JP5439707B2 - Signal processing apparatus and signal processing method - Google Patents

Description

  The present invention relates to a signal processing apparatus and a method for performing signal processing according to a predetermined purpose on an audio signal.

  A so-called noise canceling system for headphone devices that actively cancels external noise that is heard when playing sound of content such as music with a headphone device is known and put into practical use. It is becoming. And as such a noise canceling system, two systems, a feedback system and a feedforward system, are roughly classified.

For example, Patent Document 1 generates an audio signal obtained by inverting the phase of noise (noise) inside an acoustic tube collected by a microphone unit provided in the vicinity of the earphone unit in the acoustic tube attached to the user's ear, by outputting this as an earphone unit or rales, structure which is adapted to reduce external noise, that is, the configuration of the noise canceling system corresponding to the feedback scheme is described.
Further, in Patent Document 2, as a basic configuration, a configuration in which a sound signal obtained by collecting a microphone with a microphone attached to the outer casing of the headphone device is given a characteristic by a required transfer function and output from the headphone device, That is, the configuration of a noise canceling system corresponding to the feedforward method is described.

Japanese Patent Laid-Open No. 3-214892 Japanese Patent Laid-Open No. 3-96199

By the way, although it can be said about both the above-mentioned feedback system and feedforward system, what is currently put into practical use as a noise canceling system for a headphone device in a consumer device is configured by an analog circuit. It has become.
In order to effectively obtain the noise canceling effect of the noise canceling system, for example, the relationship between the external unnecessary sound collected by the microphone and the sound output from the driver for canceling the unnecessary sound is compared. It is necessary to keep the phase difference within a certain range. In other words, in the noise canceling system, a speed (response speed) from when an external unnecessary sound is input to when a canceling sound according to the input is output is required to be within a certain range.
However, if the noise canceling system is configured by a digital circuit, an A / D converter and a D / A converter are provided at the input and output. In the processing time of A / D converters and D / A converters that are widely used at present, when adopting as a noise canceling system, the delay is considerably large and it is difficult to obtain an effective noise canceling effect. For example, in the military and industrial fields, there are A / D converters and D / A converters that have a considerably high sampling frequency and a small delay, but these are extremely expensive and are not used in consumer equipment. Not realistic. This is the reason why the noise canceling system is configured by an analog circuit instead of a digital circuit.

However, replacing analog circuits with digital circuits facilitates changing and switching characteristics and operating modes without changing or replacing physical component element constants. In addition, there are many advantages that a sound-related system such as a noise canceling system can be expected to further improve sound quality.
Accordingly, an object of the present invention is to provide a practically sufficient noise canceling effect while being formed by a digital circuit, for example, as a noise canceling system for a headphone device in a consumer.

In view of the above-described problems, the present invention is configured as a signal processing apparatus as follows.
That is, a digital signal of the first format having a predetermined quantization bit number of 1 bit or more obtained by ΔΣ modulation processing is input, and expressed as n × fs (n is a natural number) with a predetermined reference sampling frequency as fs. First decimation processing means for generating and outputting a digital signal of the second format that is a pulse code modulated signal with a sampling frequency of the second, and a digital signal of the second format that is output from the first decimation processing means To generate and output a digital signal of the third format having a format as a pulse code modulation signal with a sampling frequency represented by m × fs (m is a natural number and m <n) Decimation processing means and a third format digital signal output from the second decimation processing means, and a predetermined signal corresponding to a predetermined functional purpose. A first function-corresponding signal processing means configured to execute the process and output in the same third format; and a third format signal output from the first function-corresponding signal processing means; An interpolation processing means that is converted into a format of the output and a second format digital signal output from the first decimation processing means and a predetermined signal corresponding to the functional purpose A second function-corresponding signal processing means configured to execute processing and output in the same second format; and at least a second format digital signal output from the second function-corresponding signal processing means; And a synthesizing means for synthesizing at least a digital signal of the second format output from the interpolation processing means and outputting it to the input stage for the subsequent digital-analog conversion process. The first function-compatible signal processing means includes a predetermined frequency band corresponding to the functional purpose, a predetermined frequency band that is a low frequency range, or a predetermined frequency band that is a low frequency range. A signal processing for giving a cancellation signal characteristic for canceling the cancellation target sound is performed, and the first function-corresponding signal processing means is set to a mid-range and low-frequency range below a predetermined frequency band. When the filter characteristic is set so as to give a signal characteristic for canceling the cancellation target sound component, the second function corresponding signal processing means is higher than the middle range and the low range. canceled target sound components of the frequency band set the filter characteristics to provide a signal characteristic of the order to be canceled is, serial second function corresponding signal processing means, the straight line A digital filter of a phase type finite impulse response system, or the first function corresponding signal processing means for canceling a cancellation target sound component in a frequency band below a predetermined frequency which is a low frequency range When the filter characteristic is set so as to give the cancellation signal characteristic, the second function corresponding signal processing means cancels the cancel target sound component in the frequency band higher than the low frequency band. The filter characteristic is set so as to give the signal characteristic to achieve the above, and the second function corresponding signal processing means is constituted by a digital filter of an infinite impulse response system.

  In the above configuration, for a digital signal processing system of a system that cancels (reduces or attenuates) a predetermined cancellation target sound, a signal characteristic (cancellation signal characteristic) for canceling the predetermined cancellation target sound to a signal having a sampling frequency of m × fs. ) And a system that provides a cancellation signal characteristic to a signal having a higher sampling frequency n × fs. In addition, in the latter system that gives the cancel signal characteristic to the signal of the 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. 2 is synthesized with a signal of sampling frequency n × fs given the cancel signal characteristic of the former. Thus, by omitting the decimation process and the interpolation process, the signal delay of the latter system is greatly reduced.

As described above, a system that provides a plurality of cancel signal characteristics is provided, and a signal delay (signal propagation time) in at least one of the systems is shortened. The response speed required for the signal processing system of the noise canceling system can be satisfied. That is, it becomes possible to easily realize a noise canceling system using a digital circuit method. The realization of a noise-cancelling system using digital circuits will enable the implementation of functions that have been difficult with analog circuits, and improve sound quality. Value increases.
Further, as in the present invention, by providing a plurality of systems that give cancel signal characteristics, for example, it is possible to assign a specific signal processing function to each system and share it, and noise canceling System performance and design flexibility will be improved.

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 is taken as an example.
Therefore, prior to describing the configuration of the present embodiment, the basic concept of a noise canceling system corresponding to a headphone device will be described.

  As a basic method of such a noise canceling system compatible with a headphone device, a servo control by a feedback method and a feedforward method are respectively known. First, the feedback system will be described with reference to FIG.

FIG. 1A schematically shows a model example of a noise canceling system using a feedback method on the right ear (R channel in two-channel stereo by L (left) and R (right)) of a headphone wearer (user). Is shown.
As a structure on the R channel side of the headphone device here, first, a driver 202 is provided in a position corresponding to the right ear of the user 500 wearing the headphone device in the housing portion 201 corresponding to the right ear. . The driver 202 is synonymous with a so-called speaker, and is driven (driven) by an amplified output of an audio signal so as to output the sound so as to be released into the space.

In addition, as a feedback method, the microphone 203 is provided in a position near the right ear of the user 500 in the housing unit 201. Depending on the microphone 203 provided in this way, the sound output from the driver 202 and the sound that enters the housing part 201 from the external noise source 301 and reaches the right ear, that is, the right ear can be heard. In-housing noise 302, which is an external audio signal, is collected. Note that the noise 302 in the housing is generated because the noise sound source 301 leaks as a sound pressure from a gap such as an ear pad of the housing, or the headphone device casing vibrates due to the sound pressure of the noise sound source 301. It can be mentioned that this is transmitted into the housing part.
Then, in order to cancel (attenuate or reduce) the in-housing noise 302 such as a signal having a reverse characteristic with respect to the audio signal component of the external audio from the audio signal obtained by collecting the sound with the microphone 203. Signal (cancellation audio signal) is generated, and this signal is fed back to be synthesized with a sound signal (audio sound source) of a necessary sound for driving the driver 202. As a result, at the noise cancellation point 400 set at a position corresponding to the right ear in the housing portion 201, the external sound is canceled by synthesizing the output sound from the driver 202 and the component of the external sound. A sound is obtained, and this sound is heard by the user's right ear. By providing such a configuration also on the L channel (left ear) side, a noise canceling system as a headphone device corresponding to normal L, R2 channel stereo can be obtained.

The block diagram in FIG. 1B shows a basic model configuration example of a noise canceling system using a feedback method. FIG. 1B shows a configuration corresponding only to the R channel (right ear) side as in FIG. 1A, and the L channel (left). A similar system configuration can be provided for the (ear) side. The block shown in this figure indicates one specific transfer function corresponding to a specific circuit part, circuit system, etc. in the feedback canceling noise canceling system. Here, the block is referred to as a transfer function block. To do. The character shown in each transfer function block represents the transfer function of the transfer function block, and each time a voice signal (or voice) passes through the transfer function block, the transfer function shown there Will 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 a sound signal. It is obtained as an audio signal via 101 (transfer function M). The audio signal that has passed through the transfer function block 101 is input to the synthesizer 103 via the transfer function block 102 (transfer function −β) corresponding to the FB (FeedBack) filter circuit. The FB filter circuit is a filter circuit in which a characteristic for generating the above-described cancellation audio signal is set from the audio signal obtained by collecting the sound with the microphone 203, and its transfer function is expressed as -β. It is what.

In addition, the audio signal S of the audio sound source that is the content such as music is assumed to be equalized by an equalizer here, and a synthesizer is connected via a transfer function block 107 (transfer function E) corresponding to the equalizer. 103 is input.

  The synthesizer 103 here synthesizes the above two signals by addition. The synthesized audio signal is amplified by the power amplifier and output to the driver 202 as a drive signal, so that the driver 202 outputs the audio signal. That is, the audio 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 audio. Released into the space. Note that the transfer function D of the driver 202 is determined by the structure of the driver 202, for example.

  Then, the sound output from 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. In this space, it is combined with the noise 302 in the housing. As the sound pressure P of the output sound that reaches the right ear, for example, from the noise cancellation point 400, the sound of the noise sound source 301 that enters from the outside of the housing portion 201 is canceled.

のようにして表されるものとなる。この（数１）の式において、ハウジング内ノイズ３０２であるＮに着目すると、Ｎは、１ ／（１ ＋ＡＤＨＭβ）で表される係数により減衰されることがわかる。ただし、（数１）に示される式の系が、ノイズ低減対象の周波数帯域にて発振することなく、安定して動作するためには、

が成立していることが必要となる。 In the model of the noise canceling system shown in FIG. 1B, 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 sound source. Using the transfer functions M, -β, E, A, D, and H shown in the transfer function block,

It will be expressed as follows. In the equation (Equation 1), when attention is paid to N which is 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 frequency band targeted for noise reduction,

Must be established.

As a general rule, the absolute value of the product of each transfer function in a feedback-type noise canceling system is
1 << | ADHMβ |
In combination with Nyquist's stability discrimination in classical control theory, (Equation 2) can be interpreted as follows.
Here, in the system of the noise canceling system shown in FIG. 1B, a system represented by (−ADHMβ) obtained by cutting a loop portion related to N that is the noise 302 in the housing is considered. . This system is called “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 to be disconnected, the above open loop can be formed.

The above open loop has the characteristics shown by the Bode diagram of FIG. 2, for example. In this Bode diagram, the horizontal axis represents frequency, and the vertical axis represents gain in the lower half and phase in the upper half.
When this open loop is targeted, the following two conditions must be satisfied in order to satisfy (Equation 2) based on the stability determination of Nyquist.
Condition 1: Phase 0 deg. When passing through the (0 degree) point, the gain must be less than 0 dB.
Condition 2: When the gain is 0 dB or more, the phase is 0 deg. Do not include the point.

If the above two conditions 1 and 2 are not satisfied, positive feedback is applied to the loop, which causes oscillation (howling). In FIG. 2, gain margins Ga and Gb corresponding to the above condition 1 and phase margins Pa and Pb corresponding to the condition 2 are shown. If these margins are small, the possibility of oscillation increases due to various individual differences of the user who uses 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. 2, the phase 0 deg. The gain when passing through the point is smaller than 0 db, and gain margins Ga and Gb are obtained accordingly. However, for example, if phase 0 deg. The gain when passing through the point is 0 dB or more and the gain margins Ga and Gb are eliminated, or the phase is 0 deg. Although the gain when passing through this point is less than 0 dB, when the gain margins Ga and Gb are close to 0 dB and the gain margins Ga and Gb become small, oscillation occurs or the possibility of oscillation increases.
Similarly, in FIG. 2, when the gain is 0 dB or more, the phase is 0 deg. The phase margins Pa and Pb are obtained. However, for example, when the gain is 0 dB or more, the phase 0 deg. You've passed the point of, or, phase 0deg. If the phase margins Pa and Pb become smaller and the phase margins become smaller, oscillation occurs or the possibility of oscillation increases.

Next, in the configuration of the feedback type noise canceling system shown in FIG. 1B, in addition to the external sound (noise) canceling (reducing) function described above, a necessary sound (necessary sound) is generated by the headphone device. A case of reproduction output will be described.
Here, as a necessary sound, for example, an audio signal S of an audio sound source as content such as music is shown.
Note that the audio signal S may be considered in addition to such musical or similar contents. For example, when the noise canceling system is applied to a hearing aid or the like, a microphone provided outside the housing for picking up surrounding necessary sounds (different from the microphone 203 provided in the noise cancellation system) The sound signal obtained by collecting the sound. In addition, when applied to what is called a so-called headset, it becomes an audio signal such as a speech of the other party 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 use of the headphone device.

により表される式による特性を有するものとして設定したこととする。なお、この伝達特性Ｅは、周波数軸でみた場合に、上記オープンループに対してほぼ逆特性（１＋オープンループ特性）となっている。そして、この（数３）により示される伝達関数Ｅの式を、数１に代入すると、図１（ｂ）に示されるノイズキャンセリングシステムのモデルにおける出力音の音圧Ｐについては、

のようにして表すことができる。
（数４）におけるＡＤＨＳの項において示される伝達関数Ａ、Ｄ、Ｈのうち、先ず伝達関数Ａはパワーアンプに対応し、伝達関数Ｄはドライバ２０２に対応し、伝達関数Ｈはドライバ２０２からノイズキャンセル点４００までの経路の空間伝達関数に対応するので、ハウジング部２０１内のマイクロフォン２０３の位置が耳に対して近接した位置にあるとすれば、音声信号Ｓについては、ノイズキャンセル機能を有さないようにした通常のヘッドフォンと同等の特性が得られることがわかる。 First, in (Equation 1), attention is focused on the audio signal S of the audio source. And as a transfer function E corresponding to the equalizer,

It is assumed that it has been set as having characteristics according to the equation represented by The transfer characteristic E is almost opposite to the open loop (1 + open loop characteristic) when viewed on the frequency axis. Then, when the expression of the transfer function E represented by (Equation 3) is substituted into Equation 1, the sound pressure P of the output sound in the model of the noise canceling system shown in FIG.

It can be expressed as follows.
Of the transfer functions A, D, and H shown in the ADHS term in (Equation 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 spatial transfer function of the path up to the cancellation point 400, if the position of the microphone 203 in the housing portion 201 is close to the ear, the audio signal S has a noise canceling function. It can be seen that the same characteristics as those of normal headphones that are not used can be obtained.

Next, a noise canceling system using a feedforward method will be described.
FIG. 3A shows a configuration on the side corresponding to the R channel, as in FIG. 1A, as a model example of a noise canceling system using the feedforward method.
In the feed-forward method, the microphone 203 is provided outside the housing unit 201 so as to be able to pick up sound that is supposed to arrive from the noise sound source 301. Then, the external sound collected by the microphone 203, that is, the sound that is supposed to have arrived from the noise sound source 301 is collected to obtain a sound signal, and an appropriate filtering process is performed on the sound signal to cancel the audio signal. To be generated. Then, the canceling audio signal is synthesized with the necessary sound signal. That is, the canceling audio signal that electrically simulates the acoustic characteristics from the position of the microphone 203 to the position of the driver 202 is synthesized with the sound signal of the necessary sound.
Then, the sound signal obtained by synthesizing the canceling audio signal and the necessary sound signal is output from the driver 202 in this way. You will be able to hear the sound that has intruded in the canceled sound.

FIG. 3B shows a configuration on 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 part 201 is obtained as an audio signal through the transfer function block 101 having the transfer function M corresponding to the microphone 203 and the microphone amplifier.
Next, the audio signal that has passed through the transfer function block 101 is input to the synthesizer 103 via the transfer function block 102 (transfer function −α) corresponding to an FF (Feed Forward) filter circuit. The FF filter circuit is a filter circuit in which the characteristics for generating the above-described cancellation audio signal are set from the audio signal obtained by collecting the sound with the microphone 203, and its transfer function is expressed as -α. It is what.

In addition, 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 a power amplifier and output as a drive signal to the driver 202, so that the audio signal is output from the driver 202. 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 passes through the transfer function block 105 (transfer function D) corresponding to the driver 202. It is emitted into the space as sound.
Then, the sound output from 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. Where the noise is combined with the noise 302 in the housing.

In addition, a path from the noise source 301 to the noise cancellation point 400 until the sound emitted from the noise source 301 enters the housing portion 201 and reaches the noise cancellation point 400, as shown as the transfer function block 110. Is given a transfer function (spatial transfer function F). On the other hand, the microphone 203 picks up sound that is supposed to arrive from the noise sound source 301 that is external sound. At this time, the sound (noise) emitted from the noise sound source 301 is picked up. Until reaching, a transfer function (spatial transfer function G) corresponding to the path from the noise source 301 to the microphone 203 is given as shown as the transfer function block 111. As the FF filter circuit corresponding to the transfer function block 102, the transfer function −α is set in consideration of the above-described spatial transfer functions F and G.
Thereby, as the sound pressure P of the output sound that reaches the right ear from the noise cancellation point 400, for example, the sound of the noise sound source 301 entering from the outside of the housing portion 201 is canceled.

のようにして表されるものとなる。また、理想的には、ノイズ音源３０１からキャンセルポイント４００までの経路の伝達関数Ｆは、

のようにして表すことができる。
次に、（数６）に示される式を、（数５）に代入すると、右辺の第１項と第２項とが相殺されることとなる。この結果から、出力音の音圧Ｐは、

のようにして表すことができる。このようにして、ノイズ音源３０１から到達してくるとされる音はキャンセルされ、オーディオ音源の音声信号だけが音声として得られることが示される。つまり、理論上、ユーザの右耳においては、ノイズがキャンセルされた音声が聴こえることになる。ただし、現実には、（数６）が完全に成立するような伝達関数を与えることのできる、完全なＦＦフィルタ回路の構成は困難である。また、人による耳の形状であるとか、ヘッドフォン装置の装着の仕方についての個人差が比較的大きく、ノイズの発生位置とマイク位置との関係の変化などは、特に中高域の周波数帯域についてのノイズ低減効果に影響を与えることが知られている。このために、中高域に関しては、アクティブなノイズ低減処理を控え、主として、ヘッドフォン装置の筐体の構造などに依存したパッシブな遮音をすることがしばしば行われる。
また、確認のために述べておくと、（数６）は、ノイズ音源３０１から耳までの経路の伝達関数を、伝達関数−αを含めた電気回路にて模倣することを意味している。
In the noise canceling system model of the feedforward method shown in FIG. 3B, the sound pressure P of the output sound is N for noise generated in the noise sound source 301 and the sound signal of the audio sound source. S, and using the transfer function, M, -α, A , D, F, G, H shown in each transfer function block,

It will be expressed as follows. Ideally, the transfer function F of the path from the noise source 301 to the cancellation point 400 is

It can be expressed as follows.
Next, substituting the equation shown in (Equation 6) into (Equation 5) cancels out the first and second terms on the right side. From this result, the sound pressure P of the output sound is

It can be expressed as follows. In this way, it is indicated that the sound that is supposed to arrive from the noise sound source 301 is canceled and only the sound signal of the audio sound source is obtained as sound. That is, theoretically, the user's right ear can hear a noise-cancelled voice. However, in reality, it is difficult to construct a complete FF filter circuit that can provide a transfer function that fully satisfies (Equation 6). In addition, there are relatively large individual differences in the shape of the ears of a person and the manner in which the headphone device is worn. It is known to affect the reduction effect. For this reason, with regard to the mid-high range, active noise reduction processing is refrained, and passive sound insulation mainly depending on the structure of the housing of the headphone device is often performed.
For confirmation, (Expression 6) means that the transfer function of the path from the noise source 301 to the ear is imitated by an electric circuit including the transfer function -α.

Further, in the feedforward type noise canceling system shown in FIG. 3A, the microphone 203 is provided outside the housing, and therefore the feedback point type noise canceling system shown in FIG. Unlike the case, it can be arbitrarily set in the housing portion 201 so as to correspond to the ear position of the listener. However, normally, the transfer function -α is fixed, and at the design stage, it is a decision for some target characteristic. On the other hand, the shape of the ear is different depending on the listener. For this reason, there may be a phenomenon that a sufficient noise canceling effect cannot be obtained, or noise components are added in a non-reverse phase to generate abnormal noise.
For this reason, in general, the feedforward method has low possibility of oscillation and high stability, but it is difficult to obtain a sufficient noise attenuation amount (cancellation amount). On the other hand, the feedback method is said to require attention to the stability of the system instead of expecting a large amount of noise attenuation. Thus, the feedback method and the feedforward method have their characteristics.

  By the way, as a current situation, a noise canceling system for a headphone device which is actually in practical use and is in practical use is an analog system employing an analog circuit. However, if the digital system that uses the signal processing system as a digital signal processing system for the noise canceling system, it is easy to give various functions such as variable characteristics and switching of the operation mode of the noise canceling system. In addition, it is possible to improve the sound quality. Thus, the merit of digitalizing the noise canceling system is great.

Therefore, FIG. 4 shows an example of a configuration that can be properly considered when a noise canceling system for a headphone device is constructed using a digital device known at present.
The noise canceling system shown in this figure is configured based on the feedforward method shown in FIG.
The headphone device (hereinafter simply referred to as a headphone) 1 shown here corresponds to a two-channel stereo system using L (left) and R (right). This corresponds to either the channel or the 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 that should be listened to is omitted, and only the system for canceling the external sound (noise sound source) is shown. .

In FIG. 4, the microphone 2F is for collecting external sound including external sound (external noise) around the headphones 1 to be canceled. In the case of the feedforward method, the microphone 2F is generally provided outside the housing (headphone unit) corresponding to each of the L and R channels on the headphone 1 in practice. . In this figure, a microphone 2F is shown that is provided in the headphone unit 1c corresponding to one of the channels L and R among the headphone units 1c and 1d.
A signal obtained by picking up external sound by the microphone 2F is amplified by the amplifier 3 and 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 to be originally listened to by the headphones 1. As a specific example of the digital audio source here, a digital audio signal recorded on a CD (compact disc) or the like can be given such that fs = 44.1 kHz and quantization bit number = 16 bits. . Of course, other digital audio source formats such as those with fs = 48 kHz may be adopted.

The A / D converter 50 in this case is, for example, one component or device, and an input analog signal is converted into a digital signal in a PCM (Pulse Code Modulation) signal format with a predetermined sampling frequency and the number of quantization bits. Convert to signal and output. For this purpose, for example, as shown in the figure, a ΔΣ modulator 4 and a decimation filter 5 are provided.
The ΔΣ (delta sigma) modulator 4 converts the input analog audio signal into a 1-bit digital signal having a sampling frequency of 64 fs, for example. For this digital signal, the sampling frequency is lowered to, for example, 1 fs by the decimation filter 5 and the number of quantization bits is a predetermined multi-bit (16 bits here) corresponding to the digital audio source. And is output as a digital signal from the A / D converter 50.
In such a device as the A / D converter 50, the decimation filter 5 is generally formed by a linear phase type FIR (Finite Impulse Response) system (linear phase type FIR) having a linear phase characteristic. doing.
The digital signal to be processed in this noise canceling system is an audio signal. Therefore, if faithful sound reproduction is assumed, it is ideally desired that waveform distortion does not occur. If linear phase characteristics are given by FIR, the above waveform distortion does not occur. Further, if it is an FIR system, an accurate linear phase characteristic can be easily obtained as is well known. For this reason, the digital filter as the decimation filter 5 is constituted by a linear phase type FIR.
In order to make the digital filter of the FIR system a linear phase type, as is well known, for example, for tap coefficients, the coefficient peak value is set at the center of the number of taps (order) so as to be symmetrical. It can be realized by setting.

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 for executing a required signal processing for generating an audio signal of a sound to be output from at least the driver 1a of the headphone 1 by digital signal processing, and gives a necessary function by programming. It has been made possible. As will be understood from the following description, the audio signal to be output from the driver 1a of the headphone 1 is an audio signal to be heard so that the audio signal of the digital audio source and the external sound collected by the microphone 2F are canceled. The signal (cancellation audio signal) is synthesized.
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), quantization bit number = 16 bits). It is designed to perform digital signal processing. The PCM signal format supported by the DSP is set on the assumption that it is adapted to the format of a digital audio source synthesized with the noise canceling audio signal in the noise canceling system.

In this figure, a noise cancellation signal processing unit 6 is shown as a signal processing function block implemented in the DSP 60. The noise cancellation signal processing unit 6 is configured by a digital filter that inputs and outputs data corresponding to the 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, an audio signal having a function of canceling external sound that reaches the ear of the headphone wearer corresponding to the driver 1a and is heard as sound to be output from the driver 1a (cancellation audio) Signal). As the simplest audio signal for such cancellation, for example, an audio signal input to the noise cancellation signal processing unit 6, that is, an audio signal obtained by collecting an external sound has an inverse characteristic, This is a phase signal. In addition, in practice, a characteristic (corresponding to the transfer characteristic -α in FIG. 3) is given in consideration of transfer characteristics such as circuits and spaces in the noise canceling system.

In this case, the digital signal from the noise cancellation signal processing unit 6 that is output from the DSP 60 is synthesized by a synthesizer 12 with a digital audio source signal in the PCM signal format with sampling frequency = 1 fs and quantization bit number = 16 bits. Then, it is input to the D / A converter 70.
The D / A converter 70 is also a chip component, for example, and inputs a 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 figure, an interpolation filter 7, a noise shaper 8, a PWM circuit 9, and a power drive circuit 10 are provided.

The digital signal input to the D / A converter 70 is first input to the interpolation filter 7. The interpolation (oversampling) filter 7 converts the input digital signal so that the sampling frequency is raised to a sampling frequency obtained by multiplying the sampling frequency by a factor expressed by a power of 2 and outputs it. . In this case, it is assumed that the sampling frequency is increased to 8 fs. In this case, the number of quantization bits of the output signal is converted so that the number of bits is a multi-bit number 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 previous decimation filter 5.

  The digital signal output from the interpolation filter 7 is subjected to a process called noise shaping by the noise shaper 8. The signal after the noise shaping is, for example, a predetermined quantization smaller than that at the time of input at a sampling frequency (16 fs here) obtained by multiplying the sampling frequency at the time of input by a coefficient expressed by a power of 2 Converted to a bit number format. As is well known, noise shaping is obtained as a result of the ΔΣ modulation process, and therefore the noise shaper 8 can be realized by a ΔΣ modulator. That is, the digital noise canceling system shown in this figure adopts a configuration in which ΔΣ modulation is applied to A / D conversion and D / A conversion.

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 at the subsequent stage. The power drive circuit 10 is formed by, for example, a switching drive circuit that switches and amplifies a 1-bit string signal at a 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 circuit 10 is 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 insulation capacitor C1 after a predetermined unnecessary band component is removed by the filter 11, for example. .

  The sound output from the driver 1a driven in this way is a combination of the sound component of the digital audio source and the sound component of the noise cancellation audio signal. Depending on the component, there is an effect of canceling (cancelling) the external sound that reaches the ear corresponding to the driver 1a from the outside. As a result, as the sound that the headphone wearer listens to with the ear corresponding to the driver 1a, ideally, the external sound is canceled and the sound of the digital audio source is relatively emphasized.

  The configuration shown in FIG. 4 utilizes, for example, an A / D converter, DSP, D / A converter, etc. that are easily available for consumer use. Currently, as a noise canceling system using a digital method, for example, When an attempt is made to make an audio source such as a CD, this is a structure that can be considered in order.

However, it has been found that it is difficult to obtain a sufficient noise canceling effect in the above configuration. This is because the signal processing time (propagation time) of the actual devices as the A / D converter 50 and the D / A converter 70, that is, the delay between input and output is considerably large.
Originally, these devices are supposed to process a single audio signal as an audio source such as a normal music piece. Therefore, even if a delay is caused by signal processing, this will not be a problem. It is a thing. However, when such a device is used as it is for a noise canceling ring system, the delay becomes so large that it cannot be ignored.
In other words, the entire noise canceling system configured using these devices has a large delay in the time (response speed) from when external sound is picked up by the microphone 2F until it is output as sound by the driver. Will occur. Due to this delay, for example, it becomes difficult to cancel the external sound by 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 under the sampling frequency of 44.1 kHz is 40 samples, the phase rotation of a signal of about 550 Hz or more becomes 180 ° or more. If the delay is increased to such a level, not only is it difficult to obtain a noise canceling effect, but there may be a phenomenon in which external sound is emphasized.
As described above, in the configuration of the digital noise canceling system illustrated in FIG. 4 , the allowable noise canceling effect is limited to a frequency band range lower than about 550 Hz. As compared with the case where the standard 20 Hz to 20 kHz is set, the noise canceling effect can be obtained only in the very narrow frequency band range. That is, it is difficult to obtain a noise canceling effect that is practical. This is the reason why most of the noise canceling systems for headphone devices in practical use at present are of the analog type.

  However, as described above, the advantage obtained by converting the noise canceling system into a digital system is great. In view of this, the present embodiment proposes a configuration for eliminating the above-mentioned delay problem and putting it to practical use for the noise canceling system of the headphone device as described above, although it is digital. To do.

First, the background of how the inventor of the present application has constituted the noise canceling system of the present embodiment will be described with reference to FIG. In FIG. 5, the same parts as those in FIG.
FIG. 5A shows a noise canceling signal system comprising a decimation filter 5, a noise canceling signal processing unit 6 (DSP 60), and an interpolation filter 7 in the noise canceling system having the configuration shown in FIG. Shown extracted. In FIG. 4, the decimation filter 5 is shown as a single block in the A / D converter 50, but the inventor of the present application has shown the decimation filter 5 as shown in FIG. For the above, it was decided to disassemble the decimation filters 5A and 5B and give a configuration in which these are connected in series.
As can be seen from the description of FIG. 4, the decimation filter 5 converts a sampling frequency = 64 fs signal into a 1 fs signal and outputs it, that is, the sampling frequency is down-sampled to 1/64. Therefore, in the configuration shown in FIG. 5A, the decimation filter 5 that performs 1/64 downsampling includes decimation filters 5A and 5B that perform 1/8 downsampling. The decimation filter 5B is connected in series at the subsequent stage. According to this configuration, the sampling frequency = 64 fs signal input to the decimation filter 5 is first converted to a sampling frequency = 8 fs signal by the decimation filter 5A and output. Subsequently, when the signal with the sampling frequency = 8 fs is input to the decimation filter 5B, the signal is converted into the signal with the sampling frequency = 1 fs in the PCM format. In this way, depending on the series connection of the decimation filters 5A-5B, 1/64 downsampling is generally performed as represented by 1/8 × 1/8.
For confirmation, in FIG. 5A as well, the signal processing after passing through the decimation filter 5 (decimation filter 5B) is the same as in FIG. That is, the signal (PCM signal) with the 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 the PCM format signal with the sampling frequency = 1 fs. The cancellation audio signal output from the noise cancellation signal processing unit 6 is in a PCM format with a sampling frequency = 1 fs. The interpolation filter 7 inputs the cancellation audio signal and performs upsampling (interpolation). The signal is output as a signal with a sampling frequency of 8 fs.

Here, regarding a system composed of a decimation filter 5B, a noise cancellation signal processing unit 6, and an interpolation filter 7 collectively shown by a one-dot chain line in FIG. 5A, this system is composed of an input signal and an output signal. The sampling frequency is the same at 8fs. In the following, the system enclosed by the one-dot chain line is also referred to as an 8 fs input / output signal processing system.
If this 8fs input / output signal processing system is viewed as one black box, a PCM signal with a sampling frequency of 8 fs is input, and a noise canceling audio signal in the PCM format with the same sampling frequency of 8 fs is generated and output. It can be regarded as a part that executes digital signal processing (noise cancellation signal processing).

Based on the fact that the 8fs input / output signal processing system is a part having the above function, it can be considered that the configuration shown in FIG. 5B can also be adopted.
That is, only the noise cancellation signal processing unit 6A is provided as the 8fs input / output signal processing system. The noise cancellation signal processing unit 6A directly inputs a signal with a sampling frequency of 8 fs, and generates a noise canceling audio signal with a sampling frequency of 8 fs by digital signal processing corresponding to the 8 fs PCM signal format. Output.

Comparing the configuration shown in FIG. 5B with the configuration shown in FIG. 5A, in FIG. 5B, the decimation filter 5 first converts the sampling frequency by 1/8. The decimation filter (5B) for performing is omitted, and further, 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 delays in the A / D converter 50 and the D / A converter 70 are large. The cause of these delays is that the A / D converter 50 has a decimation filter. 5, the delay by the interpolation filter 7 is dominant in the D / A converter 70. Accordingly, in FIG. 5B, since the signal input / output to / from the noise cancellation signal processing unit 6A does not pass through the decimation filter 5B and the interpolation filter 7, the 8 fs input shown in FIG. Compared with the output signal processing system, that is, the configuration of FIG. 4, the signal delay is greatly reduced.
Then, by reducing the signal delay in the noise cancellation signal processing system in this way, as derived from the above description, as a frequency band of the voice that is effective for noise cancellation, a higher frequency band. It is possible to expand to the range. That is, the problem of the noise canceling system shown in FIG. 4 is solved by adopting the configuration of FIG.

Here, when the noise canceling system is actually configured according to the model shown in FIG. 5B, the configuration of the noise cancellation signal processing unit 6A should be considered.
First, the actual noise cancellation signal processing unit 6 shown in FIG. 5A is realized by first programming the DSP as described in FIG. In addition, as a digital filter format, FIR is generally used. Therefore, even when the noise canceling system based on FIG. 5B is configured, the noise cancellation signal processing unit 6A may be configured as an FIR digital filter (FIR filter) provided in the DSP. It is possible to think of it properly.

However, the sampling frequency of the signal processed by the noise cancellation signal processing unit 6A is 8 fs, which is 8 times higher than that of the noise cancellation signal processing unit 6 in FIG. . Then, in relation to the clock, the number of computations (number of processing steps) that can be performed per cycle of the sampling frequency is smaller in the noise cancellation signal processing unit 6A than in the noise cancellation signal processing unit 6. Specifically, assuming that the clock is 1024 fs, the number of operations per sampling period in the noise cancellation signal processing unit 6A having a corresponding sampling frequency of 8 fs is 1024/8 = 128. On the other hand, the noise cancellation signal processing unit 6 having 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, the processing capability as high as a DSP that executes digital signal processing corresponding to sampling frequency = 1 fs cannot be obtained. means. From this point of view, it is preferable to configure the noise cancellation signal processing unit 6A by hardware.
Further, since the characteristics as a canceling audio signal are correspondingly complicated, the noise canceling signal processing unit 6A is configured by an FIR filter, and the signal processing for making the widest possible audio frequency band as a noise canceling target can be executed. When trying to configure, an enormous degree (number of taps) is required, and the resources for processing become very large. Therefore, the inventor of the present application considered that the noise cancellation signal processing unit 6A in the case where the model shown in FIG. 5B is actually configured is configured by an IIR (Infinite Impulse Response) digital filter (IIR filter). It has been confirmed that the IIR filter can provide necessary and sufficient characteristics as an audio signal for noise cancellation. In other words, it has been confirmed that an IIR filter that can be formed with fewer orders and smaller resources than an FIR filter can be sufficiently employed to provide equivalent signal characteristics as a noise canceling audio signal.
In this way, one conclusion was obtained that it is appropriate to configure the noise cancellation signal processing unit 6A in the configuration shown in FIG. 5B with a hardware IIR filter.

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 signal delay due to this is eliminated. Thus, the frequency band where an effective noise canceling effect can be obtained is expanded to a higher frequency range. That is, although it is digital signal processing, it is possible to obtain noise canceling performance that is practical.
However, when trying to construct a noise-cancelling system in reality, it is purely possible, including the freedom of filter characteristics and design, which are the advantages of being digital, cost reduction, miniaturization and weight reduction. It is necessary to satisfy some conditions other than the noise cancellation performance.
When the noise canceling system based on FIG. 5B is actually configured, the noise cancellation signal processing execution part (noise cancellation signal processing unit 6A) is configured by only dedicated hardware, for example. However, for example, the setting of the filter characteristic is fixed, and the setting of the filter characteristic is changed according to the switching operation or the adaptive control, or the design change of the filter later tends to be limited. Incidentally, with regard to the degree of freedom in changing the filter characteristics and design, a DSP that performs digital signal processing according to a program is more advantageous.
Further, since the noise cancellation signal processing is inherently complicated, even if a hardware IIR filter is adopted for the noise cancellation signal processing unit 6A, corresponding resources are required. For this reason, depending on conditions, the noise cancellation signal processing unit 6A that is hardware may cost more than allowable, or may have to have a circuit scale and mounting area that are higher than allowable. It is believed that there is.
In view of the above, as in FIG. 5 (b), the hardware only thus be to be actually obtain noise canceling system which performs digital signal processing of the noise cancellation signal processing is much That is not realistic.

Therefore, the inventor of the present application, as shown in FIG. 5C, has two systems of a system including the noise cancellation signal processing unit 6A and a system including the noise cancellation signal processing unit 6 for the 8fs input / output signal processing system. This is a configuration that is arranged in parallel.
As described above, in the noise canceling system, as the signal delay as the noise canceling sound increases, it becomes difficult to obtain the noise canceling effect for the high frequency range. In other words, this means that the noise canceling effect can be easily obtained on the low frequency side even if there is a considerable signal delay.
Based on this, in the configuration of FIG. 5C, the noise cancellation signal processing unit 6 generates a noise cancellation signal for performing noise cancellation for all low frequencies in the audio frequency band to be noise canceled. It was decided to make it like this. On the other hand, for the noise cancellation signal processing unit 6A, a noise cancellation signal for performing noise cancellation for the mid-high range, which is higher than the low frequency range, in all the audio frequency bands to be subjected to noise cancellation. It is supposed to be configured to be generated.
In such a configuration, the noise cancellation signal processing unit 6A in charge of the middle and high frequencies among all the audio frequency bands to be subjected to noise cancellation executes the noise cancellation signal processing as the main processing, and one noise cancellation signal processing unit 6 As a sub process, it can be regarded as a part that executes the noise cancellation signal processing unit for the low frequency band.

In such a configuration, first, for the noise cancellation signal processing unit 6A configured by a hardware IIR filter, a noise canceling audio signal whose frequency band only on the middle and high frequencies excluding the low frequency is targeted for noise cancellation. Therefore, as compared with the case where the entire audio frequency band including the low frequency band is targeted for noise cancellation, the required amount of resources can be reduced accordingly. Further, by reducing the hardware resources in this way, the power consumption in the noise cancellation signal processing unit 6A is also reduced. This leads to lower power consumption of the noise canceling system. For example, when the noise canceling system is driven by a battery, it is expected that the battery will have a longer life.
Further, as described above, the noise cancellation signal processing unit 6 that executes the digital signal processing corresponding to the sampling frequency = 1 fs has the number of operations as compared with the noise cancellation signal processing unit 6A corresponding to the sampling frequency = 8 fs. Since the arithmetic processing capability is high, there is no problem in configuring with a DSP. Therefore, if the noise cancellation signal processing unit 6 is configured as a function of the DSP, it is possible to easily change and set the filter characteristics, for example. That is, the degree of freedom regarding signal processing is improved.

In this way, in the configuration of FIG. 5C, first, the problem of deterioration of noise cancellation performance due to the delay of the noise cancellation audio signal is solved. In addition, the noise cancellation signal processing unit 6A configured by hardware logic and corresponding to the sampling frequency = 8 fs can be further reduced in resources, and at the same time, a high degree of freedom can be obtained regarding noise cancellation signal processing.
The inventor of the present application has come to the conclusion that the model form shown in FIG. 5 (c) will be the best in the present situation as a noise canceling system based on the above advantages. It is. In other words, the noise canceling system as an embodiment based on the present invention includes a noise canceling audio signal system based on the model shown in FIG. 5C.

By the way, in FIG. 5C, the system on the noise canceling signal processing unit 6A side is the main system, and the noise canceling signal processing for the mid-high range is performed, and the system on the noise canceling signal processing unit 6 side is the sub. Thus, the description has been given on the assumption that the noise canceling signal processing for the low range is supplementarily performed.
As described above, in consideration of, for example, the cost and the board mounting area, the noise canceling signal processing unit 6A configured by hardware has a small circuit by reducing resources as much as possible. It can be said that it is demanded.

Therefore, the present inventor assumes a case where it is necessary to reduce resources as much as possible for the noise canceling signal processing unit 6A, for example, because it is a cost as a noise canceling system or priority is given to reduction in size and weight. And examined. As a result, as shown in FIG. 5 (d), the noise cancellation signal processing unit 6 is in charge of main noise cancellation signal processing under the same model mode as in FIG. 5 (c), and the noise cancellation signal processing unit The present inventors have also considered a configuration in which 6A is in charge of noise cancel signal processing as a sub.
In this configuration, first, for the noise cancellation signal processing unit 6, for example, out of all audio frequency bands to be noise canceled, a high frequency audio frequency above a certain level that is difficult to obtain an effective noise cancellation effect. Except for the band, the configuration is such that noise cancellation is performed for the audio frequency band as a lower mid-low range. On the other hand, the noise cancellation signal processing unit 6A is configured, for example, as a gain adjustment circuit that performs gain adjustment on an input signal, or is configured to obtain a moving average based on a value of several samples. Such a signal processing operation of the noise canceling signal processing unit 6A supplements high frequency noise canceling signal processing (generation of an audio signal for high frequency noise cancellation) that is insufficient on the noise canceling signal processing unit 6 side, for example. Equivalent to.

  And if it is the structure corresponding to the said FIG.5 (d), as the noise cancellation signal process part 6A, it is realizable by the FIR filter of about several taps, for example. In other words, the resources are very small, and the actual hardware configuration can be made low-cost and compact.

  In this way, in the present embodiment, as described with reference to FIGS. 5C and 5D, digital signal processing corresponding to different sampling frequencies is executed for the system that executes noise cancellation signal processing. By providing two systems, it is possible to obtain a noise canceling effect that is sufficiently practical in spite of digital signal processing, and it is a hardware resource or circuit scale is kept below a certain level, and a noise canceling signal The degree of freedom of setting for processing is obtained.

  By the way, the fundamental difference when FIG. 5 (a) (b) is compared with FIG. 5 (c) (d) which is the basis of the present embodiment is first shown in FIG. 5 (a) (b). 5 performs noise cancellation signal processing (generation of an audio signal for noise cancellation) by digital signal processing of only one system corresponding to sampling frequency = 1 fs or sampling frequency = 8 fs, whereas FIG. In the configuration of (d), noise cancellation signal processing is performed simultaneously by one system of digital signal processing corresponding to sampling frequency = 1 fs and one system of digital signal processing corresponding to sampling frequency = 8 fs. It can be said that it is done. That is, in the configurations of FIGS. 5A and 5B, noise cancellation signal processing is executed by digital signal processing corresponding to a specific single sampling frequency, whereas FIGS. In this configuration, noise cancellation signal processing is executed by digital signal processing of each system corresponding to two different sampling frequencies. For confirmation, the configuration shown in FIG. 4 described above is equivalent to that shown in FIG. 5A, and is therefore included in the category of the former configuration. In the latter case, the output of the system with the lower sampling frequency (1 fs) is up-sampled (interpolated) to the higher sampling frequency (8 fs), and this up-sampled signal and the sampling frequency The output of the higher system is added and output.

  Based on the difference in configuration described above, hereinafter, the former configuration corresponding to FIGS. 5A, 5B, and 4 is also referred to as “single path” for the noise cancellation signal processing system. The latter configuration corresponding to 5 (c) and 5 (d) is also referred to as “dual path”.

Hereinafter, a more specific configuration example as the noise canceling system of the present embodiment based on the model configuration shown in FIGS. 5C and 5D will be described.
First, FIG. 6 is a block diagram showing a configuration example of a noise canceling system as the first embodiment. In this figure, 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. Also, the noise canceling system shown in FIG. 6 adopts a configuration based on the feedforward system in the same manner as FIG. 4, and corresponds to one of the L and R stereo channels. is there.
In the following embodiments, the reference sampling frequency fs is 44.1 kHz corresponding to a digital audio source such as a CD.

First, in the noise canceling system of this embodiment, A / D converter 50 shown in Figure 4, DSP 60, a portion corresponding to the D / A converter 70, LSI (Large Scale Integration) of one of the 600 It is configured so as to be accommodated in a physical structural unit as an integrated circuit component.
The LSI 600 includes two signal processing units, an analog block 700 and a digital block 800, roughly divided inside.
The analog block 700 corresponds to the input / output of an analog signal, and the ΔΣ modulator 4 that is the first stage in the A / D converter 50 and the power drive circuit 10 that is the last stage in the D / A converter 70. Formed. Also, in this figure, the analog block 700 includes the power supply unit 22 and the oscillator 21. The power supply unit 22 supplies DC power to a circuit in the LSI 600 with a predetermined voltage value. The oscillator 21 outputs a clock (CLK) for a circuit in the LSI 600 (analog block 700, digital block 800) using a signal from a crystal oscillator provided outside the LSI 600 , for example. The In this embodiment, the clock frequency is 1024 fs.
The digital block 800 inputs and outputs by a digital signal including parts other than the ΔΣ modulator 4 and the power drive circuit 10 as functions corresponding to the A / D converter 50, the DSP 60, and the D / A converter 70. It is formed including the part to be performed.
The analog block 700 and the digital block 800 here are chips 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.
In addition, since the analog circuit and the digital circuit are manufactured as one chip at present, the analog block 700 and the digital block 800 can be manufactured as one chip according to this. . That is, in actuality of the present embodiment, for example, in consideration of manufacturing efficiency and other conditions, it is determined whether the analog block 700 and the digital block 800 are configured as separate chips or as a single chip. That's fine.

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. The signal obtained by collecting the sound with the microphone 2F is amplified by the amplifier 3 to become an analog audio signal. When this analog audio signal is input to the LSI 600 , it is first input to the ΔΣ 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) format digital signal. In this case, the digital signal as the output of the ΔΣ modulator 4 is input to one input terminal of the switch SW1.

In the noise canceling system of the present embodiment, in consideration of extensibility, consideration is given so that input from a digital microphone (digital microphone) can be supported at the microphone input stage. For this reason, the LSI 600 is a digital microphone. Digital audio signals from can be input.
The digital microphone is configured, for example, by integrating at least a microphone and a ΔΣ modulator that converts a signal obtained by collecting the sound into a 1-bit digital audio signal. An input signal from the digital microphone is input to the other input terminal of the switch SW1.

The switch SW1 is switched so 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 housing in accordance with the feed forward method. The digital audio signal that is the output of the switch SW1 is input to the decimation filter 5A .

The decimation filter 5A corresponds to the decimation filter 5 shown in FIG. 4 by connecting the decimation filter 5A in series with the subsequent decimation filter 5B. Since the decimation filters 5A and 5B are formed so as to decimate to a sampling frequency of 1/8, respectively, the decimation filters 5A and 5B are connected in series, whereby (1/8) × (1/8) = As represented by 1/64, the sampling frequency is decimated to 1/64. That is, the 64 fs input signal is converted to 1 fs in the same manner as the decimation filter 5.
Here, the decimation filter 5A has a fixed filter characteristic, whereas the decimation filter 5B can be configured so that the filter characteristic can be varied as described later.

  First, the decimation filter 5A converts an input 64fs, 1-bit signal into an 8fs, 24-bit signal by performing a so-called thinning process that thins out data using a predetermined thinning pattern corresponding to the sampling period. Output. That is, the decimation filter 5A performs 1/8 decimation (downsampling) regarding the sampling frequency processing. This output is input to the decimation filter 5B and branched to be input to the noise cancellation signal processing unit 6A.

The noise cancellation signal processing unit 6A is formed by a digital filter, generates a noise cancellation audio signal of 8 fs, 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, a noise canceling audio signal is also generated by the noise canceling signal processing unit 6 in the DSP 60 as will be described later.
Therefore, hereinafter, a signal generated by the noise cancellation signal processing unit 6 is referred to as a first noise cancellation audio signal, and a signal generated by the noise cancellation signal processing unit 6A is referred to as a second noise cancellation audio signal. I will distinguish between the two.

  The decimation filter 5B performs 1/8 downsampling in the same manner as the previous decimation filter 5A. That is, the input 8 fs, 24 bit signal is converted into a PCM (Pulse Code Modulation) signal in a 1 fs, 16 bit format, for example, and output to the DSP 60.

The DSP 60 is provided to input 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 necessary signal processing on each of them. Further, the DSP 60 in this case is configured to be capable of signal processing corresponding to a PCM signal format of 1 fs, 16 bits, for example.
The signal processing function executed by the DSP 60 can be obtained by programming. This program is stored and held as instruction data in the flash memory 16, for example. The DSP 60 appropriately executes signal processing by reading out necessary instructions from here and executing them.

In the DSP 60 as the present embodiment, first, the noise cancellation signal processing unit 6 generates a first noise cancellation audio signal using the signal input from the decimation filter 5B. The noise cancellation signal processing unit 6 is formed by a digital filter.
Further, 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 as a predetermined functional part in the digital block 800. It is also possible to change and set the characteristics of the digital filter.
First, 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.
In addition, the filter characteristics can be variably set for the digital filter as the noise cancellation signal processing unit 6A.
In addition, the filter characteristics can be variably set for the digital filter as the decimation filter 5B.
Further, the filter characteristics can be variably set for the anti-imaging filter 7b in the interpolation filter 7.
In changing the filter characteristics for the digital filter, first, a filter characteristic table is stored in advance in the flash memory 16, and the filter characteristics corresponding to the analysis result are read out. The Then, parameters such as the number of taps and coefficients as the read filter characteristics are set to obtain a digital filter configuration with desired characteristics.
In addition, for example, in the RAM 15, an area for storing the filter characteristic table is secured, and the acoustic analysis processing unit 62 newly performs a calculation based on the analysis result to generate a new filter characteristic. Can be stored in the filter characteristic table of the RAM 15. In this way, if the acoustic analysis processing unit 62 can adaptively generate the filter characteristics according to the analysis result, the degree of freedom and adaptability for the characteristics to be set in the digital filter is increased, and better noise is achieved. A canceling effect can be expected.

  In addition, the equalizer 61 can output the signal of the digital audio source input as described later after adjusting and correcting the sound including the sound quality adjustment.

  The first noise cancellation 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 an 8 fs, 24-bit signal by executing a process of multiplying the sampling frequency by 8 times, and outputs this to the synthesizer 12. . Here, the interpolation filter 7 is shown as comprising an oversampling circuit 7a and an anti-imaging filter 7b. In other words, the interpolation filter 7 first converts the input 1 fs, 16 bit signal into the 8 fs, 24 bit format by the oversampling circuit 7a, and then samples it as an image frequency component by the anti-imaging filter 7b. Signal processing is performed so as to remove frequency components higher than 1/2 of the frequency 8 fs.

  In this case, an audio signal as a digital audio source is in a 1 fs, 16-bit format via the PCM interface 13 and input to the DSP 60. Further, in this case, it branches and is supplied to one input terminal of the switch SW2. In the DSP 60, the input signal of the digital audio source is subjected to predetermined equalization processing by the equalizer 61 and then input to the other input terminal of the switch SW2.

The switch SW2 is switched so as to selectively connect one 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. Accordingly, in response to the switch SW2, the digital audio source signal output from the PCM interface 13 is input to the interpolation filter 14 without passing through the DSP 60, and the digital output from the PCM interface 13 is used. The audio source signal is switched through the path through which the signal is input to the interpolation filter 7 after passing through the DSP 60.

  As described above, a digital audio signal of 1 fs and 16 bits as a digital audio source is input to the interpolation filter 14. The interpolation filter 14 executes a process of increasing the sampling frequency by 8 times for the input signal, converts the input signal into a signal having a format of 8 fs and 24 bits, and outputs the signal to the synthesizer 12.

In this case, the synthesizer 12 is a digital audio source audio signal in the format of 8 fs and 24 bits, respectively, and the first noise cancellation audio output from the noise cancellation 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 synthesizing a synthesized noise canceling audio signal obtained by synthesizing the components of the first and second noise canceling audio signals with an audio signal as a digital audio source. It will be.
This audio signal is first subjected to noise shaping by the noise shaper 8 to be a 16 fs, 4-bit digital signal, and further subjected to PWM modulation by the PWM circuit 9 to be converted to a 512 fs, 1-bit digital signal. . Then, the digital signal based on 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. This amplified analog signal is supplied to the driver 1a via the filter 11 and the capacitor C1 outside the LSI 600 .
The input signal of the power drive circuit 10 can be branched and output to the outside (external 1-bit output).

Here, when the configuration of the noise canceling system as the present embodiment shown in FIG. 6 is compared with the configuration shown in FIG. 4, the following can be said.
In the configuration shown in FIG. 6, the signal system for noise cancellation corresponding to FIG. The signal system consists of filter 7 synthesizer 12 noise shaper 8 PWM circuit 9 power drive circuit 10 filter 11 capacitor C1 driver 1a. This is a signal system for generating a first noise cancellation audio signal and outputting it as sound from the driver 1a. In addition, in FIG. 6, a noise cancellation signal processing unit 6A is provided. That is, another noise cancellation signal system 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 this embodiment, two systems are provided as systems for forming a noise canceling audio signal based on a signal obtained by collecting sound by the microphone 2F.
That is, in the system (the first noise cancellation signal processing system) for generating a first noise cancellation-use audio signal includes a noise cancellation signal processing section 6 within the DSP 60, de decimation filter 5A, the decimation filter 5B, the noise The signal is propagated in the order of the cancel signal processing unit 6, the interpolation filter 7, and the combiner 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 the second noise cancellation audio signal (second noise cancellation signal processing system), the decimation filter 5A, the noise cancellation signal processing unit 6A, and the synthesis The signal is propagated in the order of the device 12. In other words, the first noise cancellation signal processing system is similar to the configuration of the noise canceling system shown in FIG. 4 and the A / D conversion side decimation filter (5A, 5B) and the D / A conversion side interface. Whereas the signal passes through the poration (oversampling) filter 7 , the second noise cancellation signal processing system passes through the decimation filter 5A but passes through the decimation filter 5B. Further, the interpolation filter 7 is also passed so that it passes only through the noise cancellation signal processing unit 6A in which a signal of sampling frequency = 8 fs is input and output. After that, the signal obtained by the first and second noise cancellation signal processing systems is synthesized by the synthesizer 12 to obtain a comprehensive noise cancellation audio signal.
This configuration is nothing but the configuration of the noise cancellation signal processing system as “dual path” described above with reference to FIGS.

  Here, in the dual path configuration of the present embodiment having the first and second noise cancellation signal processing systems as described above, the respective model configurations shown in FIGS. Correspondingly, two basic modes can be taken based on the functions and roles to be given to each of the first and second noise cancellation signal processing systems. Therefore, first, these two functional modes will be described.

  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 mode example, which is one of the two functional mode examples, will be described with reference to FIG.

First, as a first functional mode example, as shown in FIG. 7, a noise cancellation signal processing unit for forming a noise canceling audio signal as shown in FIG. The noise cancellation signal processing unit 6 belonging to is treated as a main processing unit, and the noise cancellation signal processing unit 6A belonging to one second noise cancellation signal processing system is treated as a sub processing unit . That is, the configuration corresponds to FIG.
In this case, as described above, the digital filter of the noise cancellation signal processing unit 6 serving as the main processing unit has an effective noise cancellation effect in all the audio frequency bands targeted for noise cancellation. The configuration is such that noise cancellation signal processing corresponding to a frequency band range below a certain value obtained is executed. That is, the first noise cancellation signal processing system including the noise cancellation signal processing unit 6 includes the decimation filter 5B and the interpolation filter 7, and thus has a signal delay. Although it is difficult to expect an effect, here, a high frequency band higher than a certain level is excluded, and a noise canceling audio signal for a frequency band range as a lower middle frequency range is generated.
In addition, the digital filter of the noise canceling signal processing unit 6A serving as the sub processing unit is formed so as to generate a noise canceling audio signal having a characteristic in which noise canceling is performed for the high frequency band.
Then, as a result of the noise canceling audio signals of the main processing unit and the sub processing unit being combined by the combiner 12, the total noise canceling audio signal output from the combiner 12 is necessary as a noise cancel target. A function for producing an effective noise canceling effect over the entire audio frequency band is provided.
In this way, as described above, as the first functional mode example, first, after the first noise cancellation signal processing system performs noise cancellation for the middle and low range. In the first noise canceling signal processing system, the high frequency range where it is difficult to obtain a sufficient noise canceling effect is configured to be supplementarily canceled by the second noise canceling signal processing system with less signal delay. 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).
The noise cancellation signal processing unit 6A in this case is a simple gain adjustment circuit or a simple circuit such as a circuit for obtaining a moving average value using a FIR filter of several taps , as described above with reference to FIG. This can be realized with a simple hardware configuration, and can greatly reduce resources and circuit scale. In this case, the noise canceling signal processing unit 6 in the DSP 60 does not need to be configured with the intention of effectively canceling the noise in the high frequency range, so that the resources are reduced accordingly. This is also advantageous in terms of processing capacity. In addition, since the configuration is simpler than before, it is expected that the filter design as the noise cancellation signal processing units 6 and 6A is simplified.

Next, a second functional mode example will be described with reference to FIG. In this figure, the same parts as those in FIG.
As a second functional mode example, contrary to the first functional mode example described with reference to FIG. 7, the second noise cancellation signal processing system is the main signal processing system, and the first noise cancellation signal processing system is used. Is a sub signal processing system. Accordingly, the noise cancellation signal processing unit 6A belonging to the second noise cancellation signal processing system becomes the main processing unit, and the noise cancellation signal processing unit 6 belonging to the first noise cancellation signal processing system becomes the sub processing unit. That is, the configuration corresponds to FIG.
As for the division of roles, as described in FIG. 5C, the noise cancellation signal processing unit 6A as the main is subject to noise cancellation in the middle and high frequencies in all the audio frequency bands targeted for noise cancellation. The noise cancellation signal processing unit 6 is configured to generate a noise cancellation signal for performing noise cancellation, and the noise cancellation signal processing unit 6 as a sub is configured to perform noise cancellation for all low frequencies in the audio frequency band targeted for noise cancellation. The noise canceling signal is generated.
Also in this case, the noise canceling audio signal obtained by synthesizing the noise canceling audio signals of the main processing unit and the sub processing unit by the synthesizer 12 is the total audio frequency required as a noise canceling target. A function for producing an effective noise canceling effect over a band is provided.

  Note that, in actually configuring the noise canceling system based on the present embodiment, it is required for the noise canceling system as to which of the first functional aspect example and the second functional aspect example is adopted. Appropriate one may be selected according to various conditions such as cost and specifications. As can be understood from the description of FIGS. 5C and 5D, the first functional mode example is used when priority is given to low cost and to make the circuit scale as small as possible. Adopting is more advantageous. On the other hand, in the second functional mode example in which the noise cancellation signal processing unit 6A by hardware executes the main signal processing, a higher quality noise cancellation effect can be expected. Therefore, when priority is given to providing high-quality reproduced sound, it is appropriate to adopt the second functional example.

Here, the configuration of a digital filter employed in a predetermined functional circuit unit related to a signal processing system for noise cancellation in the digital block 800 of the noise canceling system of the present embodiment will be described.
For example, in the noise canceling system previously shown in FIG. 4, the decimation filter 5 (5A, 5B) and the interpolation filter 7 are configured by a linear phase FIR. This is based on the idea that it is generally necessary not to cause phase distortion according to the frequency because the processing target is an audio signal as described above. It is.
By adopting a linear phase FIR, a group delay occurs between input and output. However, conventional A / D converters and D / A converter devices reproduce audio sound sources that users actively listen to. Since it was assumed to be used for (recording), there was no particular problem. For example, in the case of playing an audio source, even if a corresponding delay occurs due to signal processing after the signal of the audio source is input to the signal processing device and played as a sound, This is because nothing is normally reproduced and output continuously. Therefore, when the user reproduces and listens to the audio sound source, the delay of signal processing is not regarded as a problem.
However, when trying to divert a conventional device to a noise canceling system instead of playing an audio source, it is impossible or difficult to obtain a phase that can cancel external sound due to the group delay of the device. This will come up as a problem.

In the noise canceling system of the embodiment shown in FIG. 6, this problem is first solved by the second noise cancellation signal processing including the noise cancellation signal processing unit 6 </ b> A that does not pass through the decimation filter 5 </ b> B and the interpolation filter 7. This is solved by providing a system.
However, if the signal delay that is noticeable in the decimation filter 5 (5A, 5B) and the interpolation filter 7 is further reduced on the first noise cancellation signal processing system side, the noise canceling effect can be obtained. As a result, the number of obstructive factors is reduced, and a better effect can be expected.

Therefore, in the present embodiment, as an example, a configuration in which the decimation filter 5B shown in FIG. It is made to take.
Note that the minimum phase transition FIR digital filter is based on the fact that the minimum phase can be obtained as the FIR digital filter system, and the peak value is set on the top side (side closer to the input) for the tap coefficient. Can be formed.

For example, when comparing impulse response waveforms as the characteristics of a digital filter of linear phase type FIR configured with the same number of taps and a digital filter of minimum phase shift type FIR, first, in the linear phase type FIR, The peak is obtained at a timing delayed for a certain time. This indicates that the output in response to the input has a delay (group delay) by a fixed time corresponding to the number of taps (order). On the other hand, in the minimum phase transition type FIR, a peak is obtained at a fast timing corresponding to, for example, about several taps with respect to the input timing. That is, although the same FIR digital filter is used, the minimum phase transition type FIR has a very short output delay (input / output delay) in response to the input as compared with the linear phase type FIR.
Therefore, if the minimum phase transition 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 reduced. This factor is almost eliminated. As a result, the first noise cancellation signal processing system is expected to obtain better noise cancellation capability.

As is well known, in the case of the minimum phase transition type FIR, phase distortion corresponding to the frequency occurs. Therefore, in the case of an audio signal, the possibility of sound quality degradation due to this phase distortion is unavoidable. This is the reason why the A / D converter corresponding to the audio signal and the digital filter mounted on the D / A converter have been linear phase type FIRs so far.
However, although the signal processing target in this case is an audio signal, for example, it is an external sound that is a noise cancellation target, and the required reproduction fidelity is considerably low compared to an audio source or the like. In addition, the sound component that is actually considered to have a large cancellation effect is a low frequency band called a so-called low band, and noise cancellation up to several kilohertz is effective in consideration of device characteristics. If it works, it is said to be sufficient for practical use. From this point of view, even if the decimation filter 5B and the anti-imaging filter 7b are formed by the minimum phase shift FIR, for example, there is almost no sound quality problem.

Further, according to the above description, the decimation filter 5A and the oversampling circuit 7a are not the minimum phase transition type FIR in the components of the decimation filter 5 and the interpolation filter 7. That is, these portions are linear phase FIR.
This is because signal delay factors 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, for the decimation filter 5A and the oversampling circuit 7a, signal delay in the signal processing system via the noise cancellation signal processing unit 6 is particularly a problem even if a linear phase FIR is used with emphasis on reproduction sound quality, for example. It must not be.

  If the purpose is to reduce the signal delay between input and output as described above, it is also appropriate to configure the decimation filter 5B and the anti-imaging filter 7b with IIR (Infinite Impulse Response) filters. It will be a thing. The impulse response waveform of the IIR filter also shows a characteristic that a peak is obtained at a fast timing corresponding to, for example, several taps with respect to the input timing, that is, the input / output delay is very short, and the minimum phase transition type In the same manner as in the case of the FIR configuration, the signal delay for the first noise cancellation signal processing system can be made smaller than before.

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 type 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 according to programming (instruction), for example.
In the first functional mode example in which the noise cancellation signal processing unit 6 is the main processing unit, even if the signal processing function in the DSP 60 is realized by programming, resources can be reduced. Considering this point, it is preferable to configure the noise cancellation signal processing unit 6 with an IIR filter.

The digital filter as the noise cancellation signal processing unit 6A, which belongs to the other second noise cancellation signal processing system, is mounted as dedicated hardware for generating a noise cancellation signal. The In addition, the noise cancellation signal processing unit 6A is configured by a linear phase FIR or IIR filter.
However, at present, according to the second functional mode example, the second noise cancellation signal processing system (noise cancellation signal processing unit 6A) is used as the main, and the first noise cancellation signal processing system (noise cancellation signal processing unit 6) is used. When considering the sub-configuration, as described above with reference to FIG. 5C, the noise canceling signal processing unit 6A is configured by an IIR filter while suppressing the necessary resource amount. This is advantageous for the purpose of obtaining a high-quality noise cancellation effect.

In addition, when the second functional mode example is adopted, the noise cancellation signal processing unit 6A configured as hardware is also variably set with respect to its characteristics within a certain degree of freedom. If possible, it can be said that, for example, noise canceling signal processing with higher adaptability can be performed than when the characteristics are variably set only by the noise canceling signal processing unit 6 on the DSP 60 side.
Therefore, as a case where an IIR filter is employed 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.

First, a plurality of secondary IIR filters are provided as digital filters forming the noise cancellation signal processing unit 6A. Here, in consideration of the actual number of calculation steps and the like, five IIR filters 65-1, 65-2, 65-3, 65-4, and 65-5 are prepared as secondary IIR filters. In addition, according to the characteristics required for the noise cancellation signal processing unit 6A, the connection mode patterns of these IIR filters 65-1 to 65-5 are 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 final-stage IIR filter 65-5.
FIG. 10 shows a pattern in which a system in which four IIR filters 65-1, 65-2, 65-3, 65-4 are connected in series and a system in which only the remaining one IIR filter 65-5 is provided in parallel. 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 outputted from the noise cancellation signal processing unit 6A.
FIG. 11 shows in parallel a system in which three IIR filters 65-1, 65-2, 65-3 are connected in series and a system in which the remaining two IIR filters 65-4, 65-5 are connected in series. The provided 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 outputted from the noise cancellation signal processing unit 6A.
FIG. 12 shows a system in which three IIR filters 65-1, 65-2, and 65-3 are connected in series, a system that includes only the IIR filter 65-4, and a system that includes only the IIR filter 65-5. Each pattern connected in parallel is 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 then outputted from the noise cancellation signal processing unit 6A.
FIG. 13 shows 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 filter 65-5. A pattern in which a system consisting of the above 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 outputted 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 by only IIR filter 65-3, a system by only IIR filter 65-4, and only by IIR filter 65-5. A pattern in which the 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 outputted from the noise cancellation signal processing unit 6A.
FIG. 15 shows a pattern in which five IIR filters 65-1, IIR filter 65-2, IIR filter 65-3, IIR filter 65-4, and IIR filter 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 then outputted 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) according to the time axis by using a method such as a sequencer. Is.

  In addition, when the first functional mode example is adopted, it has been described that the noise cancellation signal processing unit 6 in the DSP 60 is preferably configured by an IIR filter. 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 employed by programming the DSP 60.

Therefore, the first functional example is adopted as the noise canceling system of the present embodiment, and the IIR filter 65 in the case where the pattern shown in FIG. 9 is adopted as the noise cancellation signal processing unit 6 in the DSP 60. a setting example of a characteristic for each -1 65 -5, and that shown in FIG. 16.
In this case, first, the IIR filter 65-1 in the first stage is given a function as a gain setting circuit that gives a gain to an input signal and outputs the gain. Here, 0.035 is set as the gain coefficient (Gain).
The IIR filters 65-2 to 65-5 from the second stage to the fifth stage (final stage) are provided with a function called a so-called parametric equalizer. As the equalizer characteristics, the center frequency fc = 20 Hz, the Q value = 0.4, and the gain value G = 28 dB are set for the IIR filter 65-2, and the center frequency fc = 800 Hz for the IIR filter 65-3. The Q value = 0.6 and the gain value G = 12 dB are set. For the IIR filter 65-4, the center frequency fc = 10000 Hz, the Q value = 3.2, the gain value G = -21 dB is set, and the IIR filter 65 is set. For −5, the center frequency fc = 18500 Hz, the Q value = 2.5, and the gain value G = −16 dB are set.
Although not shown here, the noise cancellation signal processing unit 6A is configured as a gain adjustment circuit in correspondence with the configuration of the noise cancellation signal processing unit 6 described above. For the gain coefficient, for example, 0.012 is set.

Here, a noise canceling system (single-path configuration noise canceling system) based on the configuration (design) based on FIG. 4 and the noise canceling system (dual path configuration) of the present embodiment configured (designed) based on FIG. FIG. 21 is a Bode diagram showing the result of comparing the characteristics of the noise canceling system having the configuration. As a Bode diagram, FIG. 21 (a) shows frequency vs. gain characteristics and frequency vs. phase characteristics of a noise canceling system with a single path configuration based on FIG. 4, and FIG. 21 (b) shows FIG. The frequency vs. gain 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 transition FIR is adopted for the digital filter as the decimation filter 5B and the anti-imaging filter 7b in FIG. It shall consist of.
For example, here, the target frequency-to-gain characteristic to be obtained for the feed-forward noise canceling system is the characteristic indicated by the broken line in each of the frequency-to-gain characteristic diagrams in FIGS. Is assumed. The reason why the upper limit of the frequency of the target characteristic indicated by the broken line is set to 2 kHz depends on the fact that the frequency band of the voice to be controlled for noise cancellation is actually about 2 kHz. Further, in the frequency vs. gain characteristic shown in FIG. 21 (b), a gain above a certain level is maintained up to around 100 kHz, whereas in the frequency vs. gain characteristic shown in FIG. 21 (a) , around 20 kHz. It has a characteristic of steep decay. This is expressed by fs / 2 in order to avoid aliasing based on the sampling theorem because the noise canceling system having the configuration based on FIG. 4 performs noise cancellation processing on a signal having a sampling frequency of only 1 fs. This is because a band higher than the sampling frequency is removed. Incidentally, in this case, fs = 44.1 kHz, and therefore the frequency vs. gain characteristic shown in FIG. 21 (a) shows the result of attenuating a frequency band higher than 22.05 kHz.

  Here, for example, comparing FIG. 21 (a) and FIG. 21 (b), first, the frequency-to-gain characteristics are substantially the same in the frequency band range up to about 2 KHz that is actually subject to noise cancellation. It has become. However, with respect to the frequency vs. phase characteristics, the dual path configuration in FIG. 21B has a value close to 0 deg. In the range of about 2 kHz to 10 kHz, whereas FIG. Looking at the same range of about 2 kHz to 10 kHz for the single path configuration, the absolute value fluctuates so as to cause a phase rotation of 100 degrees or more. In this way, the noise canceling system based on the present embodiment has obtained the result that the phase rotation of the signal is greatly reduced even in practice, and it is sufficient for practical use even though it is a digital system. Obtaining a noise canceling system is also easily possible in reality.

  FIG. 17 shows a configuration example of a noise canceling system as the second embodiment. In this figure, the same parts as those in FIG. 6 corresponding to the previous first embodiment are denoted by the same reference numerals and description thereof is omitted.

  As described above with reference to FIGS. 1 to 3, a noise canceling system for a headphone device is roughly classified into a feed forward method and a feedback method. And the 1st Embodiment described so far is set as the structure based on a feedforward system. The present invention can be applied 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 system whose model is shown in FIG. 1 is shown.

In the case of the feedback method, as schematically shown in FIG. 17, the microphone 2B can collect sound output from the driver 1a in the vicinity of the ear of the headphone wearer inside the headphone unit 1c. It is made to provide in the position.
The sound picked up by the microphone 2B at this position includes, together with the sound output from the driver, for example, an external sound component that enters the case of the headphone device and is heard by the headphone device user's ear. It is. The sound signal collected in this way is amplified by the amplifier 3A to be an analog audio signal, and further input to the ΔΣ modulator 4A of the analog block 700 of the LSI 600, where 64fs, 1-bit digital audio. The signal 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.
Even in this case, in consideration of expandability, a digital microphone input is provided in parallel with the microphone 2B. The digital audio signal from the digital microphone input and the microphone output from the ΔΣ modulator 4A are provided. The digital audio signal from 2B can be selected by the switch SW11.

  The decimation filter 5-1 is a filter for decimating a 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. 6 corresponds to the decimation filter 5. In addition, the decimation filters 5C and 5D that form the decimation filter 5-1 correspond to the decimation filters 5A and 5B, respectively, in correspondence with FIG. The signal decimated to 8 fs by the decimation filter 5C is branched and input also to the noise cancellation signal processing unit 6B, and the signal decimated to 1 fs by the decimation filter 5D is sent to the noise cancellation signal processing unit 6 in the DSP 60. Will be entered. The noise cancellation signal processing unit 6B is provided in the second noise cancellation signal processing system that is provided corresponding to the feedback method, and the noise cancellation signal processing unit 6A corresponds to the correspondence with FIG.

In this case, the noise cancellation signal processing units 6 and 6B, for example, give necessary characteristics to the input signal, so that the external sound that reaches the ear of the headphone wearer's driver 1a side as an audio signal for noise cancellation can be heard. A sound audio signal having a characteristic that can be canceled is generated. In general, this is a process of giving a transfer function −β for noise cancellation to a collected sound signal.
For these noise cancellation signal processing units 6 and 6B, the concept of the first and second functional modes described in the first embodiment and the first and second functional modes are also described. Can be applied. Further, the same format and configuration as in the first embodiment can be applied to the format and configuration of the digital filter as the noise cancellation signal processing units 6 and 6B.

As for the feedback method, it is effective to use the equalizer 61 in the DSP 60 by including it in the first noise cancellation signal processing system in order to obtain a good noise cancellation effect.
In this case, the equalizer 61 is for giving a characteristic of a transfer function of 1 + β to the signal of the digital audio source. In the case of the feedback method, the noise cancellation audio signal output from the noise cancellation signal processing unit 6 picks up 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 1a . Ingredients are also included. That is, a characteristic corresponding to the transfer function represented by 1/1 + β is given to the sound component of the digital audio source. In view of this, the equalizer 61 preliminarily gives the characteristics of the digital audio source signal by the transfer function of 1 + β which is the inverse of 1/1 + β. Thus, at the stage where the signal of the digital audio source from the interpolation filter 14 is combined with the noise canceling audio signal by the combiner 12, the above 1/1 + β transfer characteristic is canceled. As a result, the signal output from the synthesizer 12 can be obtained by synthesizing the signal component having the characteristic of canceling the external sound and the signal component of the original digital audio source.

The configuration subsequent to the synthesizer 12 in this case is the same as 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 sound.
In this manner, in the feedback method, the external sound component mixed with the sound output from the driver in the vicinity of the ear of the headphone wearer is collected to generate a signal for noise cancellation. The noise canceling signal is output from the driver with negative feedback applied. As a result, the external sound is canceled out to the ear corresponding to the driver 1a of the headphone device person, and the sound in which the sound of the digital audio source is relatively emphasized arrives and can be heard.
Even in the configuration of the noise canceling system corresponding to such a feedback system, the first noise cancellation signal processing system that passes through the noise cancellation signal processing unit 6 of the DSP 60 is provided in the same manner as in the first embodiment. In addition, by providing the second noise cancellation signal processing system via the noise cancellation signal processing unit 6B , it is possible to obtain the same effect as that of the first embodiment.

  FIG. 18 shows a configuration example of a noise canceling system according to the third embodiment. In this figure, parts that are the same as those in FIGS. 6 and 17 corresponding to the first and second embodiments are given the same reference numerals, and descriptions thereof are omitted.

The third embodiment adopts a configuration in which a feedforward system corresponding to the first embodiment and a feedback system corresponding to the second embodiment are used in combination.
As described above, the feedback method and the feedforward method have a characteristic of a relationship that is a trade-off between them.
For example, in the feedforward system, the frequency band in which noise can be effectively canceled (attenuated) is wide and the system is highly stable, but it is difficult to obtain a sufficient amount of noise cancellation. For this reason, depending on the situation such as the positional relationship with the noise source, for example, it may not match the transfer function of the system, and noise may not be canceled or increased in a specific frequency band, for example. Has been pointed out. In this case, although noise cancellation is actually effective over a wide frequency band, a phenomenon that noise becomes conspicuous only in a specific frequency band occurs. This makes it difficult to feel the noise cancellation effect.
On the other hand, the feedback system has a feature that a sufficient amount of noise cancellation can be obtained although the frequency band in which noise can be canceled is narrow.
From this, if a noise canceling system is constructed by combining a feedback method with a feedforward method, it is easy to effectively cancel noise noise over the entire wide frequency band by making up for each other's disadvantages. It becomes possible. In other words, a better noise cancellation effect can be expected than when only one of the methods is used.

  In the configuration of the third embodiment shown in FIG. 18, first, as corresponding to the feedforward system, the microphone 2F, the amplifier 3, the ΔΣ modulator 4, A switch SW1, a decimation filter 5 (decimation filters 5A and 5B), and a noise cancellation signal processing unit 6A are shown. Further, as corresponding to the feedback system, the microphone 2B, the amplifier 3A, the ΔΣ modulator 4A, the switch SW11, the decimation filter 5-1 (decimation filters 5C and 5D), and the noise cancellation signal are provided in the same manner as in FIG. The processing unit 6B is shown.

In this case, the noise canceling signal processing unit 6 of the DSP 60 inputs a signal from the decimation filter 5B corresponding to the feedforward system and a signal from the decimation filter 5D corresponding to the Fordback system. It is shown that a noise canceling audio signal is generated and output.
Actually, in this case, the noise cancellation signal processing unit 6 inputs a signal from the decimation filter 5B and generates a noise cancellation audio signal corresponding to the feedforward method, and a signal from the decimation filter 5D. And a filter for generating an audio signal for noise cancellation corresponding to the feedback method. For example, the noise canceling signal processing unit 6 synthesizes the noise canceling audio signals generated by these filters and then outputs them to the interpolation filter 7.

  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 combined. The synthesized signal is output to a subsequent circuit (noise shaper 8).

  As described above, the noise canceling system according to the third embodiment includes the configurations of the first and second noise cancellation signal processing systems as the feedforward system corresponding to FIG. 6 and the feedback system corresponding to FIG. And the first and second noise cancellation signal processing systems. With this configuration, as described above, a better noise canceling effect can be obtained than in the case where only one of the methods is used.

FIG. 19 shows a configuration example of a noise canceling system as the fourth embodiment. Note that the noise canceling system shown in this figure corresponds to the feed-forward system, and the constituent parts are the same as those in FIG.
In the first embodiment shown in FIG. 6, the digital block 800 is described as being manufactured as one chip. However, when attention is paid to the sampling frequency of signals input / output in the functional circuit section in the digital block 800, it is understood that this is not uniform and there are several types. In this way, when the corresponding sampling frequencies are different between the functional circuit units, when considering the conditions for actually manufacturing an LSI, the corresponding sampling frequencies for the functional circuit units provided in the digital block 800 are as follows. In some cases, it may be more efficient to divide into groups based on the above, and to assemble the chips into groups for each grouped functional circuit portion.
Therefore, in this embodiment, the chip that forms the digital block 800 is configured as follows.
First, considering the main sampling frequency of signals handled in the digital block 800 shown in FIG. 19, one is 1 fs mainly composed of the DSP 60 corresponding to the first noise cancellation signal processing system. Yes, the other can be seen as 8 fs corresponding to the second noise cancellation signal processing system.

  Therefore, as shown in the present embodiment, as shown in the figure, first, the first signal processing chip 810 is manufactured as one chip at least having a circuit portion as the DSP 60 corresponding to 1 fs. At least circuit portions as a decimation filter 5 (5A, 5B), a noise cancellation signal processing unit 6A, an interpolation filter 7, an interpolation filter 14, and a synthesizer 12 are formed as functional circuit units corresponding to 8 fs. The second signal processing chip 820 is manufactured as one chip.

  It should be noted that in the figure, the functional circuit portion in the digital block 800 that is not included in either the first signal processing chip 810 or the second signal processing chip 820 is appropriately selected from the first signal processing chip 810 and the second signal. Included in these chips as included on the appropriate side of the processing chip 820 or as other chips than the first signal processing chip 810 and the second signal processing chip 820 are manufactured. It may be.

The configuration of the fourth embodiment shown in FIG. 19 is the same as that of the digital block 800 of the noise canceling system corresponding to the feedback system shown in FIG. 17 as the second embodiment. It is applicable.
That is, the first signal processing chip 810 having at least a circuit portion serving as a DSP 60 corresponding to 1 fs, and a decimation filter 5-1 (5C, 5D), a noise canceling signal processing unit 6B, as a functional circuit unit corresponding to 8 fs, The second signal processing chip 820 in which each circuit portion as the interpolation filter 7, the interpolation filter 14, and the synthesizer 12 is formed is manufactured.

Furthermore, the configuration of the fourth embodiment can be applied to the digital block 800 of the noise canceling system using both the feedforward method and the feedback method shown in FIG. 18 as the third embodiment. This configuration is shown in FIG. 20 as a fifth embodiment.
In FIG. 20, decimation filters 5, 5-1 (5A, 5B, 5C, 5D) are provided as a first signal processing chip 810 having at least a circuit portion serving as a DSP 60 corresponding to 1 fs and a functional circuit unit corresponding to 8 fs. ), The noise cancellation signal processing units 6A and 6B, the interpolation filter 7, the interpolation filter 14, and the second signal processing chip 820 in which each circuit portion as the synthesizer 12 is formed.

  Note that the sampling frequency and the number of quantization bits of signals input / output in the functional circuit section in the LSI 600 described in the embodiments so far are only representative ones, and the noise canceling system The sampling frequency and the number of quantization bits to be handled by each functional circuit unit may be changed as necessary within a range that does not cause a failure in the formation of the system.

Further, in the embodiments so far, a dual path mode is shown in which two systems of the first noise cancellation signal processing system and the second noise cancellation signal processing system are shown. For example, a configuration in which a plurality of second noise cancellation signal processing systems are further provided is also conceivable under the present invention. In such a configuration, for example, a signal having a different sampling frequency is input for each of a plurality of series of the second noise cancellation signal processing system to generate an audio signal for noise cancellation, and the roles of each system are shared. Can be considered. A 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 multipath configuration in which two or more second noise cancellation signal processing systems are provided as described above, a model example of the signal processing system that is the basis of this multipath configuration is shown in FIG. I will do it.
In FIG. 22, as a model example, a configuration in which a signal with a sampling frequency = 64 fs is converted into a multipath and finally synthesized and output in the same 64 fs format is shown.
In this figure, first, downsampling circuits 91-1 to 91-6, signal processing blocks 92-0 to 92-6, upsampling circuits 94-1 to 94-6, and synthesizers 93-0 to 93-5 are shown. Is provided.
Each of the downsampling circuits 91-1 to 91-6 downsamples the sampling frequency of the input signal to ½ and outputs it. These down-sampling circuits 91-1 to 91-6 are connected in series and input an input signal having a sampling frequency of 64 fs to the first-stage down-sampling circuit 91-1. As a result, the downsampling circuits 91-1 to 91-6 output signals obtained by converting the sampling frequency of the input signal to 32 fs, 16 fs, 8 fs, 4 fs, 2 fs, and 1 fs, respectively. Note that the number of quantization bits of a signal with a sampling frequency of 32 fs or less is set to a multi-bit with a predetermined number of bits.
The signal processing blocks 92-0 to 92-6 are portions for executing signal processing according to a predetermined purpose for the input signal, and are, for example, digital filters to which predetermined signal characteristics are given. This signal processing block corresponds to the noise cancellation signal processing unit 6A in each path when it is multipathed.
In these signal processing blocks 92-0 to 92-6, the sampling frequency = 32 fs, 16 fs, 8 fs, 4 fs output from the downsampling circuits 91-1 to 91-6, respectively. 2fs and 1fs signals are input. The signal processing blocks 92-0 to 92-6 each receive these signals and output them at the same sampling frequency (and the number of quantization bits) as the input.
The upsampling circuits 94-1 to 94-6 upsample the sampling frequency of the input signal by a factor of 2 and output it. The up-sampling circuit 94-1～94-5, then 32fs from synthesizer 93-1～93-5 described, 16 fs, 8fs, so that the 4fs, signal 2fs is input. The upsampling circuit 94-6 receives a 1 fs signal from the signal processing block 92-6.
The synthesizers 93-0 to 93-5 receive the 64fs, 32fs, 16fs, 8fs, 4fs, and 2fs signals output from the signal processing filters 92-0 to 92-5, respectively, and the upsampling circuit 94- The signals of 64 fs, 32 fs, 16 fs, 8 fs, 4 fs, and 2 fs output from 1 to 94-6 are input and synthesized. The outputs of the combiners 93-1 to 93-5 are input to the upsampling circuits 94-1 to 94-5. The output of the synthesizer 93-0 becomes the final output signal by 64fs.
In order to actually perform multipathing for the second noise cancellation signal processing system, it is necessary to obtain a system with a required sampling frequency based on the configuration shown in FIG. a down-sampling circuit, an up sampling circuit, upon configuring the combiner, in each system, is adapted to constitute a signal processing block (noise cancellation signal processing section) as appropriate signal processing is performed.

Moreover, as an embodiment, the anti-imaging filter 7b in the decimation filter 5B (5D) and the interpolation filter 7 is implemented with a minimum phase type FIR or IIR filter, thereby suppressing the phase rotation more effectively. However, the digital filter used in these functional circuit units has a delay time that is small enough to satisfy the required noise cancellation effect. For example, other conditions such as sound quality and stability satisfy a certain level. As long as this is the case, a configuration other than the minimum phase transition FIR and IIR filter is also conceivable.
Further, under the present invention, the minimum phase transition type FIR or IIR filter may be configured to be employed only in at least one of the decimation filter 5B (5D) and the anti-imaging filter 7b. . Even in such a configuration, for example, the delay of the signal processing system for noise cancellation is less than that in the case where both the decimation filter 5B (5D) and the anti-imaging filter 7b adopt the linear phase type. Therefore, a corresponding effect is expected.

In addition, each part that realizes the noise canceling system according to the present embodiment is actually mounted on the apparatus. In this regard, the noise canceling based on the present invention is actually performed. It may be arbitrarily determined appropriately depending on the device to which the system is applied, the configuration of the system, the usage, and the like.
For example, if a single headphone device having a noise canceling function is to be configured, almost all components (that is, LSI 600) that are supposed to form a noise canceling system are accommodated in the housing of the headphone device. Can be implemented. Alternatively, if a noise canceling system is to be configured by a set of devices such as a headphone device and an external adapter, the LSI 600 may be mounted on the adapter side. Further, a configuration in which the functional circuit unit in the LSI 600 is divided into a plurality of parts and at least one of these is mounted on the adapter side is also conceivable.
In addition, the noise canceling system according to the present invention is not used for a headphone device or the like, but for an audio playback device configured to play audio content and output it to a headphone terminal, a mobile phone device, a network voice communication device, or the like. Even if it is decided to mount, at least one of components other than the microphone and driver may be mounted on these devices.

  Further, the present invention is configured such that required digital signal processing corresponding to the same functional purpose is performed by a plurality of signal processing systems each corresponding to a different sampling frequency, and expecting some operational effects obtained thereby. It can be said that. From this point of view, the present invention is not limited to the function purpose for noise cancellation of the function corresponding signal processing means (procedure), and performs signal processing corresponding to another function purpose. It may be configured as follows.

It is a figure which shows the model example about the noise cancellation system of the headphone apparatus by a feedback system. It is a Bode diagram which shows the characteristic about the noise canceling system shown in FIG. It is a figure which shows the model example about the noise canceling system of the headphone apparatus by a feedforward system. It is a block diagram which shows the basic structural example of the noise cancellation system of the headphone apparatus by a digital system. It is a figure which shows the structure of the dual path | pass which the noise canceling system as this Embodiment takes as compared with the structure of a single path | pass. It is a block diagram which shows the structural example of the noise canceling system as 1st Embodiment in this invention. As an example of the first functional aspect in the present embodiment, frequency band setting examples for the noise cancellation signal processing unit of the first noise cancellation signal processing system and the noise cancellation signal processing unit of the second noise cancellation signal processing system FIG. As an example of the second functional mode in the present embodiment, frequency band setting examples for the noise cancellation signal processing unit of the first noise cancellation signal processing system and the noise cancellation signal processing unit of the second noise cancellation signal processing system FIG. It is a figure which shows the example of a connection aspect in the case of comprising by 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 by 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 by 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 by 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 by 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 by 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 by 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 mode shown in FIG. It is a block diagram which shows the structural example of the noise canceling system as 2nd Embodiment in this invention. It is a block diagram which shows the structural example of the noise canceling system as 3rd Embodiment in this invention. It is a block diagram which shows the structural example of the noise canceling system as 4th Embodiment in this invention. It is a block diagram which shows the structural example 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 path configuration based on FIG. 4 and a noise canceling system with a dual path configuration based on FIG. 6. It is a block diagram which shows the example of a model of the signal processing system used as the basis of a multipath structure.

Explanation of symbols

1a driver, 1c headphone unit, 2F and 2B microphone, 3 and 3A amplifier, 4 and 4A delta-sigma modulator, decimation filter 5 and 5-1 (5A, 5B, 5C and 5D), noise cancellation signal processing unit 6 and 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 processor, SW1, SW11 switch, 600 LSI, 700 analog block, 800 digital block, 810 1st Signal processing chip, 820 second signal processing chip

Claims (8)

Sampling represented by n × fs (where n is a natural number) by inputting a digital signal of the first format with a predetermined quantization bit number of 1 bit or more obtained by ΔΣ modulation processing, with a predetermined reference sampling frequency as fs First decimation processing means for generating and outputting a digital signal of a second format to be a pulse code modulation signal by frequency;
A digital signal of the second format output from the first decimation processing means is input, and a pulse code modulation signal with a sampling frequency represented by m × fs (m is a natural number and m <n) is used. Second decimation processing means for generating and outputting a digital signal of a third format having a format;
A third type digital signal output from the second decimation processing means is input, predetermined signal processing according to a predetermined functional purpose is executed, and output is performed in the same third format. 1 function corresponding signal processing means;
Interpolation processing means adapted to convert a third format signal output from the first function-compatible signal processing means into a second format and output the second format signal;
A second type of digital signal output from the first decimation processing means is input, predetermined signal processing is executed according to the functional purpose, and the second format is output in the same second format. Functional signal processing means of
At least the second format digital signal output from the second function corresponding signal processing means and the second format digital signal output from the interpolation processing means are combined at least, Combining means for outputting to an input stage for digital-analog conversion processing;
With
The first function corresponding signal processing means as the predetermined signal processing according to the functional purpose, a predetermined frequency band below that are midrange and low or cancellation of a predetermined frequency band below that is low It is configured to perform signal processing for giving a cancel signal characteristic for canceling the target sound,
When the filter function is set so that the first function-compatible signal processing means gives a signal characteristic for canceling a cancellation target sound component in a frequency band below a predetermined frequency range that is a middle range and a low range. In
The second function corresponding signal processing means sets a filter characteristic so as to give a signal characteristic for canceling a cancel target sound component in a frequency band higher than the middle band and the low band. ,
The second function corresponding signal processing means is constituted by a digital filter of a linear phase type finite impulse response system, or
When the filter function is set so that the first function corresponding signal processing means gives a cancel signal characteristic for canceling a cancel target sound component in a predetermined frequency band which is a low frequency or lower. ,
The second function corresponding signal processing means sets a filter characteristic so as to give a signal characteristic for canceling a cancel target sound component in a frequency band higher than the low frequency band,
The second function-compatible signal processing means is constituted by a digital filter of an infinite impulse response system. The first function corresponding signal processing means is obtained as a digital signal processing function executed in the digital signal processor by programming the digital signal processor.
The signal processing apparatus according to claim 1. The first function corresponding signal processing means forms the first function corresponding signal processing means according to an analysis result obtained by branching and inputting a digital signal to be input by the first function corresponding signal processing means and executing a predetermined analysis process. The digital filter to be formed, the digital filter to form the second function corresponding signal processing means, the digital filter to form the second decimation processing means, and the interpolation processing means are formed. An analysis means for changing and setting a filter characteristic for at least one of the digital filters;
The signal processing apparatus according to claim 1. The digital signal of the first format is obtained by performing the above-described ΔΣ modulation processing on a signal obtained by collecting a sound with a microphone provided to correspond to a noise canceling system of a headphone device using a feedforward method. Is,
The signal processing apparatus according to claim 1. The digital signal of the first format is obtained by performing the above-described ΔΣ modulation processing on a signal obtained by collecting a sound with a microphone provided to be compatible with a noise canceling system of a headphone device using a feedback method. Is,
The signal processing apparatus according to claim 1.   The signal processing apparatus according to claim 1, wherein the signal processing apparatus is provided in one chip. The signal processing device is composed of a plurality of chips. The first decimation processing means, the second decimation processing means, the first function-compatible signal processing means, the interpolation processing means, 2. The function-corresponding signal processing means and the synthesizing means are each formed in a predetermined chip of the plurality of chips based on a sampling frequency of a digital signal input / output by itself. Signal processing device. Sampling represented by n × fs (where n is a natural number) by inputting a digital signal of the first format with a predetermined quantization bit number of 1 bit or more obtained by ΔΣ modulation processing, with a predetermined reference sampling frequency as fs A first decimation processing procedure for generating and outputting a digital signal of a second format to be a pulse code modulated signal according to frequency;
A digital signal of the second format output by the first decimation processing procedure is input, and a pulse code modulation signal with a sampling frequency represented by m × fs (m is a natural number and m <n) is used. A second decimation processing procedure for generating and outputting a digital signal of a third format having a format;
The digital signal of the third format output by the second decimation processing procedure is input, the predetermined signal processing according to the predetermined functional purpose is executed, and the same third format is output. A first function-compatible signal processing procedure;
An interpolation processing procedure adapted to convert the signal of the third format output by the first function-corresponding signal processing procedure into a second format and output it;
The second format digital signal output by the first decimation processing procedure is input , predetermined signal processing is executed according to the functional purpose, and the second format is output in the same second format. 2 function compatible signal processing procedures;
At least the second format digital signal output by the second function corresponding signal processing procedure and the second format digital signal output by the interpolation processing procedure are combined at least, A synthesis procedure for outputting to an input stage for digital-analog conversion processing;
Run
The Oite the first function corresponding signal processing procedure, as the predetermined signal processing according to the functional purpose, a predetermined frequency band below that are midrange and low or predetermined frequency band below that is low Execute signal processing to give cancellation signal characteristics to cancel the cancellation target sound of
In the first function-compatible signal processing procedure, when the filter characteristic is set so as to give a signal characteristic for canceling a cancellation target sound component in a frequency band equal to or lower than a predetermined frequency band that is a middle band and a low band In
The second function corresponding signal processing means sets a filter characteristic so as to give a signal characteristic for canceling a cancel target sound component in a frequency band higher than the middle band and the low band. ,
The signal processing procedure corresponding to the second function is executed by a digital filter of a linear phase type finite impulse response system, or
In the first function-compatible signal processing procedure, when the filter characteristic is set so as to provide a cancellation signal characteristic for canceling a cancellation target sound component in a predetermined frequency band that is a low frequency or lower. ,
The second function corresponding signal processing means sets a filter characteristic so as to give a signal characteristic for canceling a cancel target sound component in a frequency band higher than the low frequency band,
The second function corresponding signal processing procedure is a signal processing method executed by a digital filter of an infinite impulse response system.

Images