellation signal processing unit 6 in the DSP 60, an interpolation filter 7, and a
synthesizer in the noise canceling system shown in FIG. The part consisting of 12 is extracted
and shown. The first functional aspect example, which is one of the two functional aspect
examples, will be described with reference to this figure.
[0080]
First, as a first functional aspect example, as shown in FIG. 7, a first noise cancellation signal
processing system corresponding to the configuration of FIG. 4 is provided for a noise
cancellation signal processing unit that forms an audio signal for noise cancellation. The noise
cancellation signal processing unit 6 belonging to the above is treated as the main processing
unit, and the noise cancellation signal processing unit 6 belonging to one of the second noise
cancellation signal processing systems is treated as the sub processing system. That is, the
configuration corresponds to FIG. In this case, as described above, the digital filter of the noise
cancellation signal processing unit 6, which is the main processing unit, has an effective noise
cancellation effect among all audio frequency bands to be subjected to noise cancellation. It is
configured to execute noise cancellation signal processing corresponding to the frequency band
range below a certain level to be obtained. That is, since the first noise cancellation signal
processing system including the noise cancellation signal processing unit 6 has a signal delay by
including the decimation filter 5B and the interpolation filter 7, noise cancellation effective for a
high frequency band above a certain level is realized. Although it is difficult to expect an effect,
here, high frequencies above a certain level are excluded to generate an audio signal for noise
cancellation that covers a frequency band range as a lower middle or lower band. In addition, the
10-04-2019
32
digital filter of the noise cancellation signal processing unit 6A, which is a sub-processing unit, is
formed so as to generate a noise cancellation audio signal of a characteristic in which the noise
cancellation is performed for the high frequency band. Then, as a result of the noise cancellation
audio signals of the main processing unit and the sub-processing unit being synthesized by the
synthesizer 12, it is necessary as a noise cancellation target as a general noise cancellation audio
signal output from the synthesizer 12. The function of producing effective noise cancellation
effect over the entire voice frequency band to be In this manner, as the first functional aspect
example, as described above, first, the first noise cancellation signal processing system performs
noise cancellation for medium and low frequencies. In the first noise cancellation signal
processing system, the second noise cancellation signal processing system with a smaller signal
delay is configured to perform cancellation in an auxiliary manner for a high frequency band
where it is difficult to obtain a sufficient noise cancellation effect.
That is, the frequency band of noise to be canceled is shared by the first and second noise
cancellation signal processing systems (noise cancellation signal processing units 6A and 6). In
this case, as described above with reference to FIG. 5D, the noise cancellation signal processing
unit 6A is a simple gain adjustment circuit or a simple circuit such as a circuit for obtaining a
moving average value by an FIR filter of several steps. Can be realized by a simple hardware
configuration, and significant reductions in resources and circuit scale can be achieved. Further,
in the noise cancellation signal processing unit 6 in the DSP 60, too, in this case, it is not
necessary to configure the noise cancellation with an intention to effectively perform noise
cancellation for the high region, so the resources are reduced accordingly. It is also advantageous
in terms of processing capacity. Further, in this way, with the simpler configuration,
simplification of the design of the filters as the noise cancellation signal processing units 6 and
6A is expected.
[0081]
Next, with reference to FIG. 8, a second functional aspect example will be described. In this
figure, the same parts as in FIG. 7 will be assigned the same reference numerals and explanation
thereof will be omitted. As an example of the second functional aspect, contrary to the first
functional aspect described with reference to FIG. 7, the second noise cancellation signal
processing system is a main signal processing system, and the first noise cancellation signal
processing system is As a sub signal processing system. Accordingly, the noise cancellation signal
processing unit 6A belonging to the second noise cancellation signal processing system becomes
a main processing unit, and the noise cancellation signal processing unit 6 belonging to the first
noise cancellation signal processing system becomes a sub processing unit. That is, the
configuration corresponds to FIG. Then, as described in FIG. 5 (c) as the assignment of roles, the
10-04-2019
33
noise cancellation signal processing unit 6A, which is the main, performs noise cancellation for
the middle and high frequencies in all audio frequency bands to be subjected to noise
cancellation. The noise cancellation signal processing unit 6 is configured to generate a noise
cancellation signal for performing noise cancellation, and performs noise cancellation on the low
frequency band in all audio frequency bands to be subjected to noise cancellation. It is
configured to generate a noise cancellation signal. Also in this case, as a noise cancellation audio
signal obtained by combining the noise cancellation audio signals of the main processing unit
and the sub processing unit by the synthesizer 12, all audio frequencies required as noise
cancellation targets The function of producing effective noise cancellation effect over the band
will be provided.
[0082]
In addition, when actually configuring the noise canceling system based on the present
embodiment, the noise canceling system is required to adopt which of the first functional aspect
example and the second functional aspect example is to be adopted. The appropriate one may be
selected according to various conditions such as cost and specifications. As understood from the
above description of FIGS. 5 (c) and 5 (d), when priority is given to low cost and to make the
circuit scale as small as possible, the first functional aspect example is used. Adoption is more
advantageous. On the other hand, in the second function example in which the noise cancellation
signal processing unit 6A by hardware is configured to execute main signal processing, a higher
quality noise cancellation effect can be expected. Therefore, in the case where priority is given to
providing high-quality reproduced sound, it is appropriate to adopt the second functional aspect.
[0083]
Here, in digital block 800 of the noise canceling system of the present embodiment, the
configuration of a digital filter employed in a predetermined functional circuit unit related to a
signal processing system for noise cancellation will be described. For example, in the noise
canceling system previously shown in FIG. 4, the decimation filters 5 (5A, 5B) and the
interpolation filter 7 described above are configured by linear phase FIR. As described above, this
is based on the idea that, since the processing object is an audio signal, in general, it is necessary
not to cause phase distortion or the like according to the frequency. It is. The linear phase type
FIR causes a group delay between input and output, but the devices of conventional A / D
converters and D / A converters play back audio sources that the user is actively listening to
Because it was assumed to be used for (recording), there was no problem in particular. For
example, in the case of reproducing an audio source, even if the signal processing of the audio
10-04-2019
34
source is input to the signal processing device and then reproduced as a sound, even if a
corresponding delay due to the signal processing occurs, This is nothing other than what is
normally reproduced and output continuously, and therefore, when the user reproduces and
listens to the audio sound source, the delay in signal processing is regarded as a problem.
However, when trying to divert an existing device to a noise canceling system instead of
reproducing an audio source, the group delay of the device makes it impossible or difficult to
obtain a phase that can cancel out the external sound. It comes as a problem, and it comes up as
a problem.
[0084]
As the noise canceling system of the embodiment shown in FIG. 6, first, this problem is caused by
the second noise cancellation signal processing including the noise cancellation signal processing
unit 6A not passing through the decimation filter 5B and the interpolation filter 7 The solution is
by providing a system. However, if the signal delay noted in decimation filter 5 (5A, 5B) and
interpolation filter 7 is further shortened also on the first noise cancellation signal processing
system side, the noise cancellation effect There will be fewer inhibiting factors, and a better effect
can be expected.
[0085]
Therefore, in the present embodiment, first, as one example, the configuration of the decimation
filter 5B shown in FIG. 6 and the digital filter as the anti-imaging filter 7b in the interpolation
filter 7 as the minimum phase shift type FIR is taken. You will be taken. As the basis of the
minimum phase shift type FIR digital filter, the minimum phase can be obtained as a system of
the FIR type digital filter, and the peak value is set on the head side (closer to the input) for the
tap coefficient. It can form by doing.
[0086]
For example, when comparing the impulse response waveform as the characteristics of a linear
phase FIR digital filter configured with the same number of taps and a digital filter of the
minimum phase shift FIR, first, in the linear phase FIR, the input timing The peak is obtained at a
timing delayed by a certain fixed time. This indicates that the output in response to the input has
a delay (group delay) by a fixed time according to the number of taps (order). On the other hand,
10-04-2019
35
in the minimum phase shift type FIR, a peak can be obtained at a timing faster than the input
timing, for example, about several taps. That is, although the same FIR digital filter is used, the
minimum phase shift type FIR has a very short delay time (input / output delay) of the output in
response to the input as compared with the linear phase type FIR. Therefore, if the minimum
phase shift type FIR is adopted for the decimation filter 5B and the anti-imaging filter 7b in the
interpolation filter 7, the signal delay time here is greatly shortened, and the signal delay Factor
will be almost eliminated. As a result, it is expected that a better noise cancellation capability can
be obtained as the first noise cancellation signal processing system.
[0087]
As is well known, in the case of the minimum phase shift type FIR, phase distortion occurs
according to the frequency. Therefore, in the case of an audio signal, the possibility of the sound
quality deterioration due to this phase distortion is inevitable. This is the reason why the linear
phase FIR is used for the digital filter implemented in the A / D converter or D / A converter
compatible with audio signals. However, although the signal processing target in this case is an
audio signal, it is, for example, an external sound to be subjected to noise cancellation, and the
required reproduction fidelity is considerably low compared to an audio source or the like.
Furthermore, the sound component that is actually considered to have a large cancellation effect
is a low frequency band called so-called low frequency, and there is also a trade-off with the
characteristics of the device, etc. It is considered to be sufficient for practical use if it works.
From this point of view, for example, even if the above-described decimation filter 5B and the
anti-imaging filter 7b are formed by the minimum phase shift FIR, almost no sound quality
problems occur.
[0088]
Further, according to the above description, in the component part of the decimation filter 5 and
the interpolation filter 7, the decimation filter 5A and the oversampling circuit 7a are not set as
the minimum phase shift type FIR. That is, linear phase FIR is used for these parts. This is
because the factors of signal delay in the decimation filter 5 and the interpolation filter 7 are
dominant in the decimation filter 5B and the anti-imaging filter 7b, respectively. Therefore, even
if linear phase FIR is used for the decimation filter 5A and the oversampling circuit 7a, for
example, by emphasizing reproduction sound quality etc., the signal delay in the signal
processing system passing through the noise cancellation signal processing unit 6 is particularly
problematic It is a must-have.
10-04-2019
36
[0089]
In addition, if it is intended to reduce the signal delay between the input and the output as
described above, it is also reasonable to configure the decimation filter 5B and the anti-imaging
filter 7b with an Infinite Impulse Response (IIR) filter. It becomes a thing. The impulse response
waveform of the IIR filter also shows a characteristic that a peak can be obtained at a timing as
fast as, for example, several taps with respect to the input timing, that is, the input / output delay
is very short As in the case of the FIR configuration, the signal delay for the first noise
cancellation signal processing system can be made smaller than ever.
[0090]
The digital filter as the noise cancellation signal processing unit 6 in the DSP 60 in the first noise
cancellation signal processing system can be formed by a linear phase FIR filter or an IIR filter.
The linear phase FIR filter or IIR filter as the noise cancellation signal processing unit 6 is a
functional circuit realized by the DSP 60 executing an operation in accordance with
programming (instruction), for example. Note that, in the first function aspect example in which
the noise cancellation signal processing unit 6 is the main processing unit, even if it is the signal
processing function in the DSP 60 realized by programming, the resource can be reduced, etc. In
consideration of the point, it is preferable to configure the noise cancellation signal processing
unit 6 by an IIR filter.
[0091]
The digital filter as the noise cancellation signal processing unit 6A, which belongs to the other
second noise cancellation signal processing system, is implemented as dedicated hardware for
generating the noise cancellation signal. Ru. Then, the noise cancellation signal processing unit
6A is configured by a linear phase FIR or IIR filter. However, under the present circumstances,
according to the second functional aspect example, the second noise cancellation signal
processing system (noise cancellation signal processing unit 6A) is main, and the first noise
cancellation signal processing system (noise cancellation signal processing unit 6) is When
considering the sub configuration, as described above with reference to FIG. 5C, configuring the
noise cancellation signal processing unit 6A with an IIR filter suppresses the amount of resources
required. This is advantageous for the purpose of obtaining high-quality noise cancellation
effects.
10-04-2019
37
[0092]
In addition, in the case of taking the second functional aspect example, the characteristics of the
noise cancellation signal processing unit 6A configured as hardware are also variably set in the
characteristics within a certain degree of freedom. If it can be done, it can be said that it is
preferable that adaptive noise cancellation signal processing can be performed higher than the
case where the characteristic is variably set only by, for example, the noise cancellation signal
processing unit 6 on the DSP 60 side. Therefore, as a case where an IIR filter is adopted for the
noise cancellation signal processing unit 6A, for example, a configuration in which the filter
characteristics can be varied as follows can be considered.
[0093]
First, a plurality of second-order IIR filters are provided as digital filters that form the noise
cancellation signal processing unit 6A. Here, five IIR filters 65-1, 65-2, 65-3, 65-4, and 65-5 are
prepared as second-order IIR filters in consideration of the actual number of operation steps and
the like. In addition, depending on the characteristics required for the noise cancellation signal
processing unit 6A, the patterns of the connection modes of these IIR filters 65-1 to 65-5 can be
appropriately selected from those shown in FIGS. To be FIG. 9 shows a pattern in which IIR filters
65-1, 65-2, 65-3, 65-4, and 65-5 are connected in series. In this case, a signal is input from the
first stage IIR filter 65-1 connected in series, and a signal is output from the last stage IIR filter
65-5. In FIG. 10, a system in which four IIR filters 65-1, 65-2, 65-3, 65-4 are connected in series
and a pattern in which a system of only one remaining IIR filter 65-5 is provided in parallel are
shown. It is shown. The input signal is branched and input to each system, and the output of each
system is synthesized by the synthesizer 66 and then output from the noise cancellation signal
processing unit 6A. In FIG. 11, a system in which four IIR filters 65-1, 65-2, and 65-3 are
connected in series and a system in which the remaining two IIR filters 65-4 and 65-5 are
connected in series are provided in parallel. Pattern is shown. The input signal is branched and
input to each system, and the output of each system is synthesized by the synthesizer 66 and
then output from the noise cancellation signal processing unit 6A. In FIG. 12, a system in which
three IIR filters 65-1, 65-2, 65-3 are connected in series, a system consisting of only IIR filter 654, and a system consisting only of IIR filter 65-5 are The patterns connected in parallel are
shown. The input signal is branched and input to these three systems, and the output of each
system is synthesized by the synthesizer 66 and output from the noise cancellation signal
processing unit 6A. In FIG. 13, a system in which two IIR filters 65-1 and 65-2 are connected in
series, a system in which two IIR filters 65-3 and 65-4 are connected in series, and only one IIR
10-04-2019
38
filter 65-5 are used. The pattern which provided the system which consists of in parallel is
shown. The input signal is branched and input to each system, and the output of each system is
synthesized by the synthesizer 66 and then output from the noise cancellation signal processing
unit 6A. In FIG. 14, a system in which two IIR filters 65-1 and 65-2 are connected in series, a
system in which only IIR filter 65-3 is used, a system in which only IIR filter 65-4 is used, and a
system which is only IIR filter 65-5 A pattern in which a system is provided in parallel is shown.
The input signal is branched and input to each system, and the output of each system is
synthesized by the synthesizer 66 and then output from the noise cancellation signal processing
unit 6A. FIG. 15 shows a pattern in which five IIR filters 65-1, IIR filters 65-2, IIR filters 65-3, IIR
filters 65-4, and IIR filters 65-5 are provided in parallel. The input signal is branched and input to
each filter, and the output of each filter is synthesized by the synthesizer 66 and output from the
noise cancellation signal processing unit 6A. The configurations shown in FIGS. 9 to 15 can be
realized with fewer hardware resources by reusing one hardware resource (resource) in
accordance with the time axis using, for example, a method such as a sequencer. It is a thing.
[0094]
Also, it has been described earlier that the noise cancellation signal processing unit 6 in the DSP
60 is preferably configured with an IIR filter in the case of taking the first functional aspect
example, but in this way noise cancellation When the signal processing unit 6 is to be configured
as an IIR filter, the configuration described with reference to FIGS. 9 to 15 can be adopted by
programming the DSP 60.
[0095]
Therefore, the IIR filter 61 in the case where the pattern shown in FIG. 9 is adopted as the noise
cancellation signal processing unit 6 in the DSP 60 after adopting the first function aspect
example as the noise canceling system of the present embodiment. An example of setting of the
characteristics for each of the points -1 to 61-5 is shown in FIG.
In this case, first, the IIR filter 65-1 of the first stage is given a function as a gain setting circuit
which gives a gain to the input signal and outputs it. Here, 0.035 is set as the gain coefficient
(Gain). Also, for each of the IIR filters 65-2 to 65-5 from the second stage to the fifth stage (final
stage), a function called a so-called parametric equalizer is given. Then, as the equalizer
characteristics, for the IIR filter 65-2, the center frequency fc = 20 Hz, the Q value = 0.4, and the
gain value G = 28 dB are set, and for the IIR filter 65-3, the center frequency fc = 800 Hz, Q value
10-04-2019
39
= 0.6, gain value G = 12 dB, and for IIR filter 65-4, center frequency fc = 10000 Hz, Q value = 3.2,
gain value G = −21 dB, IIR filter 65 For -5, the center frequency fc = 18500 Hz, the Q value =
2.5, and the gain value G = -16 dB are set. Further, although not shown here, the noise
cancellation signal processing unit 6A is configured as a gain adjustment circuit corresponding to
the configuration of the noise cancellation signal processing unit 6 described above. Then, for
example, 0.012 is set for the gain coefficient.
[0096]
Here, a noise canceling system (a noise canceling system with a single pass configuration) having
a configuration (design) based on FIG. 4 and a noise canceling system according to the present
embodiment (a dual path configured (designed) with FIG. The results of comparing the
characteristics of the configuration with the noise canceling system) are shown in the Bode
diagram of FIG. As a Bode diagram, FIG. 21 (a) shows frequency vs. gain characteristics and
frequency vs. phase characteristics of the noise canceling system with a single pass configuration
based on FIG. 4; FIG. 21 (b) shows FIG. The frequency vs. gain characteristics and frequency vs.
phase characteristics of a noise canceling system with a dual path configuration based thereon
are shown. In order to obtain the characteristics shown in FIG. 21B, the minimum phase shift
type FIR is adopted for the digital filter as the decimation filter 5B and the anti-imaging filter 7b
in FIG. 6, and the noise cancellation signal processing unit 6A is IIR. Shall be composed of For
example, here, the target frequency vs. gain characteristics to be obtained for the noise canceling
system by the feedforward method are the characteristics shown by broken lines in the
frequency vs. gain characteristics in each of FIGS. 21 (a) and (b). Is assumed. The upper limit of
the frequency of the target characteristic shown by the broken line is up to 2 kHz because the
frequency band of the voice to be controlled for noise cancellation in practice is up to about 2
kHz. Further, in the frequency vs. gain characteristic shown in FIG. 21 (b), a constant or higher
gain is maintained up to around 100 kHz, while in the frequency vs. gain characteristic shown in
FIG. 21 (b), it is around 20 kHz. It is a characteristic that attenuates sharply. This is represented
by fs / 2 to avoid aliasing based on the sampling theorem, since the noise canceling system with
the configuration based on FIG. 4 performs noise cancellation processing on a signal with a
sampling frequency of only 1 fs. To remove a band higher than the sampling frequency.
Incidentally, in this case, fs = 44.1 kHz. Therefore, the frequency vs. gain characteristic shown in
FIG. 21B shows the result of attenuating the frequency band higher than 22.05 kHz.
[0097]
Here, for example, comparing FIG. 21 (a) with FIG. 21 (b), first of all, the frequency vs. gain
10-04-2019
40
characteristics are almost equal in the range of the frequency band up to about 2 KHz which is
actually a target of noise cancellation. It has become. However, with regard to the frequency vs.
phase characteristic, in the dual path configuration of FIG. 21 (b), almost 0 deg. When the same
range of about 2 kHz to 10 kHz is observed in the single pass configuration of FIG. In order to
cause the above phase rotation, it is greatly fluctuated. In this way, the noise canceling system
according to the present embodiment has a result that the phase rotation of the signal is
substantially reduced even in practice, which is sufficient for practical use although it is a digital
system. It is also easily possible in reality to obtain a noise canceling system.
[0098]
FIG. 17 shows a configuration example of a noise canceling system according to the second
embodiment. In this figure, the same parts as those in FIG. 6 corresponding to the first
embodiment described above are assigned the same reference numerals and explanation thereof
is omitted.
[0099]
As described above with reference to FIGS. 1 to 3, the noise canceling system of the headphone
device is roughly classified into a feed forward method and a feedback method. The first
embodiment described above is configured based on the feed forward method. The present
invention is applicable not only to the feed forward method but also to the feedback method.
Therefore, as a second embodiment, a configuration example of a noise canceling system based
on a feedback method whose model is shown in FIG. 1 is shown.
[0100]
In the case of the feedback method, as schematically shown in FIG. 17, the sound output from the
driver 1a can be collected near the headphone wearer's ear inside the headphone unit 1c as
shown schematically in FIG. It is arranged in the following position. The sound collected by the
microphone 2B at this position includes, in addition to the sound output from the driver, a
component of an external sound that, for example, penetrates the housing of the headphone
device and tries to be heard by the headphone device user. It is done. The sound signal collected
in this manner is amplified by the amplifier 3A to be an analog voice signal, and is further input
to the ΔΣ modulator 4A of the analog block 700 of the LSI 600, where 64fs digital audio by 1
10-04-2019
41
bit is used. It is converted into a signal and input to the decimation filter 5C of the decimation
filter 5-1 of the digital block 800 via the switch SW11. Also in this case, in consideration of
extensibility etc., a digital microphone input is provided in parallel with the microphone 2B, and a
digital audio signal from this digital microphone input and a microphone outputted from the Δ
変 調 modulator 4A. The digital audio signal from 2B can be selected by the switch SW11.
[0101]
The decimation filter 5-1 is a filter for decimating the 64 fs, 1-bit signal obtained by A / D
conversion to a sampling frequency suitable for signal processing in the digital block 800 in a
feedback type noise cancellation signal processing system. And corresponds to the decimation
filter 5 in correspondence to FIG. The decimation filters 5C and 5D which are supposed to form
the decimation filter 5-1 respectively correspond to the decimation filters 5A and 5B in
correspondence with FIG. The signal decimated to 8 fs by the decimation filter 5C is branched
and input to the noise cancellation signal processing unit 6B, and the signal decimated to 1 fs by
the decimation filter 5D is compared to the noise cancellation signal processing unit 6 in the DSP
60. Will be input. The noise cancellation signal processing unit 6B is provided in the second
noise cancellation signal processing system provided corresponding to the feedback method, and
in correspondence with FIG. 6, the noise cancellation signal processing unit 6A corresponds.
[0102]
In this case, the noise cancellation signal processing unit 6, 6B, for example, gives required
characteristics for the input signal, thereby reaching an ear of the driver 1a of the headphone
wearer as a noise cancellation audio signal and hearing it as an external voice To generate an
audio signal of sound having the property of being able to cancel. In general, this is processing
for providing a transfer function −β for noise cancellation to a collected voice signal. In
addition, with regard to these noise cancellation signal processing units 6 and 6B, the concept of
the first and second functional mode examples and the first and second functional mode
examples described earlier also in the first embodiment. The configuration according to can be
applied. The same format and configuration as those of the first embodiment can be applied to
the format and configuration of the digital filter as the noise cancellation signal processing unit 6
and 6B.
[0103]
10-04-2019
42
Further, regarding the feedback method, using the equalizer 61 in the DSP 60 in the first noise
cancellation signal processing system is effective to obtain a good noise cancellation effect. The
equalizer 61 in this case is designed to give characteristics of the transfer function of 1 + β to
the signal of the digital audio source. In the case of the feedback method, in the noise
cancellation audio signal output from the noise cancellation signal processing unit 6, not only the
component corresponding to the external sound but also the sound of the digital audio source
output as the sound from the driver 17a Ingredients are also included. That is, the characteristic
according to the transfer function represented by 1/1 + β is given to the sound component of
the digital audio source. Therefore, this equalizer 43 gives the signal of the digital audio source
in advance a characteristic based on a transfer function of 1 + β which is the reciprocal of 1/1 +
β. As a result, at the stage where the signal of the digital audio source from the interpolation
filter 14 is combined with the noise canceling audio signal in the combiner 12, the abovementioned 1/1 + β transfer characteristic is canceled. As a result, as a signal output from the
synthesizer 12, it is possible to obtain a signal in which a signal component having a
characteristic of canceling an external sound and a signal component of the original digital audio
source are synthesized.
[0104]
The configuration after the synthesizer 12 in this case is the same as that shown in FIG. That is,
the signal as the output of the synthesizer 12 is an audio signal amplified through the noise
shaper 8, the PWM circuit 9 and the power drive circuit 10, and this is further supplied to the
driver 1a through the filter 11 and the capacitor C1. As a result, the driver 1a is driven to output
a sound. In this manner, in the feedback method, the external sound component mixed with the
sound output from the driver near the ear of the headphone wearer is collected to generate a
signal for noise cancellation. Then, this signal for noise cancellation is output from the driver as
negative feedback is applied. As a result, with respect to the ear corresponding to the driver 17a
of the headphone device user, the external sound is canceled out and the sound in which the
sound of the digital audio source is relatively emphasized arrives and can be heard. Also in the
configuration of the noise canceling system corresponding to such a feedback method, the first
noise cancellation signal processing system passing through the noise cancellation signal
processing unit 6 of the DSP 60 is the same as in the first embodiment. In addition, by providing
the second noise cancellation signal processing system via the noise cancellation signal
processing unit 6A, it is possible to obtain the same effect as that of the first embodiment.
[0105]
10-04-2019
43
FIG. 18 shows a configuration example of a noise canceling system according to the third
embodiment. In this figure, the same parts as those in FIGS. 6 and 17 corresponding to the first
and second embodiments described above are designated by the same reference numerals, and
the description thereof will be omitted.
[0106]
The third embodiment adopts a configuration in which a feedforward system to which the first
embodiment corresponds and a feedback system to which the second embodiment corresponds
are used in combination. As described above, the feedback method and the feed forward method
have the characteristic of a relationship that is mutually traded off. For example, in the feed
forward method, the frequency band where noise can be effectively canceled (attenuated) is wide
and the stability of the system is high, but it is said that it is difficult to obtain a sufficient amount
of noise cancellation. For this reason, depending on conditions such as the positional relationship
with the noise source, for example, it may not match the transfer function of the system, and for
example, the noise may not be canceled or may increase in a specific frequency band. Is pointed
out. In this case, in spite of the fact that noise cancellation works effectively over a wide
frequency band, a phenomenon occurs in which noise is noticeable only in a specific frequency
band. , It becomes difficult to feel the noise cancellation effect. On the other hand, the feedback
method is characterized in that although the frequency band where noise can be canceled is
narrow, a sufficient amount of noise cancellation can be obtained. From this point of view, if the
noise canceling system is constructed by combining the feedback method with the feed forward
method, it is easy to effectively cancel the noise over the wide frequency band by compensating
each other's disadvantages. It becomes possible. That is, a better noise cancellation effect can be
expected than in the case where only one of the methods is used.
[0107]
Then, in the configuration as the third embodiment shown in FIG. 18, first, the microphone 2F,
the amplifier 3, the ΔΣ modulator 4, and the same as in FIG. 6 as corresponding to the feed
forward system. The switch SW1, the decimation filter 5 (decimation filters 5A, 5B), and the noise
cancellation signal processing unit 6A are shown. Also, as in the case of FIG. 17, the microphone
2B, the amplifier 3A, the ΔΣ modulator 4A, the switch SW11, the decimation filter 5-1
(decimation filter 5C, 5D), and the noise cancellation signal as corresponding to the feedback
type system The processing unit 6B is shown.
10-04-2019
44
[0108]
Further, the noise cancellation signal processing unit 6 of the DSP 60 in this case inputs the
signal from the decimation filter 5B corresponding to the feedforward system and the signal
from the decimation filter 5D corresponding to the feedback system. , And are shown as
generating and outputting an audio signal for noise cancellation. In practice, in the noise
cancellation signal processing unit 6 in this case, a filter for inputting a signal from the
decimation filter 5B to generate an audio signal for noise cancellation corresponding to the
feedforward method, and a signal from the decimation filter 5D are inputted And a filter for
generating a noise canceling audio signal corresponding to the feedback system. Then, for
example, in the noise cancellation signal processing unit 6, the noise cancellation audio signal
generated by these filters is synthesized and then output to the interpolation filter 7.
[0109]
Then, in the synthesizer 12 in this case, the noise cancellation audio signals from the noise
cancellation signal processing units 6A and 6B and the interpolation filter 7 and the digital audio
source signal from the interpolation filter 14 are It synthesize | combines and it is made to
output to the circuit (noise shaper 8) of a back | latter stage.
[0110]
Thus, in the noise canceling system of the third embodiment, the configuration of the first and
second noise cancellation signal processing systems as feedforward systems corresponding to
FIG. 6 and the feedback system corresponding to FIG. And the configuration of the first and
second noise cancellation signal processing systems.
With this configuration, as described above, a better noise cancellation effect can be obtained
than in the case where only one of the methods is used.
[0111]
FIG. 19 shows an example of the configuration of a noise canceling system according to the
fourth embodiment. The noise canceling system shown in this figure corresponds to the feed
10-04-2019
45
forward method, and the constituent parts are the same as those shown in FIG. In the first
embodiment shown in FIG. 6, the digital block 800 has been described as being manufactured as
one chip. However, focusing on the sampling frequency of the signal input / output in the
functional circuit unit in the digital block 800, this is not uniform, and it can be understood that
there are several types. As described above, when the corresponding sampling frequency is
different among the functional circuit units, the corresponding sampling frequency of the
functional circuit units included in the digital block 800 can be obtained in consideration of the
conditions for manufacturing the LSI in practice. In some cases, it may be more efficient to group
and group based on the above and group the functional circuits to be grouped into chips.
Therefore, in the present embodiment, a chip forming the digital block 800 is configured as
follows. First, considering the main sampling frequencies of signals handled in the digital block
800 shown in FIG. 19, one is 1 fs mainly based on the DSP 60 corresponding to the first noise
cancellation signal processing system. The other is 8 fs corresponding to the second noise
cancellation signal processing system.
[0112]
Therefore, in the present embodiment, as illustrated, first, the first signal processing chip 810 is
manufactured as one chip on which at least a circuit portion as the DSP 60 corresponding to 1 fs
is formed, and separately from this, At least each circuit portion as decimation filter 5 (5A, 5B),
noise cancellation signal processing unit 6A, interpolation filter 7, interpolation filter 14, and
synthesizer 12 is formed as a functional circuit unit corresponding to 8 fs. The second signal
processing chip 820 is manufactured as one chip.
[0113]
As for the functional circuit unit in the digital block 800 which is not included in any of the first
signal processing chip 810 and the second signal processing chip 820 in the figure, the first
signal processing chip 810 and the second signal may be appropriately selected. To be included
on the appropriate side of the processing chip 820, or to include other chips other than the first
signal processing chip 810 and the second signal processing chip 820. You may
[0114]
The configuration as the fourth embodiment shown in FIG. 19 is similarly applied to the digital
block 800 of the noise canceling system corresponding to the feedback method shown in FIG. 17
as the second embodiment. Is applicable.
10-04-2019
46
That is, the first signal processing chip 810 having at least a circuit portion as the DSP 60
corresponding to 1 fs, a decimation filter 5-1 (5C, 5D), a noise cancellation signal processing
portion 6B, and a functional circuit portion corresponding to 8 fs. A second signal processing
chip 820 having at least each circuit portion as the interpolation filter 7, the interpolation filter
14, and the synthesizer 12 is manufactured.
[0115]
Furthermore, the configuration as the fourth embodiment can also be applied to the digital block
800 of the noise canceling system using the feedforward method and the feedback method
shown in FIG. 18 in combination as the third embodiment.
This configuration is shown in FIG. 20 as a fifth embodiment. In FIG. 20, a first signal processing
chip 810 having at least a circuit portion as the DSP 60 corresponding to 1 fs and a decimation
filter 5, 5-1 (5A, 5B, 5C, 5D as a functional circuit portion corresponding to 8 fs). , Noise
cancellation signal processing units 6A and 6B, an interpolation filter 7, an interpolation filter 14,
and a second signal processing chip 820 at least forming respective circuit portions as the
synthesizer 12.
[0116]
The sampling frequency of the signal input / output in the functional circuit unit in the LSI 600
and the number of quantization bits described in the above embodiments are merely one of
typical ones, and the noise canceling system The sampling frequency to be handled by each
functional circuit unit and the number of quantization bits may be changed as necessary, as long
as no failure occurs in the formation of the system of.
[0117]
Also, in the embodiments described so far, there is shown an aspect as a dual path in which two
systems of the first noise cancellation signal processing system and the second noise cancellation
signal processing system are shown. An arrangement in which a plurality of second noise
cancellation signal processing systems are further provided, for example, is also conceivable
under the present invention.
10-04-2019
47
In such a configuration, for example, signals of different sampling frequencies are input to
generate a noise canceling audio signal for each of a plurality of series of the second noise
cancellation signal processing system, and the roles of the respective systems are shared. Is
considered. The configuration in which two or more second noise cancellation signal processing
systems are provided in this way is also referred to as "multipath". Here, in the case of a multipass configuration in which two or more second noise cancellation signal processing systems are
provided as described above, a model example of a signal processing system that is the basis of
the multi-pass configuration is shown in FIG. To be. FIG. 22 shows, as a model example, a
configuration in which a signal of sampling frequency = 64 fs is multi-passed and finally
synthesized and output in the same 64 fs format. In this figure, first, downsampling circuits 91-1
to 91-6, signal processing filters 92-0 to 92-6, upsampling circuits 94-1 to 94-6, and
synthesizers 93-0 to 93-5. Equipped with The downsampling circuits 91-1 to 91-6 respectively
downsample the sampling frequency of the input signal to 1⁄2 and output the result. The
downsampling circuits 91-1 to 91-6 are connected in series, and input an input signal of
sampling frequency = 64 fs to the downsampling circuit 91-1 of the first stage. As a result, the
downsampling circuits 91-1 to 91-6 output signals obtained by converting the sampling
frequency of the input signal into 32 fs, 16 fs, 8 fs, 4 fs, 2 fs, and 1 fs, respectively. The number
of quantization bits of a signal having a sampling frequency of 32 fs or less is set to be a multibit
by a predetermined number of bits. The signal processing blocks 92-0 to 92-6 are portions for
performing signal processing on input signals according to a predetermined purpose, and are, for
example, digital filters or the like to which predetermined signal characteristics are given. This
signal processing block corresponds to the noise cancellation signal processing unit 6A in each of
the paths when multipath conversion is performed. These signal processing blocks 92-0 to 92-6
respectively have an input signal of sampling frequency = 64 fs, and are outputted from the
down sampling circuits 91-1 to 91-6, sampling frequency = 32 fs, 16 fs, 8 fs, 4 fs , 2 fs and 1 fs
are input.
The signal processing blocks 92-0 to 92-6 respectively input these signals and output the signals
at the same sampling frequency (and the number of quantization bits) as the input. The upsampling circuits 94-1 to 94-6 up-sample the sampling frequency by two and output the signals
to be input. Signals of 32 fs, 16 fs, 8 fs, 4 fs, 2 fs, and 1 fs are input to the up-sampling circuits
94-1 to 94-5 from the synthesizers 93-1 to 93-5 described below. . A signal of 1 fs from the
signal processing filter 92-6 is input to the up-sampling circuit 94-6. The synthesizers 93-0 to
93-5 receive 64 fs, 32 fs, 16 fs, 8 fs, 4 fs, and 2 fs signals output from the signal processing
filters 92-0 to 92-5 respectively, and the up sampling circuit 94- Signals 64 fs, 32 fs, 16 fs, 8 fs,
4 fs, 2 fs, and 1 fs output from 1 to 94-6 are input, and these signals are synthesized. The
outputs of the combiners 93-1 to 93-5 are input to the up-sampling circuits 94-1 to 94-5. The
output of the synthesizer 93-0 is the final 64 fs output signal. Then, in order to realize
multipassing for the second noise cancellation signal processing system, it is necessary to obtain
a system with the required sampling frequency based on the configuration shown in FIG. 22
10-04-2019
48
above. After the down-sampling circuit, the up-sampling circuit, and the synthesizer are
implemented, the signal processing block (noise cancellation signal processing unit) is configured
to perform appropriate signal processing in each system.
[0118]
Also, as an embodiment, for the decimation filter 5B (5D) and the anti-imaging filter 7b in the
interpolation filter 7, by implementing the minimum phase type FIR or IIR filter, the phase
rotation can be suppressed more effectively. Although it is possible, as digital filters used in these
functional circuit units, the delay time is small enough to satisfy the required noise cancellation
effect, and other conditions such as sound quality, stability, etc. satisfy a certain level or more. As
long as it is, configurations other than the above-described minimum phase shift FIR and IIR
filters are also conceivable. In the present invention, the minimum phase shift type FIR or IIR
filter may be adopted as at least one of the decimation filter 5B (5D) and the anti-imaging filter
7b. . Even with such a configuration, for example, the delay of the signal processing system for
noise cancellation is smaller than in the case where both the above decimation filter 5B (5D) and
the anti-imaging filter 7b adopt the linear phase type. It will be shortened, so a corresponding
effect is expected.
[0119]
Also, it is how to actually mount each component for realizing the noise canceling system
according to the present embodiment on a device in practice, but in this regard, noise canceling
according to the present invention is actually implemented. It may be determined as appropriate
depending on the device to which the system is applied, the configuration of the system, the
application, and the like. For example, if it is intended to construct a headphone device having a
noise cancellation function alone, almost all components (that is, the LSI 600) that are supposed
to form a noise cancellation system should be accommodated in the housing of the headphone
device. Can be implemented. Alternatively, if the noise canceling system is to be configured by a
set of devices such as a headphone device and an external adapter, it is conceivable to mount the
LSI 600 on the adapter side. Furthermore, a configuration is also conceivable in which the
functional circuit unit in the LSI 600 is divided into a plurality of parts and at least one of them is
mounted on the adapter side. Furthermore, the noise canceling system according to the present
invention is not an headphone device but an audio reproducing device configured to reproduce
audio content and output it to a headphone terminal, a mobile phone device, a network voice
communication device, etc. Also in the case of mounting, it is conceivable to mount at least one of
components other than the microphone and the driver on these devices.
10-04-2019
49
[0120]
In addition, the present invention is configured to perform required digital signal processing
according to the same functional purpose by a plurality of signal processing systems respectively
corresponding to different sampling frequencies, and to expect some operational effects obtained
thereby It can be said that And from this point of view, as the present invention, the signal
processing means (procedure) for function correspondence is not limited to the function purpose
for noise cancellation, and performs signal processing corresponding to other function purpose.
May be configured.
[0121]
It is a figure showing an example model about a noise canceling system of a headphone device by
a feedback method. It is a Bode diagram showing the characteristic about the noise canceling
system shown in FIG. It is a figure showing an example model about a noise canceling system of a
headphone device by a feedforward method. It is a block diagram showing an example of basic
composition of a noise canceling system of a headphone device by a digital method. It is a figure
which shows the structure of the dual path | pass which the noise canceling system as this
Embodiment takes, compared with the structure of a single path | pass. BRIEF DESCRIPTION OF
THE DRAWINGS It is a block diagram which shows the structural example of the noise canceling
system as 1st Embodiment in this invention. As a first functional aspect example according to the
present embodiment, a frequency band setting example for a noise cancellation signal processing
unit of a first noise cancellation signal processing system and a noise cancellation signal
processing unit of a second noise cancellation signal processing system FIG. As a second
functional aspect example according to the present embodiment, a frequency band setting
example for a noise cancellation signal processing unit of a first noise cancellation signal
processing system and a noise cancellation signal processing unit of a second noise cancellation
signal processing system FIG. It is a figure which shows the example of a connection aspect in the
case of comprising with an IIR filter about the noise cancellation signal processing part of a 2nd
noise cancellation signal processing system. It is a figure which shows the example of a
connection aspect in the case of comprising with an IIR filter about the noise cancellation signal
processing part of a 2nd noise cancellation signal processing system. It is a figure which shows
the example of a connection aspect in the case of comprising with an IIR filter about the noise
cancellation signal processing part of a 2nd noise cancellation signal processing system. It is a
figure which shows the example of a connection aspect in the case of comprising with an IIR
filter about the noise cancellation signal processing part of a 2nd noise cancellation signal
10-04-2019
50
processing system. It is a figure which shows the example of a connection aspect in the case of
comprising with an IIR filter about the noise cancellation signal processing part of a 2nd noise
cancellation signal processing system. It is a figure which shows the example of a connection
aspect in the case of comprising with an IIR filter about the noise cancellation signal processing
part of a 2nd noise cancellation signal processing system. It is a figure which shows the example
of a connection aspect in the case of comprising with an IIR filter about the noise cancellation
signal processing part of a 2nd noise cancellation signal processing system. It is a figure which
shows the example of a characteristic setting about each IIR filter in the connection aspect shown
in FIG.
It is a block diagram showing an example of composition of a noise canceling system as a 2nd
embodiment in the present invention. It is a block diagram which shows the example of a
structure of the noise canceling system as 3rd Embodiment in this invention. It is a block
diagram which shows the example of a structure of the noise canceling system as 4th
Embodiment in this invention. It is a block diagram which shows the example of a structure of
the noise canceling system as 5th Embodiment in this invention. FIG. 7 is a Bode diagram
showing characteristics of a noise canceling system with a single pass configuration based on
FIG. 4 and a noise canceling system with a dual pass configuration based on FIG. 6. It is a block
diagram which shows the example of a model of the signal processing system which becomes the
basis of a multipath structure.
Explanation of sign
[0122]
DESCRIPTION OF SYMBOLS 1a driver, 1c headphone unit, 2F 2B microphone, 3 3A amplifier, 4
4A delta sigma modulator, decimation filter 5 5-1 (5A 5B 5C 5D), noise cancellation signal
processing unit 6 6A 6B, 7.14 interpolation filter, 7a oversampling circuit, 7b anti-imaging filter,
8 noise shaper, 9 PWM circuit, 10 power drive circuit, 11 filter, 12 synthesizer, 13 PCM
interface, 15 RAM, 16 flash memory, 65-1 to 65-5 IIR filter (secondary), 60 DSP, 61 equalizer, 62
acoustic analysis processing unit, SW1 and SW11 switch, 600 LSI, 700 analog block, 800 digital
block, 810 first signal processing chip 820 second signal processing chip
10-04-2019
51
Документ
Категория
Без категории
Просмотров
0
Размер файла
89 Кб
Теги
description, jp2008250270
1/--страниц
Пожаловаться на содержимое документа