Microphone with adjustable signal processing and signal processing method thereof

By supporting multiple connection methods and a built-in DSP microphone system, the problems of difficult multi-microphone connection and proximity effect are solved, achieving convenient connection and high-quality voice recording.

CN114946195BActive Publication Date: 2026-04-21SHURE ACQUISITION HLDG INC
View PDF 9 Cites 0 Cited by

Patent Information

Authority / Receiving Office
CN · China
Patent Type
Patents(China)
Current Assignee / Owner
SHURE ACQUISITION HLDG INC
Filing Date
2020-09-30
Publication Date
2026-04-21

AI Technical Summary

Technical Problem

Existing microphones suffer from connectivity difficulties and proximity effects when using multiple microphones, and users need good technical skills to obtain consistent voice recording results.

Method used

A microphone system was designed that supports multiple connection methods (such as USB and XLR), has a built-in DSP to automatically or manually adjust signal processing modes, including multi-band compression and EQ settings, to adapt to different speech scenarios and tones, and is controlled through a user interface.

Benefits of technology

It enables convenient connection of multi-microphone systems and consistent audio output, reduces proximity effect, and improves the quality of voice recording and user experience.

✦ Generated by Eureka AI based on patent content.

Smart Images

  • Figure CN114946195B_ABST
    Figure CN114946195B_ABST
Patent Text Reader

Abstract

A microphone can include a microphone element to detect sound, and a digital signal processor configured to process a first audio signal based on the sound according to a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes can be to process the first audio signal differently. For example, the DSP modes can account for a distance of a person speaking (e.g., close versus far) and / or a desired tonal quality (e.g., a lower, neutral, or bright tonal quality). At least some of the modes can have, for example, an automatic level control setting to provide more consistent volume as a user changes their distance from the microphone or changes their speaking level, and the automatic level control setting can be associated with particular default (and / or adjustable) values of a parameter attack, hold, decay, maximum gain, and / or target gain, each of which depends on which DSP is being applied.
Need to check novelty before this filing date? Find Prior Art

Description

[0001] Cross-references to related applications

[0002] This application claims priority to U.S. Provisional Patent Application Serial No. 62 / 939,347, filed November 22, 2019, entitled "Microphone With Adjustable Signal Processing," the entire contents of which are incorporated herein by reference. Technical Field

[0003] This disclosure generally relates to the field of audio processing, and more specifically to a microphone having adjustable signal processing and a signal processing method performed by the microphone. Background Technology

[0004] Each type of microphone has its advantages and limitations. For example, unidirectional (e.g., cardioid) dynamic microphones offer several advantages for voice recording. These advantages include a rich low end and greater immunity to room noise. However, due to the proximity effect and lower sensitivity, such microphones often require users to have some knowledge of "good" microphone techniques to obtain consistent results.

[0005] Furthermore, while Universal Serial Bus (USB) connected microphones are generally simple and convenient to use in a single-microphone setup, using more than one microphone at a time in the same setup can be quite difficult, and sometimes even impossible. Summary of the Invention

[0006] The following overview presents a simplified overview of some features. This overview is not a comprehensive overview and is not intended to identify key or essential components.

[0007] This article describes an example of a microphone and the methods for operating and implementing that microphone.

[0008] Depending on some aspects, a microphone may include any type of microphone, such as, but not limited to, a unidirectional microphone, a multi-directional microphone, an omnidirectional microphone, a dynamic microphone, a cardioid dynamic microphone, or a condenser microphone.

[0009] According to another aspect, the microphone may include various types of signal connectors, such as one or more Universal Serial Bus (USB) connectors and / or one or more XLR connectors, which can be used with a variety of other devices, such as Apple Mac computers and portable devices, Windows PC computers and portable devices, Android devices, XLR mixers and interfaces, etc. The microphone connectors may include one or more digital signal connectors (e.g., USB) and / or one or more analog signal connectors (e.g., XLR). Each connector can be used as an input connector, an output connector, or both. Users of the microphone can conveniently use one or more of the connectors to expand the microphone to make it part of a larger setup using multiple microphones. For example, the microphone's XLR connector may be passive and can be configured so that a user can daisy-chain the output from another microphone's XLR connector to the microphone's XLR connector. In such an arrangement, the output from two microphones can be output through another connector of the microphone, such as a USB connector.

[0010] According to another aspect, the circuitry providing audio output via the USB connector may include a preamplifier and / or a digital signal processor (DSP) in the signal chain. The preamplifier and / or DSP may be low-noise circuitry.

[0011] According to another aspect, the DSP can process incoming audio according to any of a plurality of modes. These modes may take into account, for example, the speaker's distance (e.g., near vs. far) and / or the desired pitch (e.g., lower, neutral, or bright). Some modes may be automatic modes, which can be tuned, for example, to suit different speech center scenarios. More specifically, automatic modes may be divided into, for example, multiple microphone position scenarios (e.g., near and far position scenarios) and / or multiple pitch scenarios (e.g., lower, neutral, and bright pitch scenarios). Users can set these manually, or the DSP can automatically set any combination of these two scenarios. Based on the selected scenario combination, the microphone DSP can adjust one or more of the settings to provide the desired output audio signal for various audio applications, such as voice applications. Near modes may include, for example, multi-band compression and / or EQ settings designed to reduce the amount of proximity effect and sibilance. Far modes may include, for example, multi-band compression and / or EQ settings designed to increase the low end and reduce the thinness experienced when using a unidirectional microphone from a distance greater than a predetermined distance (e.g., greater than 12 inches). Furthermore, depending on the selected tone scenario, tone adjustment can, for example, adjust the equalization based on personal preference to provide the user with a deeper or brighter sound. Additionally, the automatic mode can include automatic level control to provide a more consistent volume as the user changes their distance from the microphone or their speaking level.

[0012] According to another aspect, one or more of the other modes can be manual modes, which allow the user to manually control one or more of the settings (although some settings may or may not still be set automatically).

[0013] According to another aspect, the DSP mode can be selected via a user interface (such as via a USB connector) on a device connected to the microphone. This device may include, for example, a computing device such as a smartphone, tablet, or personal computer. The device can present a user interface to the user, through which the user can select desired operating characteristics of the microphone, such as the desired DSP mode or any other desired microphone settings.

[0014] According to another aspect, the microphone itself may include a user interface, and the user interface may include a capacitive touch interface, which may be a curved capacitive touch interface. This user interface allows the user to manually select one or more microphone settings as needed.

[0015] According to another aspect, the microphone can be connected to a retainer, such as a yoke that can be installed in a tabletop and / or suspended studio configuration. The retainer can be, for example, U-shaped.

[0016] According to another aspect, a microphone may be provided, comprising a microphone element configured to detect sound. The microphone may also include a digital signal processor configured to process a first audio signal based on sound according to a selection of a plurality of digital signal processing modes. Each of the plurality of digital signal processing modes may be used to process the first audio signal in a different manner and to automatically adjust one or more parameters to achieve a target gain or other target characteristics of the audio.

[0017] According to another aspect, a method can be implemented that includes generating a first audio signal based on sound detected by a microphone element of a microphone. The microphone can select a chosen digital signal processing mode from a plurality of digital signal processing modes. Each of the plurality of digital signal processing modes can be used to process the first audio signal in a different manner. The microphone can also automatically adjust one or more parameters applied to the digital signal processing mode to achieve a target gain or other target characteristics of the audio. The microphone can also perform digital signal processing of the first audio signal according to the selected digital signal processing mode. Such digital signal processing may further include automatically adjusting the gain based on the first audio signal. As a result of the digital signal processing, the microphone can also generate a digital audio signal and output the digital audio signal via a first connector of the microphone, such as a USB connector.

[0018] According to another aspect, a method can be implemented that includes receiving an audio signal from a second microphone or other device via a first microphone through a first connector (such as an XLR connector). The first microphone can detect sound (e.g., by using a microphone element). The microphone can also output a digital signal based on both the audio signal received via the first connector and the detected sound via a second connector (such as a USB connector).

[0019] These and other features, as well as their potential advantages, are described in more detail below. Attached Figure Description

[0020] In the accompanying drawings, some features are shown as examples rather than limitations. Similar reference numerals in the drawings denote similar elements.

[0021] Figure 1A A side view of an example microphone is shown according to the aspects described in this article.

[0022] Figure 1B It shows Figure 1A Example front view of a microphone.

[0023] Figure 2 The microphone shown is illustrated according to the aspects described herein (such as...). Figure 1A and Figure 1B A block diagram showing an example of a microphone interconnected with one or more other devices.

[0024] Figure 3 Example microphones (such as) are shown according to the aspects described in this article. Figure 1A and Figure 1B Example block diagram of one or more parts of a microphone.

[0025] Figure 4A Example microphones (such as) are shown according to the aspects described in this article. Figure 1A and Figure 1B Another example block diagram of one or more parts of a microphone.

[0026] Figure 4B It shows Figure 4A Another part of the block diagram.

[0027] Figure 5 Example microphones (such as) are shown according to the aspects described in this article. Figure 1A and Figure 1B Another example block diagram of one or more parts of a microphone.

[0028] Figure 6 A block diagram of an example computing device based on the aspects described herein is shown.

[0029] Figure 7A An example microphone (such as a digital signal processing system (DSP)) including at least a portion thereof, is shown according to the aspects described herein. Figure 1A and Figure 1B Another example block diagram of one or more parts of a microphone.

[0030] Figure 7B It shows Figure 7A A block diagram showing further examples of details of the desirer and bass tamer.

[0031] Figure 8 An example flowchart is shown, illustrating a method that can be implemented according to the aspects described herein.

[0032] Figure 9 This document illustrates how various aspects of a device can be connected to a microphone (such as...) Figure 1A and Figure 1B The example user interface shown is for a device with a microphone. Detailed Implementation

[0033] The accompanying drawings, which form part of this document, illustrate examples of this disclosure. It should be understood that the examples shown in the drawings and / or discussed herein are non-exclusive, and other examples exist regarding how this disclosure may be practiced.

[0034] Figure 1A and Figure 1B Side and front views of an example microphone 100 are shown, respectively. The microphone 100 may include a body 101 that may house one or more other components of the microphone, such as those described herein. Figure 2 , Figure 3 , Figure 4A , Figure 4B , Figure 5 , Figure 7A and / or Figure 7B Any of the aforementioned electronic circuits. The microphone 100 may also include a windscreen 102 covering the microphone elements.

[0035] The body 101 may include a ring portion 103 or other portions on and / or within which a user interface 104 may be disposed. The user interface 104 may include, for example, a capacitive touch interface that can be controlled by a user via touch (e.g., by tapping and / or swiping a user's finger). The user interface 104 may have a shape consistent with the external shape of a portion of the body 101, and the user interface 104 may be disposed on / within the body 101. For example, if a portion of the body 101 is curved, the user interface 104 may also be curved in the same manner as the portion of the body 101.

[0036] The body 101 can be connected to a retainer 105, such as a yoke (e.g., mounted on retainer 105). The retainer can be used to mount the microphone to other objects, such as a table or a wall.

[0037] Body 101 may have one or more connectors, such as connectors 106a and 106b. The connectors (generally referred to herein as one or more connectors 106) may include, for example, one or more Universal Serial Bus (USB) connectors, one or more XLR connectors, one or more power connectors, and / or any other type of data and / or power connector, adapted to transmit signals such as power, digital data (including digital audio signals), and / or analog audio signals to and from the circuitry of microphone 100. In a particular instance, connector 106a may be an XLR connector, and connector 106b may be a USB connector.

[0038] Figure 2 A block diagram illustrating an example interconnection between a microphone (such as microphone 100) and one or more other devices according to various aspects described herein is shown. In this example, microphone 100 is shown connected to another XLR-compatible microphone 201 via an XLR cable (such as via connector 106a). The other microphone 201 can transmit an audio signal representing the sound detected by the other microphone 201 to microphone 100 via the XLR cable. Microphone 100 can process a combination of the audio signal on the XLR cable and the audio signal representing the sound detected by the microphone element of microphone 100. The microphone element can be any type of microphone element, such as a dynamic element or a condenser element.

[0039] Microphone 100 can also be connected to another device, in this example, a networked device 202, via a USB cable (such as via connector 106b). Device 202 can be a computing device, such as a smartphone (e.g., an iPhone or Android phone), tablet computer, laptop computer, desktop computer, server, etc. As described later, microphone 100 may include a digital signal processing system (DSP), and device 202 can be used to control one or more settings of microphone 100 via signals transmitted on the USB cable connecting device 202 and microphone 100, including the mode in which the DSP operates. In addition to transmitting setting information, microphone 100 can also use the USB cable to transmit digital signals representing audio. For example, the DSP of microphone 100 can process both audio signals received from another microphone 201 via an XLR cable and audio signals generated by sound detected by the microphone elements of microphone 100. Digital signals generated at least in part by the processing of the DSP can be sent to device 202 via the USB cable. Therefore, the XLR connector (which may be passive) of microphone 100 (connected to another microphone 201) can be configured to allow a user to daisy-chain the output of the XLR connector from the other microphone 201 to the XLR connector of microphone 100. Thus, the user of the microphone can conveniently use one or more of the connectors of microphone 100 to expand microphone 100 to make it part of a larger setup using multiple microphones (in this example, microphones 100 and 201).

[0040] While a USB connection between microphone 100 and device 202 has been discussed, other types of wired or wireless connections can also be used. For example, the connection between microphone 100 and device 202 can alternatively be a wireless connection, such as a WiFi connection, Bluetooth connection, near field connection (NFC), and / or infrared connection. In the case of a wireless connection, microphone 100 and device 202 may include a wireless communication interface.

[0041] Microphone 100 can also be connected to one or more other devices (such as another XLR-compatible device 203) via another connector (such as another XLR connector). Examples of other XLR-compatible devices may include yet another microphone, mixer, amplifier, computing device, etc.

[0042] Figure 3An example block diagram of one or more portions of an example microphone (such as microphone 100) is shown. In the illustrated example, microphone 100 may have one or more XLR connectors (labeled "XLR Output" in the figures; and these may be one or more of the previously discussed connectors 106, such as connector 106a); at least one microphone barrel, which may include one or more microphone elements; audio output, such as a 3.5 mm stereo audio jack connector; microphone preamplifier; headphone amplifier and / or line driver; capacitive touch interface (which may be the previously discussed user interface 104); and / or one or more USB connectors (which may be one of the previously discussed connectors 106, such as connector 106b), such as one or more USB micro B connectors or conventional USB connectors. As an example, Figure 3 It also identifies several XLR connection configurations, several output configurations, and several USB connection types.

[0043] Figure 4A and Figure 4B Together with the above, another example block diagram of one or more portions of an example microphone, such as microphone 100, is shown. In the example shown, microphone 100 can be considered to be logically divided into an audio subsystem function (referred to herein as "audio subsystem 401") and a digital subsystem function (referred herein as "digital subsystem 402").

[0044] The audio subsystem 401 is responsible for routing and processing audio and digital signals representing audio. The audio subsystem 401, which may physically share circuit components with the digital subsystem 402, may have a front end including the aforementioned microphone, one or more XLR connectors, the aforementioned audio jack (which may be a stereo headphone jack), and the aforementioned microphone preamplifier.

[0045] The audio subsystem may also include circuitry such as amplifiers (e.g., adjustable gain amplifiers), input mixers, analog-to-digital converters (ADCs), digital-to-analog converters (DACs), control register 420, data interfaces (such as I2C interfaces), audio interfaces (such as serial I2S interfaces 406), and DSP 403, such as... Figure 4A and Figure 4B The interconnections are shown. (As shown in the diagram) Figure 4A As indicated, an audio path exists from the microphone and / or XLR connector (as an analog audio signal), which is digitized by the ADC via the input mixer to generate a digital audio signal, through the DSP, through a set of amplifiers, and then finally provided as PCM digital audio data processed by the DSP, so as to be transmitted by the CODEC 405 to the controller 404 via the serial audio I2S interface 406. Figure 4BThe controller 404 can be implemented as, for example, a system-on-a-chip (SoC). Digital audio and / or other data can be transmitted bidirectionally via an I2S connection (from CODEC 405 to controller 404, and / or from controller 404 to CODEC 405). Figure 4A and Figure 4B Any of the circuits in the diagram can be implemented as, for example, a programmable gate array (PGA). An example amplifier component embodied by one or more PGAs is... Figure 4A The circuit is marked "PGA". Some or all of the circuits described above may be incorporated as part of CODEC 405, which may be an integrated circuit device.

[0046] The CODEC 405 may include an adjustable gain stage and / or mixer for audio inputs. For example, the amplifier component embodied in the indicated PGA for the microphone input may have adjustable gain (“Adj Gain”), such as approximately 36 dB. The line input may also have adjustable gain, such as approximately 6 dB. The ADC is capable of accepting, for example, dual microphone inputs, a combination of microphone and line inputs, or dual line inputs (e.g., two channels in each of these cases). The amplifier component embodied in the indicated PGA may have a non-inverting input with variable impedance, allowing the microphone 100 to be cross-connected to two PGAs, with each branch having the same input impedance.

[0047] Digital subsystem 402 can be interconnected with analog audio subsystem 401 via one or more signaling lines, such as shown in the example of... Figure 4A circuit and Figure 4B The signaling lines are interconnected by the circuitry. The digital subsystem 402 can be responsible for the overall control of the microphone 100 and may include components such as one or more processors. The controller 404 may also embody the features discussed above and... Figure 4A One or more PGAs are shown in the figure.

[0048] Controller 404 may have various signal inputs and outputs for communication with other parts of digital subsystem 402 and analog subsystem 401, such as those shown in the accompanying drawings. For example, digital audio signals provided by audio subsystem 401 may be received by controller 404 for further routing and / or processing, and the resulting processed and / or routed digital signals may then be transmitted from controller 404 to USB connector 106b via a USB interface (labeled "USB IF" in the accompanying drawings). Therefore, the audio path in audio subsystem 401 may also include controller 404 and USB connector 106b, ultimately transmitting the resulting audio data via paths labeled "data+" and "data-" between controller 404 and USB connector 106b. In addition to the aforementioned audio data, these "data+" and "data-" lines connected to USB connector 106b may also transmit control signaling to and from microphone 100 bidirectionally. Controller 404 may be configured to implement both iAP and standard USB audio endpoints. Onboard MCU 407 may be used for system control and may be linked to iAP control.

[0049] Controller 404 may also include one or more processors (such as MCU 407) for performing various functions. Controller 404 may also store executable code (e.g., software) for performing various steps in a non-transitory computer-readable storage medium (such as memory, which may be part of or connected to MCU 407). When executed by controller 404 or at least by one or more of its processors, the code can cause controller 404 to perform steps defined by the code.

[0050] For example Figure 4B As shown, the user interface 104 may include a plurality (e.g., seven) of capacitive sensors 408 and / or a plurality (e.g., 21) of indicator lights (e.g., LEDs) 409 having corresponding drivers 410. Figure 1A In the diagram, indicator lights (and / or other indicators) are shown as an example as a series of black dots on user interface 104. The user interface, capacitive sensors, and / or indicator lights can be controlled by MCU 407. The capacitive sensors can be controlled by a controller, such as capacitive sensor controller 411, to assist MCU 407 in controlling the capacitive sensors.

[0051] Figure 5 Another example block diagram shows one or more portions of an example microphone, such as microphone 100. In this block diagram, an example audio output chain of microphone 100 is shown, including analog outputs / inputs to / from XLR connector 106a, an analog output to headphone (HP) jack 501, and a digital output to USB connector 106b. Figure 5This includes at least a portion of each of the audio subsystem 401 and the digital subsystem 402, as shown from different logical and functional perspectives, specifically from the perspective of the audio subsystem 401.

[0052] Figure 7A and Figure 7B Another example block diagram shows one or more portions of an example microphone (such as microphone 100), including details of at least a portion of DSP 403. DSP 403 may include one or more modules for processing audio, including, for example, one or more equalizers, such as a high-pass / single-frequency boost equalizer 702 and / or a modal equalizer 703; a desirer 704; a bass equalizer, such as a bass tamer (which may be used to reduce proximity effect) 705; a limiter 706; a compressor 707; and / or automatic level control (ALC) 708. Each of modules 702 to 708 may be embodied as, for example, physical dedicated circuitry and / or software executed by one or more processors, such as a processor as part of controller 404.

[0053] As previously discussed, DSP 403 can operate in any of a plurality of modes. Each of these modes can be associated with different combinations or sub-combinations of settings for any or all modules 702 to 708. In order to set modules 702 to 708 according to a desired DSP mode setting, each of these modules 702 to 708 can be controlled to operate according to a mode setting signal 701 (in... Figure 7A The term "dsp mode (n)" is used, where n can indicate various configuration operations of one or more signals (the dimension of mode setting signal 701). For example, mode setting signal 701 and / or another signal can be presented as a data packet (e.g., a bus data packet) that serially transmits n bits or n bytes of data, or as n parallel transmission lines that each transmit one or more bits or one or more bytes of data. As an example, mode setting signal 701 and / or another signal can represent the setting of each of modules 702 to 708. For example, mode setting signal 701 and / or another signal can indicate a specific setting for module 702, another specific setting for module 703, another specific setting for module 704, another specific setting for module 705, another specific setting for module 706, another specific setting for module 707, and / or another specific setting for module 708. Examples of such settings will be discussed further below.

[0054] Figure 7B A detailed example implementation of the desirer 704 and bass tamer 705 is shown.

[0055] Figure 8An example flowchart of a method that can be executed is shown. Some of these steps can be performed by a microphone (such as microphone 100), others by a device connected to the microphone (such as device 202), and still others by a user of that device and / or microphone. While the method shows specific steps in a particular order, the method can be further subdivided into additional sub-steps, steps can be combined, and steps can be performed in other orders without departing from the concept described herein.

[0056] In operation, the user can connect the microphone 100 to the device 202, for example, by using a USB cable connecting the USB connector of the device 202 to the USB connector 106b of the microphone 100. Figure 8 (Step 801). The user can also run software (e.g., an app) on device 202 that is configured to control one or more settings of microphone 101 (Step 802). For example, when the software is executed, it can cause device 202 to display a user interface that the user can interact with (e.g., via a touchscreen, mouse, keyboard, etc.) to select the DSP mode in which microphone 100 should operate (Step 803). Figure 9 An example of such a user interface is shown. In response to user selection, device 202 can transmit control signals (such as data) via USB cable indicating settings according to one or more DSP modes selected by the user (step 804). These control signals can be received at the USB connector 106b of microphone 100.

[0057] Next, the control signal at USB connector 106b can be sent from the USB interface ( Figure 4B The received data is then presented to the MCU 407 for processing. Based on the received USB control signal, the MCU 407 can generate mode setting data (step 805), which can be transmitted via... Figure 4A and Figure 4B The "I2C data" connection identified in the code is used to send the data for storage in one or more control registers 420. This stored mode setting data can be retrieved and used as the mode setting signal 701 (step 806). Figure 7AAs indicated, the mode setting signal 701 may also include or be supplemented by other data such as "input gain()", "setting box (limiter)", and "setting box (compressor)" to set the DSP mode (step 807). Based on the mode setting signal 701 and / or such other signals, the DSP 403 may configure itself, including some or all of the configuration modules 702 to 708, to process sound and / or other audio signals (such as XLR audio signals received from another microphone 201) detected by the microphone 100 (step 808). As will be further described below, each mode may have one or more settings adjustable within that mode, either manually by the user or automatically and dynamically adjusted by the DSP 403 itself without manual intervention (step 809). If the user selects another DSP mode, then Figure 8 The process can return to step 803 to select the next DSP mode, and the remainder of the process can be repeated as needed.

[0058] Figure 9 An example user interface 900 is shown that can be displayed by a device connected to a microphone (such as device 202 connected to microphone 100). User interface 900 can be the user interface with which a user interacts in step 803. However, other user interfaces for selecting or otherwise setting the DSP mode can be used alternatively to implement step 803.

[0059] User interface 900 may include a microphone selection section 901, which may include, for example, a drop-down menu, and may be used to select one or more microphones (e.g., microphone 100) connected to device 202. User interface 900 may also include an automatic / manual selection section 902, which may be used to select between automatic DSP mode operation and manual DSP setting operation. In automatic DSP mode operation, the user can select from a plurality of provided predetermined DSP modes and configure DSP 403 according to the selected DSP mode. In manual DSP setting operation, the user can manually configure each of a plurality of DSP settings.

[0060] User interface 900 may also include a microphone mute switch 904, which can be used to selectively mute sounds detected by microphone 100. User interface 900 may also include monitor mixing settings (e.g., a slider) 905, which can be used to select the mixing or ratio of audio signals output by microphone 100 (e.g., based on sound detected by the microphone barrel and / or received from another microphone via XLR connector 106a) sent to headphone jack 501 for real-time monitoring purposes. This allows the user to hear audio in real-time and access it through headphones via a processing chain.

[0061] User interface 900 may also include microphone position settings (such as a slider) 906, which can be used to select the expected distance (or range of distances) from which microphone 100 will be used from a sound source (e.g., from a person speaking or singing into microphone 100). While specific distances of “within 6 inches” and “6 to 18 inches away” are shown as potential settings, any other distance may be used as a potential setting.

[0062] User interface 900 may also include tone settings (such as a slider) 907, which can be used to select from multiple tone configurations (such as a lower tone, a neutral tone, and / or a brighter tone).

[0063] As previously discussed, DSP 403 can be configured to operate in multiple DSP modes. For example, these multiple DSP modes can each be organized as a combination of a specific microphone distance (e.g., near, such as within six inches, versus far, such as 6 to 18 inches) and a specific tone (e.g., lower pitch versus neutral versus brighter pitch). Given such an organization, the multiple DSP modes can therefore include, for example, the following six DSP modes: near-neutral mode, near-lower pitch mode, near-brighter pitch mode, far-neutral mode, far-lower pitch mode, and far-brighter pitch mode.

[0064] Based on the selected DSP mode, DSP 403 can adjust one or more settings to provide the desired output audio signal for various audio applications, such as voice applications. Proximity modes (e.g., near-neutral, near-lower, near-brighter) may include, for example, multi-band compression and / or EQ settings designed to reduce the amount of proximity effect and sibilance. Far modes (e.g., far-neutral, far-lower, and far-brighter) may include, for example, multi-band compression (using compressor module 707) and / or equalization (EQ) settings (using mode equalizer module 702) designed to increase the low end and reduce the thinness experienced when using a unidirectional microphone from a distance greater than a predetermined distance (e.g., greater than 12 inches). Furthermore, depending on the selected tonal scenario (e.g., neutral, lower, brighter), tone adjustment may, for example, adjust the EQ based on personal preference to provide the user with a lower or brighter sound. Additionally, the DSP mode may include automatic level control (using module 708) to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level.

[0065] More specifically, each of the DSP modes can be associated with different combinations or sub-combinations of audio settings, some or all of which can be represented by the mode setting signal 701 and implemented by the DSP 403. Examples of audio settings that can be associated with the DSP modes include settings for configuring any of the modules 702 to 708. These settings may include, for example, automatic level control settings, desiccant settings, bass taming settings, limiter settings, and / or equalization settings. Each of these settings is discussed below.

[0066] Automatic Level Control (ALC) is a method of adjusting amplifier gain based on the desired output signal level. By adjusting the amplifier's input gain, the ALC module 708 provides a way to maintain the desired (e.g., maximized) signal-to-noise ratio of the audio output. Automatic Level Control settings can be set to specific values, such as parameter attack, hold, attenuation, maximum gain, and / or target gain, each depending on which of several DSP modes is selected. Table 1 below summarizes such example settings for the ALC module 708.

[0067]

[0068] As examples, values ​​A1 to A6 can range from approximately 10 to approximately 50 milliseconds, values ​​H1 to H6 can range from approximately 1 to approximately 2 seconds, values ​​D1 to D6 can range from approximately 500 to approximately 1000 milliseconds, values ​​MG1 to MG6 can range from approximately +10 to approximately +20 dB, and values ​​TG1 to TG6 can range from approximately -5 to approximately -12 dB. Each of these values ​​depends on which of the multiple DSP modes is selected. The ranges of values ​​described above are merely examples and are not intended to limit the possible values ​​that can be used; appropriate values ​​will depend on the specific characteristics of the microphone 100.

[0069] As previously discussed, some or each of the DSP modes can also be configured to dynamically and automatically adjust (dynamic intra-frame mode adjustment) one or more parameters of the DSP mode over time within the DSP mode (e.g., any of the audio settings described herein). For example, an automatic level control setting can dynamically and automatically adjust any of the parameters attack, hold, and / or attenuation within a given DSP mode to automatically achieve and hold below the indicated target gain (e.g., any of TG1 to TG6) associated with the given DSP mode. Other parameters of the DSP mode can be additionally or alternatively adjusted to ensure that the maximum gain of the automatic level control in the far DSP mode is at a higher level (e.g., a higher decibel level) compared to the near DSP mode. Furthermore, the automatic level control setting can include a noise gate configured to maintain a constant gain when the input drops below a predetermined threshold (e.g., in the range of approximately -50 to approximately -60 dB) to potentially reduce or even avoid excessive gain on the noise floor.

[0070] Desiring is a technique used to reduce the amount of sibilant constants ('s', 'ch', 'z', 'sh') in a recording. For example... Figure 7B As shown, a multi-band compressor can be used to implement the desirer module 704. Desirer settings can be used to configure the desirer module 704 and can be set to, for example, a specific desiring band, and specific values ​​for parameters such as attack, attenuation, ratio, and / or threshold, at which desiring is automatically performed. Each of these specific values ​​depends on which of several DSP modes is selected. The ratio parameter is a compression parameter that determines how much compression is applied to the input audio signal after exceeding the threshold parameter. Table 2 below summarizes such example settings for the desirer module 704.

[0071]

[0072] As examples, values ​​EB1 to EB6 can be specific frequencies in the range of approximately 3 to approximately 10 kHz, values ​​A7 to A12 can be in the range of approximately 3 to approximately 100 milliseconds, values ​​D7 to D12 can be in the range of approximately 50 to approximately 100 milliseconds, values ​​R1 to R6 can be in the range of approximately 1 to approximately 2, and values ​​T1 to T6 can be in the range of approximately -40 to approximately -20 dB. The ranges of values ​​above are merely examples and are not intended to limit the possible values ​​that can be used; appropriate values ​​will depend on the specific characteristics of the microphone 100.

[0073] Bass conditioning is a technique used to dynamically reduce the amount of low-frequency signal in a recording. For example... Figure 7BAs shown, the bass tamer module 705 can achieve this using a multi-band compressor, the purpose of which is to smooth the increase in low frequencies (also known as the proximity effect) when the user moves closer to the directional microphone. Bass tamer settings (which can configure the bass tamer module 705) can be set to specific values ​​for parameters such as crossover, attack, hold, attenuation, ratio, and / or threshold, at which bass taming is performed automatically. The ratio parameter can be a compression parameter that determines how much compression is applied to the input audio signal after exceeding the threshold parameter. Table 3 below summarizes such example settings for the bass tamer module 705.

[0074]

[0075] As examples, values ​​C1 to C6 can be in the range of approximately 100 to approximately 300 Hz, or even disabled; values ​​A13 to A18 can be in the range of approximately 10 to approximately 100 milliseconds; values ​​H7 to H12 can be in the range of approximately 10 to approximately 40 milliseconds; values ​​D7 to D12 can be in the range of approximately 50 to approximately 1000 milliseconds; values ​​R7 to R12 can be in the range of approximately 2 to approximately 3; and values ​​T7 to T12 can be in the range of approximately -20 to approximately -35 dB. Each of these values ​​depends on which of the multiple DSP modes is selected. The ranges of values ​​described above are merely examples and are not intended to limit the possible values ​​that can be used; appropriate values ​​will depend on the specific characteristics of the microphone 100.

[0076] Limiter module 706 may include a compressor for preventing audio signal clipping. Limiter settings (which can configure limiter module 706) can be set to specific values ​​for parameters such as attack, hold, attenuation, and / or threshold, at which limiting is automatically performed. Table 4 below summarizes such example settings for limiter module 706.

[0077]

[0078] As examples, values ​​A19 to A24 can range from approximately 50 to approximately 100 milliseconds, values ​​H13 to H18 can range from approximately 0 to approximately 20 milliseconds, values ​​D13 to D18 can range from approximately 500 to approximately 1000 milliseconds, and values ​​T13 to T18 can range from approximately -2 to approximately -5 dB, each depending on which of the multiple DSP modes is selected. In some instances, the values ​​for attack, hold, attenuation, and threshold can be the same for all six DSP modes listed above, and in other instances, these values ​​can differ between the six DSP modes listed above. The ranges of values ​​described above are merely examples and are not intended to limit the possible values ​​that can be used; appropriate values ​​will depend on the specific characteristics of the microphone 100.

[0079] Equalization is the selective increase or decrease of certain frequencies. This increase or decrease can be static or can be adjusted over time (typically relatively slowly). To achieve equalization, equalization settings can be configured using the mode equalizer module 703 and / or the HP single-band boost EQ module 702 (the latter can emphasize higher mid-band frequencies, e.g., approximately 4 kHz to approximately 8 kHz) and can help set the intelligibility of equalization parameters for one or more frequency bands. For example, a given DSP mode can be associated with one or more frequency bands to be equalized in a specific manner. For each frequency band in a given DSP mode, there can be a specified specific filter type (e.g., high-pass filter, low-pass filter, or peaking filter) and specified specific values ​​for boost / cutoff and / or Q parameters.

[0080]

[0081] As examples, values ​​B1 to B6 can be low frequencies such as those in the 50 to 100 Hz range, or higher frequencies such as those in the hundreds or even thousands of Hz range. Also as examples, values ​​T1 to T6 can indicate the filter shape, such as a high-pass filter, a low-profile filter, or a peaking filter. Again as examples, values ​​BC1 to BC6 can be in the range of approximately -3 to approximately 3, and values ​​Q1 to Q6 can be in the range of approximately 0.5 to approximately 2. The ranges of values ​​described above are merely examples and are not intended to limit the possible values ​​that can be used; appropriate values ​​will depend on the specific characteristics of the microphone 100.

[0082] Each of the aforementioned DSP modes can be selected by the user via the user interface of control device 202, such as in the "Auto" setting 902 of user interface 900. However, the user can alternatively use the "Manual" setting 902 of user interface 900 to manually set (e.g., overridden, if previously set automatically) any or all of the parameters discussed above. User interface 900 can present a user interface for allowing the user to manually set the value of any desired parameter of any of modules 702 to 708. Additionally or alternatively, the user can set the desired DSP mode and / or any of the values ​​discussed above, such as directly on the microphone via user interface 104.

[0083] Figure 6A block diagram of an example computing device 600 is shown. The computing device 600 can be used to implement at least a portion of device 202 and / or at least a portion of microphone 100. For example, some or all of controller 404, some or all of MCU 407, some or all of DSP 403 and / or some or all of its supporting circuitry can alternatively be implemented by computing device 600. Computing device 600 can be any type of computing device and can be physically implemented as a single unit or a system of multiple interactive units. For example, computing device 600 may include one or more smartphones, one or more tablet computers, one or more laptop computers, one or more desktop computers, and / or one or more audio devices with computing capabilities, etc. Figure 6 The box drawn around computing device 600 is not intended to limit computing device 600 to a single physical unit (e.g., having a single physical housing).

[0084] In the illustrated example, computing device 600 may include processor 601, which itself may be physically implemented by one or more processors (such as one or more microprocessors, CPUs, MCUs, etc.). Computing device 600 may also include data storage device 602, which may be implemented as one or more computer-readable media (which may be non-transitory), such as one or more memories (e.g., RAM, ROM, FLASH, etc.), hard disk drives, removable drives, memory sticks, etc. Computing device 600 may also include input devices 603, such as a touch interface for a display, a mouse, a keyboard, a voice control device, etc. Computing device 600 may also include a device controller 604 for controlling output devices such as display device 605. Display device 605 may be touch-sensitive, in which case display device 605 may also serve as input device 603. Computing device 600 may also include a data interface 606 for communicating with one or more devices external to computing device 600. For example, data interface 606 may include a USB interface, an XLR interface, and / or a wireless interface (e.g., WiFi, Bluetooth, NFC, infrared). As another example, data interface 606 may implement an XLR interface for another microphone 201, a USB interface for device 202, an XLR interface for XLR-compatible device 203, a USB interface 421 for USB connector 106b, and / or an XLR interface for XLR connector 106a.

[0085] Data storage device 602 may store computer-executable instructions that, when executed by processor 601, may cause computing device 600 to perform various steps. For example, when executed by any processor of microphone 100, the instructions may cause microphone 100 to perform any or all of steps 805 to 809, and / or perform any other function of microphone 100. As another example, when executed by any processor of device 202, the instructions may cause device 202 to perform any or all of steps 802 to 804, and / or perform any other function of device 202.

[0086] The data storage device 602 may also store other data. For example, when the data storage device 602 is part of the device 202, it may store its operating system and / or software applications that execute steps 802 to 804, user preferences such as preferred DSP modes, a list of microphones (such as microphone 100) previously configured by the software application, communication protocol settings, and / or data supporting any other functions of the device 202. When the data storage device 602 is part of the microphone 100, it may, for example, embody a control register 420 and / or store any data used to select and configure DSP modes, any other settings of the DSP 403, communication protocol settings, and / or data supporting any other functions of the microphone 100.

[0087] Although examples have been described above, the features and / or steps of these examples can be combined, divided, omitted, rearranged, modified, and / or expanded in any desired manner. Various changes, modifications, and improvements will readily occur to those skilled in the art. Although not expressly stated herein, such changes, modifications, and improvements are intended to be part of this description and are intended to fall within the spirit and scope of this disclosure. Therefore, the foregoing description is merely illustrative and not limiting.

Claims

1. A microphone, comprising: A microphone element configured to detect sound; Digital signal processor, the digital signal processor being configured to: The first analog audio signal based on the sound is processed according to a selected digital signal processing mode from a plurality of digital signal processing modes, wherein each of the plurality of digital signal processing modes is used to process the first analog audio signal in a different manner. Based on the selected digital signal processing mode among the plurality of digital signal processing modes, an automatic level control gain target value is determined from a plurality of automatic level control gain target values, and each automatic level control gain target value is associated with a different digital signal processing mode among the plurality of digital signal processing modes; Based on the automatic level control gain target value, automatic level control is performed on the first analog audio signal; as well as Based on the first analog audio signal and according to the selected digital signal processing mode among the plurality of digital signal processing modes, digital signal processing is performed to generate a digital audio signal; and A first connector is configured to output the digital audio signal.

2. The microphone of claim 1, wherein each of the plurality of digital signal processing modes includes different combinations of microphone position settings and tone settings.

3. The microphone of claim 1, wherein each of the plurality of digital signal processing modes is associated with a different setting including at least one of the following: gain control setting, equalization setting, desiring setting, bass setting, limiter setting, or audio compression setting.

4. The microphone according to claim 1, wherein the microphone is a directional dynamic microphone.

5. The microphone of claim 1, further comprising a body including a capacitive touch interface configured to allow manual adjustment of at least one setting of the microphone, wherein the body includes at least a portion having a curved external shape, and wherein the capacitive touch interface is curved in accordance with the curved external shape.

6. The microphone of claim 1, wherein the first connector is a Universal Serial Bus (USB) connector.

7. The microphone of claim 1, wherein each of the plurality of digital signal processing modes is associated with a different target gain.

8. The microphone of claim 1, wherein the digital signal processor is configured to select the selected digital signal processing mode among the plurality of digital signal processing modes based on a control signal received via the first connector.

9. The microphone of claim 1, wherein the first connector is a Universal Serial Bus (USB) connector, and further comprises an XLR connector configured to receive a second analog audio signal, wherein the microphone is configured to output a digital signal based on both the second analog audio signal received via the XLR connector and the first analog audio signal via the USB connector.

10. The microphone of claim 1, wherein the digital signal processor is further configured to adjust one or more parameters of the selected digital signal processing mode among the plurality of digital signal processing modes based on a target gain.

11. A signal processing method performed by a microphone, comprising: Sound is detected using a microphone; The first analog audio signal based on the sound is processed according to a selected digital signal processing mode from a plurality of digital signal processing modes, wherein each of the plurality of digital signal processing modes is used to process the first analog audio signal in a different manner. Based on the selected digital signal processing mode among the plurality of digital signal processing modes, an automatic level control gain target value is determined from a plurality of automatic level control gain target values, and each automatic level control gain target value is associated with a different digital signal processing mode among the plurality of digital signal processing modes; Based on the automatic level control gain target value, automatic level control is performed on the first analog audio signal; Based on the first analog audio signal and according to the selected digital signal processing mode among the plurality of digital signal processing modes, digital signal processing is performed to generate a digital audio signal; as well as The digital audio signal is output via the first connector of the microphone.

12. The method of claim 11, wherein each of the plurality of digital signal processing modes includes different combinations of microphone position settings and tone settings.

13. The method of claim 11, wherein each of the plurality of digital signal processing modes is associated with a different setting including at least one of the following: gain control setting, equalization setting, desiring setting, bass setting, limiter setting, or audio compression setting.

14. The method of claim 11, wherein the microphone is a directional dynamic microphone.

15. The method of claim 11, further comprising adjusting at least one setting of the microphone based on user input via a capacitive touch interface of the microphone.

16. The method of claim 11, wherein the first connector is a Universal Serial Bus (USB) connector.

17. The method of claim 11, wherein each of the plurality of digital signal processing modes is associated with a different target gain.

18. The method of claim 11, further comprising selecting the selected digital signal processing mode among the plurality of digital signal processing modes based on a control signal received via the first connector.

19. The method of claim 11, wherein the first connector is a Universal Serial Bus (USB) connector, wherein the microphone further includes an XLR connector configured to receive a second analog audio signal, and wherein the method further includes outputting a digital signal based on both the second analog audio signal and the first analog audio signal received via the XLR connector via the USB connector.

20. The method of claim 11, further comprising adjusting one or more parameters of the selected one of the plurality of digital signal processing modes based on a target gain.

Citation Information

Patent Citations

  • USB microphone converter

    CN209170600U

  • Phantom powered audio dynamics processor

    GB2488182A

  • Recording / playback device

    JP1993250801A

  • Handy microphone apparatus with touch sensor function

    JP2007325141A

  • Adaptive gain control system

    US20070003078A1