audio equipment

Abstract

This invention provides an audio device that can interpolate data surpluses and deficits caused by errors in the sampling frequency of the digital sound source, thereby preventing the generation of noise caused by data surpluses and deficits. [Solution] The processor of the audio equipment performs the following steps: count the number of data D1 to D9 stored in RAM per unit time determined by its sampling frequency Dn; calculate the difference n between the number Dn and the number of samplings Sn performed per unit time; if the difference n is not "0", calculate the same number of interpolated data S1 to S10 as the number of samplings Sn based on the data D1 to D9 corresponding to the number Dn; and transmit the same number of interpolated data S1 to S10 as the number of samplings Sn to the D / A converter in time series.

Description

Technical Field

[0001] The present invention relates to an acoustic apparatus configured to process data of digital audio signals continuously input from a digital device, convert the processed data into analog audio signals and output the same.

Background Art

[0002] At present, it has become common to produce music and audio content using personal computers. Producing music using a personal computer is called desktop music (DTM). In addition, podcasts are widely known as social media for distributing audio content. To produce music or audio content using a personal computer, an audio interface is required to convert analog audio signals into digital signals and input them into the personal computer. Furthermore, software called a digital audio workstation (DAW) is required for editing music or audio content on a personal computer.

[0003] On the other hand, since the outbreak of the novel coronavirus disease (COVID-19) at the end of 2019, the introduction of remote work has progressed among many private companies, and communication between people has come to be carried out through online conference systems. With the expanding use of online conference systems, microphones equipped with USB terminals have become widely popular among the general public. A microphone with a USB terminal is configured to convert the analog signal of the collected sound into a digital signal and output it from the USB terminal. Therefore, a microphone equipped with a USB terminal can input digital audio signals to the USB terminal of a personal computer without an intervening audio interface.

Prior Art Literature

Patent Literature

[0004] [Patent Document 1] Japanese Patent Publication No. 2005-39531 [Patent Document 2] Japanese Patent Publication No. 2006-332923 [Overview of the project] [Problems that the invention aims to solve]

[0005] • Desktop music issues Creating music and audio content using a personal computer requires an audio interface and a digital audio workstation. Of these, digital audio workstations are expensive and require a certain level of musical knowledge from professionals and amateurs to use satisfactorily. Therefore, digital audio workstations hinder the general public, who lack sufficient musical knowledge, from creating music and audio content using personal computers. For example, while microphones with USB ports have become widespread among the general public, they are not used for creating music or audio content using personal computers.

[0006] • Problem with sampling frequency error Let's assume that audio equipment has been developed to create music and audio content, serving as an alternative to personal computers and digital audio workstations. In this case, a problem arises: a sampling frequency error occurs between the audio equipment and the digital sound source. For example, let's assume the audio equipment is a recorder and the digital sound source is a microphone with a USB port. Let's assume that the nominal sampling frequencies of both the recorder and the microphone are 48,000 Hz. The microphone performs A / D conversion 48,000 times per second and outputs a digital sound signal. On the other hand, the recorder performs D / A conversion 48,000 times per second and outputs an analog sound signal. The sampling frequency is determined by the clock frequency of the oscillators in each of the recorder and the microphone. An extremely small error, expressed in units of PPM (parts per million), occurs in the oscillator's clock frequency. The error in the clock frequency directly translates to an error in the sampling frequency. The sampling frequency error between the recorder and the microphone accumulates and increases with each repeated sampling. When sampling frequency errors accumulate and reach an integer multiple of the recorder's sampling period Δt, the recorder will experience over- or under-data conversion. This over- or under-data conversion generates noise. If sampling continues for a long period, the sampling frequency errors in the recorder will periodically accumulate to an integer multiple of the sampling period Δt, resulting in periodic noise generation. Audio equipment that periodically generates noise is not practically worth implementing.

[0007] Such sampling frequency errors can occur between multiple digital devices, including audio equipment. Therefore, when two or more digital devices are connected to audio equipment, the audio equipment needs to eliminate the sampling frequency errors of each of the two or more digital devices to prevent the generation of noise.

[0008] • Objectives of the present invention The present invention has been made in view of the above problems, and aims to provide an audio device that can interpolate data surpluses and deficits caused by errors in the sampling frequency of digital devices that serve as sound sources, can prevent the generation of noise caused by data surpluses and deficits, and can produce music and audio content using multiple digital devices. [Means for solving the problem]

[0009] (1) To achieve the above objective, the present invention provides an audio device configured to process digital audio signal data continuously input from a digital device, convert the processed data into an analog audio signal and output it, comprising: at least one USB terminal for inputting the digital signal; a first memory for storing the digital signal data input from the USB terminal in chronological order; a processor for processing the data stored in the first memory; a D / A converter for converting the processed data into an analog signal; and a device for outputting the analog signal. The processor comprises at least one output terminal, and performs the following steps: count the number of data items (Dn) stored in the first memory per unit time determined by its sampling frequency; calculate the difference (n) between the number (Dn) and the number of samplings (Sn) performed per unit time; if the difference (n) is not "0", calculate the same number of interpolated data items as the number of samplings (Sn) based on the data corresponding to the number (Dn); and transmit the same number of interpolated data items as the number of samplings (Sn) to the D / A converter in time series.

[0010] (2) Preferably, in the audio equipment described in (1) above, the processor calculates at least one interpolated data by performing calculations based on two pieces of data stored in the first memory that are in an orderly fashion.

[0011] (3) Preferably, the above (2) further comprises an inverse characteristic filter used for processing by the processor, wherein the inverse characteristic filter is an arithmetic formula for giving a frequency characteristic that is the opposite of the known frequency characteristic shown by the same number of interpolated data as the number of times (Sn), and the processor flattens the known frequency characteristic by performing an arithmetic process using the inverse characteristic filter on the same number of interpolated data as the number of times (Sn).

[0012] (4) Preferably, in the audio equipment of (1) above, the processor performs the steps of: calculating a time-axis waveform based on at least the same number of data and / or interpolated data as the number of times (Sn); converting the time-axis waveform to a frequency-axis waveform by using a fast Fourier transform; calculating an envelope showing the variation of the frequency-axis waveform; identifying one or more amplitude peaks included in the frequency-axis waveform based on the frequency-axis waveform and the envelope; setting a threshold at a level lower than the amplitude peaks based on the envelope; attenuating frequency components below the threshold included in the frequency-axis waveform; re-converting the frequency-axis waveform, from which the frequency components below the threshold have been attenuated, back to the time-axis waveform by using an inverse fast Fourier transform; and generating at least the same number of data and / or interpolated data as the number of times (Sn) based on the re-converted time-axis waveform.

[0013] (5) Preferably, the audio equipment of (1) above is provided with a transmission path corresponding to the number of USB terminals, the USB terminals are electrically connected to the transmission path, and a mute switch, a first variable resistor, a second variable resistor, a second memory, the processor, and the D / A converter are electrically connected to the transmission path, the mute switch is used to set the volume of the analog signal output from the output terminal to zero, the first variable resistor is used to adjust the volume of the digital signal input from the USB terminal, the second variable resistor is used to adjust the volume of the analog signal output from the output terminal, and the second memory is configured to store the data processed by the processor.

[0014] (6) Preferably, in the audio equipment described in (5) above, a level indicator is electrically connected to the transmission line, and the level indicator is configured to visually display the volume of the digital signal input from the USB terminal. [Effects of the Invention]

[0015] The audio device of the present invention, when a difference occurs between the number of data points per unit time (Dn) and the number of sampling cycles (Sn), calculates the same number of interpolated data points as the number of sampling cycles (Sn) based on the data corresponding to the number of data points (Dn), and transmits the interpolated data to the D / A converter in time series. This makes it possible to interpolate data surpluses and deficiencies caused by errors in the sampling frequency of the digital sound source device. Furthermore, it is possible to prevent the generation of noise caused by data surpluses and deficiencies. In addition, it becomes possible to produce music and audio content using multiple digital devices. [Brief explanation of the drawing]

[0016] [Figure 1] Figure 1A is a plan view showing a recorder as an embodiment of the audio equipment of the present invention. Figure 1B is a bottom view showing the recorder. [Figure 2]Fig. 2A is a rear view showing the recorder. Fig. 2B is a front view showing the recorder. Fig. 2C is a left side view showing the recorder. Fig. 2D is a right side view showing the recorder. [Figure 3] Fig. 3 is a schematic diagram showing an electrical connection between the recorder and a peripheral device. [Figure 4] Fig. 4 is a block diagram showing an electrical circuit of the recorder. [Figure 5] Fig. 5A is a schematic diagram showing sampling frequencies of the microphone and the recorder. Fig. 5B is a schematic diagram showing a digital audio signal input from the microphone to the recorder. Fig. 5C is a schematic diagram showing excess or deficiency of data caused by a sampling frequency error between the microphone and the recorder. [Figure 6] Fig. 6 is a flowchart showing a flow of data processing by a processor constituting the recorder. [Figure 7] Fig. 7A is a schematic diagram showing interpolation data calculation processing by the processor. Fig. 7B is a schematic diagram for comparing the data and the interpolation data. Fig. 7C is a schematic diagram showing a specific example of the interpolation data calculation method. [Figure 8] Fig. 8A is a schematic diagram showing a time-axis waveform of the interpolation data. Fig. 8B is a schematic diagram showing a frequency-axis waveform of the interpolation data. Fig. 8C is a schematic diagram showing a frequency-axis waveform of an inverse characteristic filter having a frequency characteristic inverse to that of the frequency-axis waveform in Fig. 8B. Fig. 8D is a schematic diagram showing a frequency-axis waveform of the interpolation data whose frequency characteristic is flattened by the inverse characteristic filter. [Figure 9] Fig. 9A is a schematic diagram showing a time-axis waveform of 96000 pieces of digital audio signal data. Fig. 9B is a schematic diagram showing a frequency-axis waveform obtained by performing fast Fourier transform on the time-axis waveform of Fig. 9A, and an envelope calculated based on the frequency-axis waveform. Fig. 9C is a schematic diagram showing attenuation of frequency components equal to or lower than a threshold included in the frequency-axis waveform of Fig. 9B. Fig. 9D is a schematic diagram showing a time-axis waveform obtained by performing inverse fast Fourier transform on the frequency-axis waveform of Fig. 9C. MODE FOR CARRYING OUT THE INVENTION

[0017] Hereinafter, a recorder as an embodiment of the acoustic device of the present invention will be described with reference to the drawings. The recorder described below is an example of the acoustic device of the present invention, and the acoustic device of the present invention is not limited to the form of a recorder.

[0018] 1. Overview of the Recorder FIGS. 1A, 1B and 2A to 2D show an external appearance of the recorder 1 according to the present embodiment. The recorder 1 is used by a user to produce audio content, and is configured to be capable of recording audio content data and distributing the same via the Internet.

[0019] FIG. 3 shows a plurality of peripheral devices electrically connected to the recorder 1. The plurality of peripheral devices are classified into three types: digital devices, analog devices, and power sources. The digital devices include first and second microphones 501, 502, and any one of a personal computer 201, a tablet computer 202, and a smartphone 203. The analog devices include first and second headphones 601, 602. The power sources include a household power outlet 401 and a mobile battery 402. The first and second microphones 501, 502 are used by first and second users to input digital audio signals to the recorder 1. Any one of the personal computer 201, the tablet computer 202, and the smartphone 203 is used for connection to the Internet, for inputting digital audio signals to the recorder 1, and for outputting digital audio signals from the recorder 1. The first and second headphones 601, 602 are used by the first and second users to monitor sound input to the recorder 1.

[0020] 2. External Configuration of the Recorder As shown in Figure 1A, Recorder 1 has a rectangular housing that is elongated vertically when viewed from above. Each of the four corners of the housing is chamfered by a curved surface. As shown in Figures 2C and 2D, the plane of the housing is an inclined surface that slopes downward from top to bottom. As shown in Figure 1A, the inclined plane of the housing is configured with a user interface that includes multiple switches, knobs, and indicators.

[0021] On the upper part of the housing of recorder 1, three large switch groups and five small switch groups are arranged in two rows, horizontally. The three large switches in the upper row, from left to right, are the power switch 101, the recording switch 102, and the play / pause switch 103. The five small switches in the lower row, from left to right, are the first mute switch 111, the tone switch 114, the noise reduction switch 113, the compressor switch 115, and the second mute switch 112. The first mute switch 111 is operated by the user to reduce the volume of the sound input to recorder 1 from the first microphone 501 shown in Figure 3 to zero. For this reason, the first mute switch 111 is located on the left side of the housing's plane to correspond with the first microphone 501, which is located on the left side of recorder 1. The second mute switch 112 is operated by the user to reduce the volume of the sound input to the recorder 1 from the second microphone 502, as shown in Figure 3, to zero. For this reason, the second mute switch 112 is positioned on the right side of the housing plane to correspond with the second microphone 502, which is positioned on the right side of the recorder 1. The tone switch 114, noise reduction switch 113, and compressor switch 115 are operated by the user to apply a common acoustic effect to both the sound input to the recorder 1 from the first and second microphones 501 and 502. For this reason, the tone switch 114, noise reduction switch 113, and compressor switch 115 are all positioned between the first mute switch 111 and the second mute switch 112.

[0022] On the lower part of the housing of recorder 1, from left to right, are the first input volume knob 131, the output volume knob 133, and the second input volume knob 132. The first input volume knob 131 and the second input volume knob 132 are positioned on the left and right sides, at equal distances from each other from the vertical central axis of the housing's plane. The output volume knob 133 is positioned above the vertical central axis of the housing's plane and below the first input volume knob 131 and the second input volume knob 132. The first input volume knob 131 is operated by the user to adjust the volume of the sound input to recorder 1 from the first microphone 501 shown in Figure 3. For this reason, the position of the first input volume knob 131 is positioned on the left side of the housing's plane to correspond with the first microphone 501, which is positioned on the left side of recorder 1. The second input volume knob 132 is operated by the user to adjust the volume of the sound input to recorder 1 from the second microphone 502 shown in Figure 3. Therefore, the second input volume knob 132 is positioned on the right side of the housing's plane to correspond with the second microphone 502, which is located on the right side of the recorder 1. The output volume knob 133 is operated by the user to adjust the volume of both the sound output from the recorder 1 to the first and second headphones 601 and 602. For this reason, the output volume knob 133 is positioned between the first input volume knob 131 and the second input volume knob 132.

[0023] In the center of the housing of recorder 1, a first level indicator 121 and a second level indicator 122 are positioned on the left and right sides with a gap between them. Each of the first level indicator 121 and the second level indicator 122 consists of six LEDs arranged vertically. The first level indicator 121 visually displays the volume of sound input to recorder 1 from the first microphone 501 by turning the six LEDs on and off. For this reason, the first level indicator 121 is positioned on the left side of the housing's plane to correspond with the first microphone 501, which is positioned on the left side of recorder 1. The second level indicator 122 visually displays the volume of sound input to recorder 1 from the second microphone 502 by turning the six LEDs on and off. For this reason, the second level indicator 122 is positioned on the right side of the housing's plane to correspond with the second microphone 502, which is positioned on the right side of recorder 1.

[0024] As shown in Figure 2C, a first USB terminal 21 is provided on the left side of the housing of the recorder 1. The type of the first USB terminal 21 is not particularly limited. For example, a USB "Type-A" can be used as the first USB terminal 21.

[0025] As shown in Figure 2D, a second USB terminal 22 and a microSD card slot 150 are provided on the right side of the housing of the recorder 1. The type of the second USB terminal 22 is not particularly limited. For example, a USB "Type-A" can be used as the second USB terminal 22. Meanwhile, a microSD card 60 (see Figure 4) is inserted into the microSD card slot 150. When the recording switch 102 of the recorder 1 shown in Figure 1A is turned ON, the digital signal data of the sound input to the recorder 1 is stored in the microSD card 60.

[0026] As shown in Figure 2A, third and fourth USB terminals 23 and 24 are provided on the back of the housing of the recorder 1. The type of the third and fourth USB terminals 23 and 24 is not particularly limited. For example, USB "Type-C" can be used as the third and fourth USB terminals 23 and 24.

[0027] As shown in Figure 2B, first and second output terminals 81 and 82 are provided on the front of the housing of the recorder 1. The type of the first and second output terminals 81 and 82 is not particularly limited. For example, stereo mini jacks with a diameter of 3.5 mm and 3 poles can be used as the first and second output terminals 81 and 82.

[0028] As shown in Figure 1B, a battery compartment 140 is provided on the bottom surface of the housing of the recorder 1. In Figure 1B, reference numeral 140 indicates a lid that covers the battery compartment 140. A knob 141 is integrally formed on the lid. The lid can be removed by pushing the knob 141 upwards in the figure. Inside the battery compartment 140, for example, four AA batteries are stored. Batteries that can be used with the recorder 1 include, for example, alkaline batteries, nickel-metal hydride batteries, and lithium batteries.

[0029] 3. Peripheral devices connected to the recorder As shown in Figure 3, the first microphone 501 is electrically connected to the first USB terminal 21 of the recorder 1 via a USB cable 301. The first microphone 501 is used, for example, to capture the voice of a first user and convert it into a digital sound signal. The digital sound signal output from the first microphone 501 is input to the recorder 1 via the first USB terminal 21.

[0030] The second microphone 502 is electrically connected to the second USB terminal 22 of recorder 1 via a USB cable 301. The second microphone 502 is used, for example, to capture the voice of a second user and convert it into a digital sound signal. The digital sound signal output from the second microphone 502 is input to recorder 1 via the second USB terminal 22.

[0031] One of the following is electrically connected to the third USB terminal 23 of recorder 1 via a USB cable 301: a personal computer 201, a tablet computer 202, or a smartphone 203. The personal computer 201, tablet computer 202, and smartphone 203 are used to connect recorder 1 to the internet via the third USB terminal 23. For example, let's assume that personal computer 201 is electrically connected to the third USB terminal 23. Recorder 1 can transmit digital audio signals input from the first and second microphones 501 and 502, and digital audio signal data stored on the microSD card 60, to the internet via personal computer 201. Recorder 1 can also input digital audio signals provided by a third user via the internet and personal computer 201. Furthermore, recorder 1 can input digital audio signals of music and sound effects downloaded to personal computer 201 via the third USB terminal 23. The digital signals of music and sound effects are mixed with, for example, the digital signals of sound input from the first and second microphones 501 and 502.

[0032] The fourth USB terminal 24 of recorder 1 is electrically connected to a household power outlet 401 via a power cable 302. Alternatively, a mobile battery 402 is electrically connected to the fourth USB terminal 24 via a USB cable 301.

[0033] A first headphone 601 is electrically connected to the first output terminal 81 of recorder 1 via a connecting cable. A second headphone 602 is electrically connected to the second output terminal 81 of recorder 1 via a connecting cable. The first and second headphones 601 and 602 are used to monitor the sound input to recorder 1. For example, the first headphone 601 is used by a first user using the first microphone 501. The second headphone 602 is used by a second user using the second microphone 502.

[0034] 4. Recorder's electrical circuit Figure 4 shows the electrical circuitry that constitutes Recorder 1. Note that the electrical circuitry shown in Figure 4 includes only the components related to the data processing of the digital audio signal. Components not related to the data processing of the digital audio signal, such as power supply, recording, and playback components, are not shown in Figure 4.

[0035] The electrical circuit of recorder 1 includes first to third transmission lines 210, 220, and 230, corresponding to the first to third USB terminals 21, 22, and 23, respectively. That is, the digital audio signal data input from each of the first to third USB terminals 21, 22, and 23 is processed individually through the first to third transmission lines 210, 220, and 230. The components related to the data processing of the digital audio signal are divided into a first group common to the first to third transmission lines 210, 220, and 230, a second group common to the first and second transmission lines 210 and 220, a third group dedicated to the first transmission line 210, and a fourth group dedicated to the second transmission line 220.

[0036] The first group includes RAM 30, processor 40, microSD card 60, and D / A converter 70. RAM 30, processor 40, microSD card 60, and D / A converter 70 are all electrically connected to the first to third transmission lines 210, 220, and 230, respectively.

[0037] RAM 30 stores digital signal data input from the first to third USB terminals 21, 22, and 23 in chronological order. Processor 40 takes the data stored in RAM 30 and processes it. The data processing flow by processor 40 will be explained in detail later with reference to Figure 6. MicroSD card 60 stores the data processed by processor 40. D / A converter 70 converts the data processed by processor 40 into analog signals. The analog signals converted by D / A converter 70 are output to the first and second output terminals 81 and 82. The data processed by processor 40 and the data stored in microSD card 60 can also be output from the third USB terminal 23.

[0038] The second group includes a noise reduction switch 113, a tone switch 114, a compressor switch 115, and a third variable resistor 53. The noise reduction switch 113, the tone switch 114, the compressor switch 115, and the third variable resistor 53 are all electrically connected to the first and second transmission lines 210 and 220, respectively.

[0039] The noise reduction switch 113, tone switch 114, and compressor switch 115 are used to select whether or not to apply a predetermined sound effect to the digital sound signal. Based on whether these switches 113, 114, and 115 are ON (or not OFF), the processor 40 performs data processing to apply the predetermined sound effect to the digital sound signal. The predetermined sound effect applied by the processor 40 will be described in detail later with reference to Figure 6. The third variable resistor 53 is used to adjust the volume of the analog signals output from the first and second output terminals 81 and 82. The output volume knob 133 shown in Figure 1A is attached to the third variable resistor 53. By rotating the output volume knob 133, the resistance value of the third variable resistor 53 can be changed. Based on the resistance value of the third variable resistor 53, the processor 40 performs data processing to adjust the volume of the digital sound signal.

[0040] The third group includes a first mute switch 111, a first variable resistor 51, and a first level indicator 121. The first mute switch 111, the first variable resistor 51, and the first level indicator 121 are all electrically connected to the first transmission line 210.

[0041] The first mute switch 111 is used to reduce the volume of the analog signals output from the first and second output terminals 81 and 82 with respect to the first transmission line 210 to zero. Based on the fact that the first mute switch 111 is ON (or not OFF), the processor 40 executes a process to stop the transmission of the digital sound signal to the first transmission line 210. As a result, the first microphone 501, which is electrically connected to the first USB terminal 21, is muted. That is, the sound picked up by the first microphone 501 is not output from the first and second output terminals 81 and 82. Furthermore, the sound picked up by the first microphone 501 is not transmitted to the third USB terminal 23, which is electrically connected to the third transmission line 230.

[0042] The first variable resistor 51 is used to adjust the volume of the digital audio signal input from the first USB terminal 21. The first input volume knob 131, shown in Figure 1A, is attached to the first variable resistor 51. By rotating the first input volume knob 131, the resistance value of the first variable resistor 51 can be changed. The processor 40 performs data processing to adjust the volume of the digital audio signal input from the first USB terminal 21 based on the resistance value of the first variable resistor 51. The first level indicator 121 visually displays the volume of the digital audio signal input from the first USB terminal 21. The processor 40 performs processing to turn the six LEDs constituting the first level indicator 121 on and off in stages based on the volume of the digital audio signal input from the first USB terminal 21.

[0043] The fourth group includes a second mute switch 112, a second variable resistor 52, and a second level indicator 122. The second mute switch 112, the second variable resistor 52, and the second level indicator 122 are all electrically connected to the second transmission line 220.

[0044] The second mute switch 112 is used to reduce the volume of the analog signals output from the first and second output terminals 81 and 82 to zero with respect to the second transmission line 220. Based on the fact that the second mute switch 112 is ON (or not OFF), the processor 40 executes a process to stop the transmission of the digital sound signal to the second transmission line 220. As a result, the second microphone 502, which is electrically connected to the second USB terminal 22, is muted. That is, the sound picked up by the second microphone 502 is not output from the first and second output terminals 81 and 82. Furthermore, the sound picked up by the second microphone 502 is not transmitted to the third USB terminal 23, which is electrically connected to the third transmission line 230.

[0045] The second variable resistor 52 is used to adjust the volume of the digital audio signal input from the second USB terminal 22. The second input volume knob 132, shown in Figure 1A, is attached to the second variable resistor 52. By rotating the second input volume knob 132, the resistance value of the second variable resistor 52 can be changed. The processor 40 performs data processing to adjust the volume of the digital audio signal input from the second USB terminal 22 based on the resistance value of the second variable resistor 52. The second level indicator 122 visually displays the volume of the digital audio signal input from the second USB terminal 22. The processor 40 performs processing to turn the six LEDs that make up the second level indicator 122 on and off in stages based on the volume of the digital audio signal input from the second USB terminal 22.

[0046] 5. Data overload or underload due to sampling frequency error To simplify the explanation, the first microphone 501 will be used as an example to explain the excess or deficiency of data caused by sampling frequency errors. The second microphone 502 will also experience the same excess or deficiency of data due to sampling frequency errors.

[0047] Figure 5A shows the sampling frequencies of the first microphone 51 and recorder 1. The nominal sampling frequencies of both the first microphone 51 and recorder 1 are 48,000 Hz. The first microphone 51 performs A / D conversion 48,000 times per second and outputs a digital sound signal. On the other hand, recorder 1 performs D / A conversion 48,000 times per second and outputs an analog sound signal.

[0048] Figure 5B shows the digital sound signal input from the first microphone 51 to the recorder 1. The digital sound signal is data of amplitude values ​​converted into binary code. The first microphone 51 outputs the digital sound signal to the recorder 1 in packets. The processor 40 of the recorder 1 restores the input digital sound signal in packets to the original data and stores it in RAM 30 in chronological order. The processor 40 retrieves and processes the data stored in RAM 30. Here, for the sake of simplicity in explaining this embodiment, one packet is assumed to contain 10 data items enclosed by the dashed line in Figure 5B.

[0049] Figure 5C shows the data surplus or deficit caused by the sampling frequency error between the first microphone 51 and the recorder 1. As mentioned above, the nominal value of the sampling frequency has an extremely small error expressed in units of PPM (parts per million). The sampling frequency error accumulates and increases with each repeated sampling. In Figure 5C, +α and -α both represent the sampling frequency error of the first microphone 51 relative to the sampling frequency of the recorder 1. When the sampling frequency error +α accumulates and becomes an integer multiple of the sampling period Δt of the recorder 1, there is a shortage of data to be D / A converted in the recorder 1. On the other hand, when the sampling frequency error -α accumulates and becomes an integer multiple of the sampling period Δt of the recorder 1, there is an excess of data to be D / A converted in the recorder 1. The surplus or deficit of D / A converted data causes noise.

[0050] 6. Data processing flow by the processor Figure 6 shows the data processing flow by the processor 40. As described above, the digital audio signal data input from the first to third USB terminals 21, 22, and 23 is stored in RAM 30 in chronological order. The processor 40 processes the data stored in RAM 30 according to the flow shown in Figure 6. The data processing by the processor 40 described below includes processing to interpolate data surpluses and deficits caused by sampling frequency errors.

[0051] To simplify the explanation, the following describes the data processing flow of the processor 40 for the first microphone 501. The processor 40 also performs data processing for the second microphone 502 in the same manner as described below.

[0052] 6.1 Data Import In step S1, the processor 40 retrieves a number of data items stored in the RAM 30 per unit time determined by its own sampling frequency (the same as the sampling frequency of recorder 1). For example, the processor 40 retrieves 10 data items contained in one packet (see Figure 5B). The sampling frequency of the processor 40 is 48000Hz, and the unit time required to retrieve 10 data items is 1 / 4800 seconds. The processor 40 retrieves all the data stored in the RAM 30 within 1 / 4800 seconds and counts the number of data items Dn. As shown in Figure 5C, if the sampling frequency error α is not accumulated to an integer multiple of the sampling period Δt of the processor 40, the number of data items Dn will be "10". On the other hand, if the sampling frequency error α is accumulated to an integer multiple of the sampling period Δt of the processor 40, the number of data items Dn will not be "10".

[0053] 6.2 Determining whether there is excess or deficiency in the data In step S2, the processor 40 calculates the difference n between the number of data points Dn and the number of samples Sn performed per unit time. The processor 40 then determines whether the difference n is "0" or not. The number of samples Sn that the processor 40 performs in 1 / 4800 of a second is "10". A difference n of "0" means that there is no surplus or deficit in the number of data points Dn. On the other hand, a difference n that is not "0" means that there is a surplus or deficit in the number of data points Dn. If the processor 40 determines that the difference n is "0" (YES), it executes the process in step S5. On the other hand, if the processor 40 determines that the difference n is not "0" (NO), it executes the process in step S3.

[0054] 6.3 Calculation of Interpolated Data In step S3, the processor 40 calculates the same number of interpolated data as the sampling count Sn, based on the data corresponding to the number Dn. Figure 7A is a schematic diagram showing the interpolated data calculation process by the processor 40. Assume that the number Dn of data D1 to D9 taken into the processor 40 is "9". In this case, the processor 40 generates the same number of interpolated data S1 to S10 as the sampling count Sn, "10", based on the data D1 to D9.

[0055] A specific example of the calculation process for interpolated data S1 to S10 will be explained with reference to Figures 7B and 7C. Figure 7B is a schematic diagram for comparing data D1 to D9 with interpolated data S1 to S10. Figure 7B shows the sound waveform, data D1 to D9, and interpolated data S1 to S10. The horizontal axis shown in Figure 7B is time. The vertical axis not shown in Figure 7B is amplitude. Data D1 to D9 and interpolated data S1 to S10 represent the amplitude values ​​of the same sound waveform at different times. However, interpolated data S1 to S10 are not sampled from the sound waveform, but are generated by calculation. For example, the processor 40 calculates at least one interpolated data by calculation processing based on two data stored in RAM 30 at different times. As shown in Figure 7C, interpolated data S2 is calculated based on data D1 and D2 stored in RAM 30. Interpolated data S2 is calculated by the following formula (1).

number

[0056] Interpolated data S1, S3~S10 are also calculated by replacing data D1 and D2 in equation (1) above with two other data points that occur in a different time sequence. Note that interpolated data S1 is calculated based on data D0 which is not shown and occurs before interpolated data S1, and data D1 which occurs after interpolated data S2. After that, the processor 40 executes the process in step S4.

[0057] 6.4 Flattening of frequency response The interpolated data S1-S10 are generated by calculation and have different frequency characteristics from the data D1-D10 sampled from the analog sound signal. Figure 8A shows the time-domain waveform of the interpolated data S1-S10. Figure 8B shows the frequency-domain waveform of the interpolated data S1-S10 obtained by performing a Fast Fourier Transform on the time-domain waveform of Figure 8A. As shown in Figure 8B, the interpolated data S1-S10 have a frequency characteristic in which the amplitude value decreases at high frequencies. The frequency characteristics shown in Figure 8B indicate that the sound reproduced by the interpolated data S1-S10 has a muffled quality.

[0058] In step S4, the processor 40 flattens the known frequency characteristics of the interpolated data S1 to S10. The software for operating the processor 40 includes software for an inverse characteristic filter. The inverse characteristic filter is a calculation formula that gives the interpolated data S1 to S10 a frequency characteristic that is the opposite of the known frequency characteristics. Figure 8C shows the frequency axis waveform of the inverse characteristic filter. The processor 40 flattens the known frequency characteristics of the interpolated data S1 to S10 by performing calculations using the inverse characteristic filter. Figure 8D shows the frequency axis waveform of the interpolated data S1 to S10 after the frequency characteristics have been flattened by the inverse characteristic filter. After that, the processor 40 performs the processing in step S5.

[0059] 6.5 Muting In step S5, the processor 40 determines whether the first mute switch 111 shown in Figure 4 is OFF or not. If the processor 40 determines that the first mute switch 111 is not OFF (No), it executes the mute process in step S6. In step S6, the processor 40 executes a process to stop the transmission of the digital sound signal to the first transmission line 210 shown in Figure 4. That is, the processor 40 prevents the transmission of data D1 to D10 and interpolated data S1 to S10 to the first transmission line 210. Thereafter, the processor 40 repeats the processes in steps S1 to S6 until the first mute switch 111 is OFF.

[0060] On the other hand, in step S5, if the processor 40 determines that the first mute switch 111 is OFF (YES), it executes the input volume processing in step S7.

[0061] 6.6 Input Volume Processing In step S7, the processor 40 performs data processing to adjust the volume of the digital sound signal based on the resistance value of the first variable resistor 51 shown in Figure 4. That is, the processor 40 amplifies the data D1 to D10 and the interpolated data S1 to S10 by a ratio corresponding to the resistance value of the first variable resistor 51. After that, the processor 40 performs the processing in step S8.

[0062] 6.7 Tone Processing In step S8, the processor 40 determines whether the tone switch 114 shown in Figure 4 is OFF or not. If the processor 40 determines that the tone switch 114 is not OFF (No), it executes the tone processing in step S9. In step S9, the processor 40 performs data processing to adjust the sound quality of the digital audio signal. That is, the software for operating the processor 40 includes programs for adjusting sound quality, such as low-cut filters, de-essers, and equalizers. The processor 40 applies settings for adjusting sound quality, such as low-cut filters, de-essers, and equalizers, to the data D1 to D10 and the interpolated data S1 to S10. After that, the processor 40 executes the processing in step S10.

[0063] On the other hand, in step S8, if the processor 40 determines that the tone switch 114 is OFF (YES), it executes the process of step S10.

[0064] 6.8 Noise Reduction Processing In step S10, the processor 40 determines whether the noise reduction switch 113 shown in Figure 4 is OFF or not. If the processor 40 determines that the noise reduction switch 113 is not OFF (No), it executes the noise reduction process in step S11.

[0065] The noise reduction process in step S11 will be explained with reference to Figures 9A to 9D. First, the processor 40 acquires 96,000 data points stored in the RAM 30 within 2 seconds after the noise reduction switch 113 is turned ON. If there is an excess or deficiency of data due to sampling frequency errors when acquiring the 96,000 data points, the 96,000 data points will include interpolated data generated through steps S3 and S4. Based on the 96,000 data points and the interpolated data, the processor 40 calculates the time-domain waveform shown in Figure 9A.

[0066] Next, the processor 40 converts the time-domain waveform shown in Figure 9A into the frequency-domain waveform shown in Figure 9B by using the Fast Fourier Transform (FFT). Next, the processor 40 calculates an envelope representing the fluctuations of the frequency-domain waveform based on the frequency-domain waveform shown in Figure 9B. Next, the processor 40 identifies multiple amplitude peaks contained in the frequency-domain waveform based on the frequency-domain waveform and the envelope shown in Figure 9B. Next, the processor 40 sets a threshold level lower than the multiple amplitude peaks based on the envelope.

[0067] Next, the processor 40 attenuates the frequency components below the threshold in the frequency-axis waveform shown in Figure 9C. Then, the processor 40 uses the inverse fast Fourier transform (IFFT) to re-transform the frequency-axis waveform shown in Figure 9C, from which the frequency components below the threshold have been attenuated, into the time-axis waveform shown in Figure 9D. Next, the processor 40 generates 96,000 data points and interpolated data based on the time-axis waveform shown in Figure 9D. After that, the processor 40 performs the processing in step S12.

[0068] The noise reduction process in step S11 described above is continuously performed on the data retrieved from RAM 30 and the interpolated data until the noise canceling switch 113 is turned OFF.

[0069] On the other hand, in step S10, if the processor 40 determines that the noise reduction switch 113 is OFF (YES), it executes the process in step S12.

[0070] 6.9 Compressor Processing In step S12, the processor 40 determines whether the compressor switch 115 shown in Figure 4 is OFF or OFF. If the processor 40 determines that the compressor switch 115 is not OFF (No), it executes the compressor processing in step S13. Here, the recorder 1 mixes the digital audio signals input from the first and second microphones 501 and 502, the personal computer 201, etc., shown in Figure 3, and outputs them from the first and second output terminals 81 and 82. In step S13, the processor 40 performs data processing to increase the sound pressure of the mixed digital audio signals (data and interpolated data) and to prevent distortion. After that, the processor 40 executes the processing in step S14.

[0071] On the other hand, in step S12, if the processor 40 determines that the compressor switch 115 is OFF (YES), it executes the process in step S14.

[0072] 6.10 Output Volume Processing In step S14, the processor 40 performs data processing to adjust the volume of the digital sound signal based on the resistance value of the third variable resistor 53 shown in Figure 4. That is, the processor 40 amplifies the data D1 to D10 and the interpolated data S1 to S10 by a ratio corresponding to the resistance value of the third variable resistor 53. This adjusts the volume of the analog signals output from the first and second output terminals 81 and 82. After that, the processor 40 performs the processing in step S15.

[0073] 6.11 D / A Conversion Processing In step S15, the processor 40 transmits the data and interpolated data processed in steps S1 to S14 to the D / A converter 70 in time series. The D / A converter 70 converts 48,000 data and interpolated data points per second into analog signals and transmits them to the first and second output terminals 81 and 82.

[0074] 7. Effects According to the recorder 1 of this embodiment described above, it is possible to interpolate data surpluses and deficiencies caused by errors in the sampling frequency of the digital sound source device, thereby preventing the generation of noise caused by data surpluses and deficiencies. This makes it possible to produce music and audio content using multiple digital devices.

[0075] 8. Other The audio equipment of the present invention is not limited to the recorder 1 of the embodiment described above. For example, the data processing in steps S2, S3, and S4 shown in Figure 6 can be applied to audio equipment other than recorder 1. Furthermore, the digital equipment electrically connected to the audio equipment of the present invention is not limited to the first and second microphones 501 and 502. An electric musical instrument configured to output an analog sound signal can be electrically connected to the USB terminal provided in the audio equipment of the present invention via an audio interface. Therefore, the audio equipment of the present invention may take the form of an effects pedal or a multi-effects pedal board. [Explanation of symbols]

[0076] 1. Recorder (audio equipment) 11. First transmission line 12 Second transmission line 13. Third transmission line 21. First USB port 22. Second USB port 23. Third USB port 24. 4th USB port 30 RAM (First Memory) 40 processors 51. First variable resistor 52. Second variable resistor 53. Third variable resistor 60 microSD card (second memory) 70 D / A Converters 81 First Output Terminal 82 Second Output Terminal 101 Power switch 102 Recording switch 103 Play / Pause Switch 111 First mute switch 112 Second mute switch 113 Noise Reduction Switch 114 Tone Switch 115 Compressor Switch 121 Level 1 Indicator 122 Second Level Indicator 131. First Input Volume Knob 132 Second Input Volume Knob 133 Output volume knob 140 Battery compartment 141 snacks 150 microSD card slots 201 Personal Computers 202 Tablet Computers 203 Smartphone 210 First transmission line 220 Second transmission line 230 Third transmission line 301 USB Cable 302 Power Cable 401 Outlet 402 Mobile Battery 501 Microphone 1 502 Microphone 2 601 First Headphones 602 Second Headphone

Claims

1. An audio device configured to process digital audio signal data continuously input from a digital device, and to convert the processed data into an analog audio signal for output, At least one USB terminal for inputting the aforementioned digital signal, A first memory for storing the data of the digital signal input from the USB terminal in chronological order, A processor for processing the data stored in the first memory, A D / A converter for converting the processed data into the analog signal, It comprises at least one output terminal for outputting the aforementioned analog signal, The aforementioned processor, Counting the number (Dn) of data stored in the first memory per unit time determined by its sampling frequency, The difference (n) between the aforementioned number (Dn) and the number of samplings performed per unit time (Sn) is calculated. If the difference value (n) is not "0", calculate the same number of interpolated data as the number of occurrences (Sn) based on the data corresponding to the number (Dn). An audio device that performs the step of transmitting the same number of interpolation data as the number of times (Sn) to the D / A converter in time series.

2. The audio device according to claim 1, wherein the processor calculates at least one interpolated data by performing an arithmetic operation based on two pieces of data stored in the first memory in an orderly fashion.

3. The processor further comprises an inverse characteristic filter used for processing, The inverse characteristic filter is a calculation formula for obtaining a frequency characteristic that is the opposite of the known frequency characteristic shown by the same number of interpolated data (Sn) as the number of times, The audio device according to claim 2, wherein the processor flattens the known frequency characteristics by performing an arithmetic operation using the inverse characteristic filter on the same number of interpolation data as the number of times (Sn).

4. The aforementioned processor, Calculate the time-axis waveform based on at least the same number of data and / or interpolated data as the number of times (Sn), By using the Fast Fourier Transform, the time-domain waveform is converted into a frequency-domain waveform. To calculate the envelope showing the fluctuation of the frequency axis waveform, Based on the frequency axis waveform and the envelope, identify one or more amplitude peaks included in the frequency axis waveform. Based on the envelope, set a threshold at a level lower than the amplitude peak. To attenuate frequency components below the threshold included in the frequency axis waveform, By using the inverse fast Fourier transform, the frequency-axis waveform, with frequency components below the threshold attenuated, is re-converted into the time-axis waveform. The audio device according to claim 1, which performs the step of generating at least the same number of data and / or interpolated data as the number of times (Sn) based on the re-converted time-axis waveform.

5. The device is equipped with a transmission path corresponding to the number of USB terminals, The USB terminal is electrically connected to the transmission line. The transmission line is electrically connected to a mute switch, a first variable resistor, a second variable resistor, a second memory, the processor, and the D / A converter. The mute switch is used to set the volume of the analog signal output from the output terminal to zero. The first variable resistor is used to adjust the volume of the digital signal input from the USB terminal. The second variable resistor is used to adjust the volume of the analog signal output from the output terminal. The audio device according to claim 1, wherein the second memory is configured to store the data processed by the processor.

6. A level indicator is electrically connected to the aforementioned transmission line. The audio device according to claim 5, wherein the level indicator is configured to visually display the volume of the digital signal input from the USB terminal.

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