JP5210040B2 - Signal processing device - Google Patents

Description

  The present invention relates to a signal processing apparatus that automatically adjusts the level of an input signal.

  In the input path for recording the moving image audio of a digital video camera or digital still camera, there is a device for converting the input analog audio signal into a digital signal and adjusting the sound quality (hereinafter, signal processing device). Provided.

  FIG. 3 is a diagram illustrating a configuration example of a conventional signal processing apparatus 100. The signal processing apparatus 100 includes an input terminal 102, an analog signal processing block 104, a digital signal processing block 106, an output terminal 108, and an ALC (Auto Level Controller) 110.

  An analog audio signal input to the input terminal 102 via a microphone (not shown) is input to the analog signal processing block 104. The analog signal processing block 104 includes a PGA (Proguramable Gain Amplifire) 112 and an ADC (Analog-to-Digital Converter) 114 which are input amplifiers. The analog audio signal input to the analog signal processing block 104 is first amplified by the PGA 112, converted to a digital audio signal by the ADC 114, and input to the subsequent digital signal processing block 106.

  The PGA 112 is an externally controllable amplifier, and its gain can be set to an arbitrary value. Here, the gain of the PGA 112 is set to a value corresponding to the control signal from the ALC 110, and the input analog audio signal is amplified with the set gain. In recent years, a ΣΔ ADC or the like can be generally used as the ADC 114. In the case of the ΣΔ ADC 114, the output digital audio signal is data of a low bit high sampling rate such as 1 bit and 128 Fs (x128 times sampling rate). Accordingly, in the digital signal processing block 106 at the subsequent stage, it is necessary to first execute a thinning process to reduce (thinning out) the sampling rate. The decimation filter 116 included in the digital signal processing block 106 performs this thinning process.

  The digital signal processing block 106 further includes a DC cut high pass filter (hereinafter referred to as HPF) 118, and after the digital audio signal sampling rate is reduced by the decimation filter 116, the DC cut high pass filter (hereinafter referred to as “HP cut”). HPF) 118 cuts excess DC components generated by digital conversion of ADC 114.

  An ALC 110 is provided downstream of the DC cut HPF 118. The ALC 110 controls the gain of the PGA 110 based on the level of the digital audio signal input from the DC cut HPF 118, and automatically adjusts the input audio level to a certain level. However, if there is no limit and it is constant, there will be no feeling of inflection of the voice, and there will be a sense of incongruity in the sense of hearing, so it is common to apply a certain degree of restriction by setting the Max gain value or the like. The ALC 110 is a circuit that detects the level of the input digital audio signal and performs gain control. The ALC 110 does not perform signal processing on the input digital audio signal, and directly adjusts the sound quality of the digital signal processing block 106. Output to the filter processing unit 120.

  The sound quality adjustment filter processing unit 120 executes filter processing for sound quality adjustment. The sound quality adjustment filter processing unit 120 includes a wind noise removal high pass filter (HPF) 122 and a notch filter 124. The wind noise removal HPF 122 removes the low frequency wind noise input from the microphone. This filter is used at a predetermined cutoff frequency (for example, about 100 Hz to 200 Hz). After removing the wind noise, the notch filter 124 removes noise of a specific frequency generated according to the device to be mounted. The digital audio signal filtered by the sound quality adjustment filter processing unit 120 is output from the output terminal 108 as recorded data.

  In addition to the signal processing apparatus 100 shown in FIG. 3, a signal processing apparatus 140 as shown in FIG. 4 is also known.

  The signal processing device 140 includes an input terminal 142, an analog signal processing block 144, a digital signal processing block 146, an ALC 148, and an output terminal 150. The digital signal processing block 146 includes a sound quality adjustment filter processing unit 160, and the ALC 148 is arranged at a subsequent stage of the sound quality adjustment filter processing unit 160.

  The analog audio signal input from the input terminal 142 is input to the analog signal processing block 144. In the analog signal processing block 144, first, an analog signal is amplified by the PGA 152, and the amplified analog audio signal is converted into a digital audio signal by the ADC 154. The function of the PGA 152 is the same as that of the PGA 112 of the signal processing apparatus 100, but the gain of the PGA 152 in FIG. 4 is always fixed.

Subsequently, when the digital audio signal into a digital signal processing block 1 4 6 is input, the decimation performs decimation processing filter 156, to cut a DC component in the DC cut HPF158, wind noise sound quality adjustment filter processing unit 160 Wind noise is removed by the removal HPF 162, and noise is removed by the notch filter 164 of the sound quality adjustment filter processing unit 160.

  Thereafter, the digital audio signal is input to the ALC 148 for level adjustment. The ALC 148 has an amplification function, sets the gain of the amplification function in accordance with the level of the input digital audio signal, amplifies the input digital audio signal by multiplying the set gain, and performs level adjustment . The digital audio signal is level-adjusted by ALC 148 and then output as recording data from output terminal 150.

  In the case of the configuration of FIG. 3, the gain of the PGA 112 is controlled by the ALC 110, and the ALC 110 is inserted before the sound quality adjustment filter processing unit 120 of the digital signal processing block 106. Filter processing in the adjustment filter processing unit 120 is performed. Therefore, for example, when a sound with a very low level is input, the low frequency part is cut by the wind noise removal HPF 122 after the level adjustment, and the volume of the final recording data is reduced. There is a problem of becoming.

  On the other hand, in the case of the configuration of FIG. 4, the gain of the PGA 112 is always fixed, and the level adjustment is performed after the filter processing in the sound quality adjustment filter processing unit 160. . That is, even when the level is reduced by the wind noise removal HPF 162 or the like, the gain is increased by the subsequent processing in the ALC 148, and the volume is kept constant to some extent. However, in the case of the configuration of FIG. 4, there is no level adjusting means for the analog signal processing block 144. Therefore, when an excessive input that exceeds the full scale level of the ADC 154 is made as a microphone input, the ADC 154 There is a problem that saturation occurs and sound quality deteriorates.

  As described above, the configurations shown in FIGS. 3 and 4 have advantages and disadvantages, respectively, and cannot cope with all cases. Therefore, in any case, an apparatus is desired in which the level of a digital signal to be output is constant without deterioration in sound quality.

Various techniques have been proposed as gain control techniques for input signals. For example, a voice input method has been proposed in which gain correction is performed for each frequency band of the signal collected by the sound collecting means (see Patent Document 1 below). Also, in a system that performs analog-to-digital conversion after high-frequency enhancement emphasis is applied to the analog input signal, the setting gain of the analog level adjuster that adjusts the level of the analog input signal is a value corresponding to the level of the signal before emphasis. There is also proposed an auto gain control device that is set based on the amount of the signal after the emphasis exceeds the predetermined level (see Patent Document 2 below).
JP 2001-94370 A JP-A-9-148862

  However, the method described in Patent Document 1 has a problem that since the gain is adjusted for each frequency, the processing becomes complicated and the circuit scale increases.

  In addition, since the apparatus described in Patent Document 2 only performs gain adjustment on the analog side after all, when performing filter processing for sound quality adjustment that can change the signal level after digital conversion, No matter how much the gain is adjusted on the analog side, a constant level is not always guaranteed for the output digital audio signal after the filtering process.

  The present invention has been proposed to solve the above-described problems, and is a signal processing apparatus capable of keeping the level of an output digital audio signal constant without degrading sound quality with a simple circuit configuration. The purpose is to provide.

In order to achieve the above object, a signal processing apparatus according to claim 1 of the present invention amplifies the level of an analog audio signal input from the outside with a set gain, and is amplified by the amplification means. Conversion means for converting the analog audio signal into a digital audio signal, sound quality adjustment processing means for performing signal processing for sound quality adjustment on the digital audio signal, and digital audio before signal processing by the sound quality adjustment processing means Setting means for detecting a signal level and setting the gain of the amplification means based on the detected level so that an analog audio signal input to the amplification means is amplified to a predetermined level by the amplification means; , the level of the signal preprocessing of the digital audio signal by the audio quality adjustment processing unit, the digital audio signal after the signal processing by the audio quality adjustment processing unit Detecting a difference between the level, with a gain corresponding to the difference of the detected, and a, a correction means for performing amplifying and level correction digital audio signal after the signal processing by said sound quality adjustment processing unit.

  When signal processing for sound quality adjustment is performed, the level of the digital audio signal may fluctuate. Therefore, as in the above invention, the signal processing is performed with a gain corresponding to the difference between the level of the digital audio signal before the signal processing for sound quality adjustment and the level of the digital audio signal after the signal processing. By amplifying the digital audio signal and performing level correction, the level fluctuation is compensated for, and a constant level of output is always obtained, thereby preventing the overall volume reduction caused by sound quality adjustment signal processing, etc. can do.

According to a second aspect of the present invention, there is provided a signal processing apparatus comprising: amplifying means for amplifying the level of an externally input analog audio signal with a set gain; and the analog audio signal amplified by the amplifying means is digitally converted. Conversion means for converting into an audio signal; sound quality adjustment processing means for performing signal processing for sound quality adjustment on the digital audio signal; and detecting the level of the digital audio signal before signal processing by the sound quality adjustment processing means. When a level equal to or higher than the first level is detected, the gain setting value of the amplifying unit is set so that the level of the digital audio signal before the signal processing by the sound quality adjusting processing unit is less than the first level. lowered, when detecting a level below the first level after setting value reduced, the setting means to restore the gain setting value of said amplifier means, said Detecting a level of the digital audio signal after the signal processing by the quality adjustment processing unit, said digital audio signal by amplifying the digital audio signal after the signal processing by said sound quality adjustment processing means with a gain corresponding to the level of the detected that the Correction means for performing level correction so as to be level 2.

  As described above, after the signal processing for sound quality adjustment, level correction is performed so that the digital audio signal becomes the second level (for example, a predetermined level suitable as a recording level), and the sound quality adjustment is performed. If the level of the digital audio signal before the signal processing is detected and the detected level is equal to or higher than the first level (for example, a level slightly smaller than the full scale level of the converting means), the gain of the amplifying means is increased. When the set value is lowered and a level lower than the first level is detected after the set value is lowered, the gain set value of the amplifying means is returned to the original value, so that a constant level output is always obtained. It is possible to reduce the occurrence of saturation of the conversion means, and to prevent deterioration of sound quality.

  As described above, according to the present invention, there is an effect that the level of an output digital signal can be kept constant with a simple circuit configuration without deteriorating sound quality.

  Hereinafter, preferred embodiments of the present invention will be described in detail with reference to the drawings.

  [First Embodiment]

  FIG. 1 is a diagram illustrating a configuration example of a signal processing device 10 according to the first embodiment. A signal processing apparatus 10 according to the present embodiment is provided in a recording path for recording input sound, and includes an input terminal 12, an analog signal processing block 14, a digital signal processing block 16, an ALC (Auto Level Controller) 18, and a level difference. A detector 20, a correction gain device 22, and an output terminal 24 are provided.

  An analog audio signal input to the input terminal 12 via a microphone (not shown) is input to the analog signal processing block 14. The analog signal processing block 14 includes a PGA (Proguramable Gain Amplifire) 32 and an ADC (Analog-to-Digital Converter) 34 which are input amplifiers. The analog audio signal input to the analog signal processing block 14 is first amplified by the PGA 32, converted into a digital signal by the ADC 34, and input to the digital signal processing block 16 at the subsequent stage.

  The PGA 32 is an externally controllable amplifier, and its gain can be set to an arbitrary value. Here, the gain of the PGA 32 is set to a value corresponding to the control signal from the ALC 18, and the PGA 32 amplifies the input analog audio signal with the set gain. In the present embodiment, a ΣΔ ADC is applied as the ADC 34. The ΣΔ ADC 34 outputs a digital audio signal at a low bit high sampling rate such as 1 bit, 128 Fs (x128 times sampling rate), for example.

  The digital audio signal converted by the ADC 34 is input to the digital signal processing block 16. The digital signal processing block 16 includes a decimation filter 36, a DC cut high pass filter (hereinafter, HPF) 38, and a sound quality adjustment filter processing unit 40.

  The decimation filter 36 performs a thinning process on the digital audio signal input from the ADC 34, reduces the sampling rate, and inputs it to the DC cut HPF 38. As described above, since a digital audio signal with a low bit and high sampling rate is input from the ΣΔ ADC 34, first, a thinning process is executed here to lower the sampling rate (thinning out). When the thinned digital audio signal is input, the DC cut HPF 38 cuts an excess DC component generated by digital conversion in the ADC 114.

  An ALC 18 is provided at the subsequent stage of the DC cut HPF 38. The ALC 18 detects the level of the digital audio signal input from the DC cut HPF 38, and based on the detected level, adjusts the gain of the PGA 32 so that the analog audio signal input to the PGA 32 is amplified to a certain level. Set. As a result, the input sound level is automatically adjusted to a constant level. However, if it is made unlimited and constant, there will be no feeling of inflection of the voice, and there will be a sense of incongruity in the sense of hearing. In this embodiment, a certain degree of restriction is imposed by setting the Max gain value or the like. The ALC 18 is a control circuit that detects the level of the digital audio signal and outputs a control signal to the PGA 32. The ALC 18 does not perform any signal processing on the input digital audio signal, and converts the input digital audio signal to the subsequent stage. To the sound quality adjustment filter processing unit 40 and the level difference detector 20.

  The sound quality adjustment filter processing unit 40 executes filter processing for sound quality adjustment. The sound quality adjustment filter processing unit 40 includes a wind noise removal high pass filter (HPF) 42 and a notch filter 44. The wind noise removal HPF 42 removes the low frequency wind noise component input from the microphone. This filter is used at a predetermined cutoff frequency (for example, about 100 Hz to 200 Hz). After removing the wind noise, the notch filter 44 removes noise of a specific frequency generated according to the device to be mounted.

  The digital audio signal whose sound quality has been adjusted by the sound quality adjustment filter processing unit 40 is input to the level difference detector 20 and the correction gain device 22 provided in the subsequent stage.

  The level difference detector 20 receives the digital audio signal before the filter processing of the sound quality adjustment filter processing unit 40 and the digital audio signal after the filter processing of the sound quality adjustment filter processing unit 40. The level difference detector 20 detects the difference between the levels (amplitude, power, etc.) of the two input digital audio signals, and outputs the detected difference to the correction gain unit 22 as a correction gain value. For example, if the value obtained by subtracting the level after filtering from the level before filtering is 6 dB, this is output to the correction gain unit 22 as a correction gain value.

  The correction gain unit 22 is supplied with a digital audio signal whose sound quality has been adjusted from the notch filter 44 and a correction gain value from the level difference detector 20. The correction gain unit 22 amplifies the input digital audio signal so as to have a predetermined level according to the input correction gain value. As a result, the gain can be increased and amplified by the level attenuated by the sound quality adjustment filter processing unit 40, so that it is possible to output at a level before the sound quality adjustment filter processing unit 40 performs the filter process.

  After level correction by the correction gain unit 22, the digital audio signal after level correction is output from the output terminal 24 as recorded data.

  As described above, in the present embodiment, the correction gain unit 22 corrects only the level attenuated by the sound quality adjustment filter processing unit 40 (wind noise removal HPF 42 and notch filter 44). An output can be obtained, and for example, it is possible to prevent a decrease in overall sound volume due to, for example, cut of reduced sound generated in a conventional apparatus.

  [Second Embodiment]

  FIG. 2 is a diagram illustrating a configuration example of the signal processing device 50 according to the second embodiment. The signal processing apparatus 50 according to the present embodiment is provided in a recording path for recording input voice, and includes an input terminal 52, an analog signal processing block 54, a digital signal processing block 56, a peak limiter 58, an ALC 60, and an output terminal 62. It has.

  An analog audio signal input to the input terminal 52 via a microphone (not shown) is input to the analog signal processing block 54. The analog signal processing block 54 includes a PGA 72 and an ADC 74. The functions of the PGA 72 and the ADC 74 are the same as those of the PGA 32 and the ADC 34 of the first embodiment, but the gain of the PGA 72 is set and controlled by the peak limiter 58 instead of the ALC.

  Similar to the digital signal processing block 16 of the signal processing apparatus 10 of the first embodiment, the digital signal processing block 56 provided at the subsequent stage of the analog signal processing block 54 includes a decimation filter 76, a DC cut HPF 78, and A sound quality adjustment filter processing unit 80 is provided. Since the functions of the decimation filter 76 and the DC cut HPF 78 are the same as those in the first embodiment, the description thereof is omitted. Similarly to the sound quality adjustment filter processing unit 40 of the first embodiment, the sound quality adjustment filter processing unit 80 also includes a wind noise removal HPF 82 and a notch filter 84, and is the same as in the first embodiment. Remove wind noise and noise.

  In the present embodiment, an ALC 60 is provided after the sound quality adjustment filter processing unit 80. The ALC 60 has an amplification function, sets the gain of the amplification function according to the level of the input digital audio signal, amplifies the input digital audio signal by multiplying the set gain, and always has a constant level. The level is adjusted so that the output is obtained. The digital audio signal is level-adjusted by the ALC 60 and then output as recording data from the output terminal 62.

  In the present embodiment, a peak limiter 58 is provided before the sound quality adjustment filter processing unit 80.

  The peak limiter 58 receives a digital audio signal that has been DC cut by the DC cut HPF 78. The peak limiter 58 has a function of detecting the level of the input digital audio signal. When an excessive digital audio signal that is likely to exceed the full-scale level of the ADC 74 is input, the gain of the PGA 72 is lowered to reduce the ADC 74. This module suppresses the occurrence of saturation. The peak limiter 58 detects the signal level and controls the gain, but does not perform any signal processing on the input digital audio signal and outputs it to the subsequent sound quality adjustment filter processing unit 80.

  Here, the operation of the peak limiter 58 will be specifically described. First, in the peak limiter 58, a value slightly smaller than the full scale level of the ADC 74 (for example, about 90% of the full scale level) is set in advance as a threshold value. The peak limiter 58 is enabled when a digital audio signal having a level equal to or higher than the set threshold is detected, and the gain of the PGA 72 is set so that the input digital audio signal has a level lower than the threshold. Decrease the set value. As a result, the analog audio signal input to the ADC 74 can be reduced to a level at which no saturation occurs in the ADC 74.

  If the peak limiter 58 detects a level less than the threshold value after reducing the set value, the peak limiter 58 returns the set value of the gain of the PGA 72 to the original value. In addition, after the set value is returned to the original value, it is designed to be invalidated until a digital audio signal having a level equal to or higher than the threshold value is input.

  With such a configuration, the peak limiter 58 is invalid while a normal level digital audio signal equal to or less than the threshold set in the peak limiter 58 is being recorded, so the digital audio signal passes through the recording path as it is. Then, after being filtered by the sound quality adjustment filter processing unit 80, it is adjusted to an appropriate level by the last ALC 60 and output (usually, the ALC 60 is amplified by + gain).

  On the other hand, when an excessively high level audio signal that causes saturation in the ADC 74 is input, the peak limiter 58 becomes effective, and the set value of the gain of the PGA 72 is reduced to a level at which saturation is eliminated. When an excessive level digital audio signal is no longer detected by the peak limiter 58 and returns to a normal level where no saturation occurs, the peak limiter 58 returns the gain setting value of the PGA 72 to the original value (increases it to the original value before the decrease). ).

  As described above, in this embodiment, the peak limiter 58 is provided in the previous stage of the sound quality adjustment filter processing unit 80, and the ALC 60 for correcting the level of the digital audio signal is provided in the subsequent stage of the sound quality adjustment filter processing unit 80. As a result, it is possible to reduce the occurrence of saturation of the ADC 74, and as a result, it is possible to improve the sound quality even when an excessive level audio signal is input.

  The present invention is not limited to the above-described embodiments, and various design changes can be made within the scope of the invention described in the claims.

  For example, in the gain control of the correction gain device 22 in the first embodiment or the gain control in the peak limiter 58 or the ALC 60 in the second embodiment, the attack is not changed to the target set value at once. The time (here, the time from the start of gain change until it reaches the target value), the decay time (here, the time required to return to the original value after changing the gain), etc. It may be set to increase or decrease the gain. As a result, the sense of incongruity caused by the increase or decrease in gain can be alleviated.

It is a figure which shows the structural example of the signal processing apparatus of 1st Embodiment. It is a figure which shows the structural example of the signal processing apparatus of 2nd Embodiment. It is a figure which shows the structural example of the conventional signal processing apparatus. It is a figure which shows the structural example of the other conventional signal processing apparatus.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 Signal processing apparatus 12 Input terminal 14 Analog signal processing block 16 Digital signal processing block 20 Level difference detector 22 Correction gain device 24 Output terminal 32 PGA
34 ADC
36 Decimation filter 38 DC cut HPF
40 Sound quality adjustment filter processing unit 42 HPF for removing wind noise
44 notch filter 50 signal processing device 52 input terminal 54 analog signal processing block 56 digital signal processing block 58 peak limiter 62 output terminal 72 PGA
74 ADC
76 Decimation filter 78 DC cut HPF
80 Sound quality adjustment filter processing unit 82 HPF for removing wind noise
84 Notch filter

Claims (2)

Amplifying means for amplifying the level of the analog audio signal input from the outside with a set gain;
Conversion means for converting the analog voice signal amplified by the amplification means into a digital voice signal;
Sound quality adjustment processing means for performing signal processing for sound quality adjustment on the digital audio signal;
The level of the digital audio signal before the signal processing by the sound quality adjustment processing means is detected, and based on the detected level, the analog audio signal input to the amplifying means is amplified to a predetermined level by the amplifying means. Setting means for setting the gain of the amplifying means;
The difference between the level of the digital audio signal before the signal processing by the sound quality adjustment processing means and the level of the digital audio signal after the signal processing by the sound quality adjustment processing means is detected, and with the gain corresponding to the detected difference, Correction means for amplifying the digital audio signal after signal processing by the sound quality adjustment processing means and performing level correction;
A signal processing apparatus comprising: Amplifying means for amplifying the level of the analog audio signal input from the outside with a set gain;
Conversion means for converting the analog voice signal amplified by the amplification means into a digital voice signal;
Sound quality adjustment processing means for performing signal processing for sound quality adjustment on the digital audio signal;
When the level of the digital audio signal before the signal processing by the sound quality adjustment processing means is detected and a level equal to or higher than the first level is detected, the level of the digital audio signal before the signal processing by the sound quality adjustment processing means is When the set value of the gain of the amplifying unit is lowered so as to be less than the first level, and the level less than the first level is detected after the set value is lowered, the set value of the gain of the amplifying unit is set to A setting means to restore,
The level of the digital audio signal after the signal processing by the sound quality adjustment processing means is detected, the digital audio signal after the signal processing by the sound quality adjustment processing means is amplified by a gain according to the detected level, and the digital audio signal is Correction means for performing level correction so as to be the second level;
A signal processing apparatus comprising:

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