Audio acquisition device for digital interphone

By using a combination of microphone arrays and other components in a digital intercom, the problem of insufficient sensitivity and anti-interference capability of the audio acquisition device is solved, and high-quality audio signal acquisition and processing is achieved, which is suitable for complex environments.

CN222263005UActive Publication Date: 2024-12-27QUANZHOU DECHUNHE ELECTRONIC TECH CO LTD
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Patent Information

Application Number
CN202420893272.8
Authority / Receiving Office
CN · China
Patent Type
Utility models(China)
Current Assignee / Owner
Filing Date
2024-04-27
Publication Date
2024-12-27
Estimated Expiration
2034-04-27

AI Technical Summary

Technical Problem

The existing digital intercom audio acquisition devices have shortcomings in terms of sensitivity, anti-interference ability and audio fidelity, especially in noisy environments, and there are distortion problems during audio conversion and processing.

Method used

Microphone array technology is used to combine low-pass filters, amplifiers, analog-to-digital converters, digital signal processors, adaptive gain controllers and output adjustment modules to realize efficient acquisition, filtering, amplification, conversion and processing of sound signals, and enhance the efficiency of the sound signal. The clarity and stability of the audio signal.

Benefits of technology

It improves the sensitivity and accuracy of audio acquisition, effectively filters out high-frequency noise, ensures the purity and stability of audio signals, and realizes high-quality audio output, which is suitable for various complex environments.

✦ Generated by Eureka AI based on patent content.

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Abstract

The utility model relates to the technical field of communication, in particular to an audio acquisition device for a digital interphone, the interphone is internally provided with an acquisition device, the acquisition device comprises a microphone array which is composed of at least 3 + 2n microphones, n is any one of 1, 2 and 3; and the plurality of microphones are annularly arranged around one microphone at equal intervals. According to the audio acquisition device for the digital interphone, omnibearing sound acquisition is realized by adopting the microphone array, so that the sensitivity and the accuracy are improved; the low-pass filter effectively filters high-frequency noise to ensure that the audio is pure; the adjustable gain and sampling rate setting can adapt to different environments and sound intensities; the high-quality audio output is realized through the strong processing capability of the digital signal processor; the adaptive gain controller and the output adjustment module further optimize the audio signal; the interface and the power management device provide stable connection and power supply, and on the whole, the device can meet audio acquisition requirements in various complex environments.
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Description

Technical Field

[0001] The utility model relates to the technical field of communication, in particular to an audio acquisition device for a digital intercom. Background Art

[0002] With the rapid development of communication technology, digital walkie-talkies have been widely used in various fields as an efficient and convenient communication tool. As the core component of digital walkie-talkies, the performance of audio acquisition devices directly affects the communication quality and user experience. However, the existing audio acquisition devices still have certain shortcomings in terms of sensitivity, anti-interference ability and audio fidelity. For example, some devices are difficult to accurately capture sound signals in noisy environments, resulting in a decrease in communication quality; other devices have distortion problems during audio conversion and processing, affecting the audio fidelity. Therefore, how to design an audio acquisition device for digital walkie-talkies with high performance, high sensitivity and high anti-interference ability has become a problem that needs to be solved urgently. Utility Model Content

[0003] The purpose of the utility model is to solve the above technical problems. The utility model provides an audio acquisition device for a digital intercom, which adopts advanced microphone array technology, combined with components such as low-pass filters, amplifiers, analog-to-digital converters, digital signal processors, adaptive gain controllers and output adjustment modules, to achieve efficient acquisition, filtering, amplification, conversion and processing of sound signals.

[0004] In order to achieve the above-mentioned purpose, the utility model specifically adopts the following technical solutions:

[0005] The utility model proposes an audio collection device for a digital intercom, wherein the intercom is provided with a collection device, and the collection device comprises:

[0006] The microphone array is composed of at least 3+2n microphones, where n is any one of 1, 2, and 3; multiple microphones are arranged in a ring around a microphone at equal intervals, and this layout can receive sound signals in all directions, thereby improving the sensitivity and accuracy of audio collection; the frequency response range of the microphone array is 20Hz-20kHz, and the signal-to-noise ratio is not less than 70dB, thereby ensuring the clarity and purity of the audio signal;

[0007] A low-pass filter is connected to the microphone array and is used to filter high-frequency noise in the sound signal; the low-pass filter has an adjustable cutoff frequency range, specifically 300 Hz to 3.4 kHz, which can be adjusted according to actual needs to filter out unnecessary noise;

[0008] an amplifier connected to the low-pass filter and used to enhance the amplitude of the filtered sound signal; the amplifier has an adjustable gain range of at least 10 dB to 60 dB and can be adaptively adjusted according to the strength of the sound signal to ensure the stability of the signal strength;

[0009] An analog-to-digital converter, connected to the amplifier, for converting the amplified analog sound signal into a digital sound signal; the sampling rate of the analog-to-digital converter is not less than 48 kHz, and the bit number is not less than 24 bits, thereby ensuring high fidelity and high resolution of the audio signal;

[0010] A digital signal processor, connected to the analog-to-digital converter, is used to further process the digital sound signal; the digital signal processor has a main frequency of at least 100 MHz and has strong computing power, and can achieve functions such as noise suppression and signal enhancement to improve the quality of the audio signal;

[0011] An adaptive gain controller, connected to the digital signal processor, is used to automatically adjust the gain level according to the strength of the sound signal; through the function of the adaptive gain controller, the stability and consistency of the audio signal can be ensured;

[0012] The output adjustment module is connected to the adaptive gain controller to make the final adjustment and optimization of the audio signal after gain; by adjusting the volume, sound quality and other parameters, the audio signal can be made more in line with the usage requirements.

[0013] As a preferred technical solution of the utility model, the acquisition device also includes an interface and power management device; the interface and power management device is responsible for providing power and managing the connection and communication between the microphone array, low-pass filter, amplifier, analog-to-digital converter, digital signal processor, adaptive gain controller, and output adjustment module; the interface in the interface and power management device is I²S or PDM in the digital audio interface, which is convenient for connection and data transmission with other audio devices; the power management device is a TPS65023 power management module, which has 3 DC / DC converters, 3 LDOs, I2C interface and DVS technology power management functions, which can provide stable power supply and effectively manage power distribution between components.

[0014] The beneficial effects of the utility model are as follows:

[0015] The audio acquisition device for digital intercom of the utility model adopts a microphone array to realize omnidirectional sound acquisition, which improves sensitivity and accuracy; the low-pass filter effectively filters out high-frequency noise to ensure audio purity; the adjustable gain and sampling rate settings can adapt to different environments and sound intensities; the powerful processing capability of the digital signal processor realizes high-quality audio output; the adaptive gain controller and output adjustment module further optimize the audio signal; the interface and power management device provide stable connection and power supply. Overall, the device can be applied to audio acquisition needs in various complex environments. BRIEF DESCRIPTION OF THE DRAWINGS

[0016] Figure 1 It is a structural schematic diagram of the utility model;

[0017] Figure numerals: 1-interphone body, a-collection device, a1-microphone array, a2-low-pass filter, a3-amplifier, a4-analog-to-digital converter, a5-digital signal processor, a6-adaptive gain controller, a7-output adjustment module. DETAILED DESCRIPTION

[0018] In this technical solution:

[0019] In order to make the purpose, technical solution and advantages of the utility model more clear, the utility model is further described in detail below in conjunction with the accompanying drawings and embodiments. It should be understood that the specific embodiments described here are only used to explain the utility model and are not used to limit the utility model.

[0020] like Figure 1 As shown, the utility model proposes: an audio acquisition device for a digital intercom, the intercom is provided with an acquisition device, and the acquisition device comprises:

[0021] The microphone array is composed of at least 3+2n microphones, where n is any one of 1, 2, and 3; multiple microphones are arranged in a ring around a microphone at equal intervals, and this layout can receive sound signals in all directions, thereby improving the sensitivity and accuracy of audio collection; the frequency response range of the microphone array is 20Hz-20kHz, and the signal-to-noise ratio is not less than 70dB, thereby ensuring the clarity and purity of the audio signal;

[0022] A low-pass filter is connected to the microphone array and is used to filter high-frequency noise in the sound signal; the low-pass filter has an adjustable cutoff frequency range, specifically 300 Hz to 3.4 kHz, which can be adjusted according to actual needs to filter out unnecessary noise;

[0023] an amplifier connected to the low-pass filter and used to enhance the amplitude of the filtered sound signal; the amplifier has an adjustable gain range of at least 10 dB to 60 dB and can be adaptively adjusted according to the strength of the sound signal to ensure the stability of the signal strength;

[0024] An analog-to-digital converter, connected to the amplifier, for converting the amplified analog sound signal into a digital sound signal; the sampling rate of the analog-to-digital converter is not less than 48 kHz, and the bit number is not less than 24 bits, thereby ensuring high fidelity and high resolution of the audio signal;

[0025] A digital signal processor, connected to the analog-to-digital converter, is used to further process the digital sound signal; the digital signal processor has a main frequency of at least 100 MHz and has strong computing power, and can achieve functions such as noise suppression and signal enhancement to improve the quality of the audio signal;

[0026] An adaptive gain controller, connected to the digital signal processor, is used to automatically adjust the gain level according to the strength of the sound signal; through the function of the adaptive gain controller, the stability and consistency of the audio signal can be ensured;

[0027] The output adjustment module is connected to the adaptive gain controller to make the final adjustment and optimization of the audio signal after gain; by adjusting the volume, sound quality and other parameters, the audio signal can be made more in line with the usage requirements.

[0028] Among them, the acquisition device also includes an interface and power management device; the interface and power management device is responsible for providing power and managing the connection and communication between the microphone array, low-pass filter, amplifier, analog-to-digital converter, digital signal processor, adaptive gain controller, and output adjustment module; the interface in the interface and power management device is I²S or PDM in the digital audio interface, which is convenient for connection and data transmission with other audio devices; the power management device is a TPS65023 power management module, which has 3 DC / DC converters, 3 LDOs, I2C interface and DVS technology power management functions, which can provide stable power supply and effectively manage power distribution between components.

[0029] Working principle:

[0030] When the audio collection device for a digital intercom of the utility model is working, it first receives the sound signal in the environment through the microphone array. Since the microphone array adopts a plurality of microphones arranged in a ring with equal spacing, it can capture the sound in all directions, thereby improving the sensitivity and accuracy of the sound collection.

[0031] Subsequently, the received sound signal will be pre-processed by a low-pass filter. The main function of the low-pass filter is to filter out high-frequency noise in the sound signal to ensure the purity of the audio. The cutoff frequency of the low-pass filter can be adjusted according to actual needs to cope with noise conditions in different environments.

[0032] The filtered sound signal will enter the amplifier for signal enhancement. The amplifier will adaptively adjust the gain level according to the strength of the sound signal to ensure the stability of the signal strength.

[0033] Next, the enhanced analog sound signal will be converted into a digital sound signal by an analog-to-digital converter. The high sampling rate and bit number of the analog-to-digital converter ensure the high fidelity and high resolution of the audio signal.

[0034] The digital sound signal then enters the digital signal processor for further processing. The digital signal processor uses its powerful computing power to achieve functions such as noise suppression and signal enhancement to improve the quality of the audio signal;

[0035] The processed audio signal is then passed through an adaptive gain controller for gain adjustment. The adaptive gain controller can automatically adjust the gain level according to the strength of the sound signal to ensure the stability and consistency of the audio signal.

[0036] Finally, the audio signal after gain adjustment will enter the output adjustment module for final adjustment and optimization. The output adjustment module can adjust parameters such as volume and sound quality to make the audio signal more in line with usage requirements;

[0037] At the same time, the interface and power management device is responsible for providing a stable power supply for the entire acquisition device and managing the connection and communication between the components. The digital audio interface (I²S or PDM) enables the acquisition device to easily connect and transmit data with other audio devices, while the TPS65023 power management module ensures the stability and efficient management of the power supply.

[0038] The audio acquisition device for digital intercom of the utility model adopts a microphone array to realize omnidirectional sound acquisition, which improves sensitivity and accuracy; the low-pass filter effectively filters out high-frequency noise to ensure audio purity; the adjustable gain and sampling rate settings can adapt to different environments and sound intensities; the powerful processing capability of the digital signal processor realizes high-quality audio output; the adaptive gain controller and output adjustment module further optimize the audio signal; the interface and power management device provide stable connection and power supply. Overall, the device can be applied to audio acquisition needs in various complex environments. Embodiment 1

[0039] This embodiment provides an audio acquisition device for a digital intercom, which mainly includes a microphone array, a low-pass filter, an amplifier, an analog-to-digital converter, a digital signal processor, an adaptive gain controller, an output adjustment module, and an interface and power management device;

[0040] Specifically, the microphone array consists of five microphones arranged in a ring. This layout enables the device to capture sound signals in all directions, and the sound from the front and the side can be accurately collected. The frequency response range of the microphone array is 20Hz to 20kHz, and the signal-to-noise ratio reaches 75dB, ensuring the clarity and purity of the audio signal.

[0041] The low-pass filter is set to have a fixed cutoff frequency range of 500 Hz to 2.5 kHz; this range can effectively filter out high-frequency noise, making the collected sound signal purer;

[0042] The amplifier is connected to a low-pass filter, and its gain is adjustable from 10 dB to 40 dB. According to the strength of the input sound signal, the amplifier automatically adjusts the gain to ensure the stability of the output signal.

[0043] The analog-to-digital converter uses 24-bit sampling and the sampling rate is set to 48 kHz, which can convert analog sound signals into digital signals with high precision, providing a high-quality data foundation for subsequent processing;

[0044] The main frequency of the digital signal processor is 120 MHz, which can quickly process digital sound signals and realize functions such as noise suppression and signal enhancement. Through the processing of the digital signal processor, the quality of the audio signal has been significantly improved;

[0045] The adaptive gain controller automatically adjusts the gain level according to the signal strength after being processed by the digital signal processor, ensuring the stability and consistency of the output audio signal;

[0046] The output adjustment module makes the final adjustment to the audio signal after gain, including the optimization of volume and sound quality, so that the audio signal better meets the user's usage requirements;

[0047] The interface and power management device adopts I²S interface, which is convenient for connection and data transmission with other audio devices; the power management device adopts TPS65023 module to provide stable power supply for the entire device and effectively manage the power distribution between components; Embodiment 2

[0048] The audio acquisition device for a digital intercom provided in this embodiment is similar to that in the first embodiment, but differs in the configuration and parameters of some components;

[0049] In this embodiment, the microphone array is composed of 7 microphones, which are also arranged in a ring shape; such a design makes the audio collection more comprehensive, especially in a noisy environment, and can better capture the target sound signal;

[0050] The cutoff frequency range of the low-pass filter is set to be adjustable. Users can select the appropriate cutoff frequency range according to the actual use environment, and flexibly adjust it from 300 Hz to 3.4 kHz;

[0051] The adjustable gain range of the amplifier is expanded to 10 dB to 60 dB to accommodate a wider range of sound signal strength changes; this ensures the stability of the output signal in both low and high sound environments;

[0052] The sampling rate of the analog-to-digital converter has been increased to 96 kHz, further improving the resolution and fidelity of the audio signal; this makes the collected audio signal more delicate and able to restore more sound details;

[0053] The main frequency of the digital signal processor has been increased to 150 MHz to meet the processing requirements of higher performance; with faster processing speed, the digital signal processor can better implement functions such as noise suppression and signal enhancement, further improving the quality of audio signals;

[0054] The configurations of other components such as the adaptive gain controller, the output adjustment module, and the interface and power management device are the same as those in the first embodiment, and the stability and reliability of the audio acquisition device are also ensured;

[0055] Through the description of the above two embodiments, it can be seen that the audio acquisition device for a digital intercom provided by the utility model has the characteristics of high performance, high sensitivity and high anti-interference ability, and is suitable for audio acquisition needs in various environments.

[0056] The above shows and describes the basic principles and main features of the utility model and the advantages of the utility model. For those skilled in the art, it is obvious that the utility model is not limited to the details of the above exemplary embodiments, and can be implemented in other specific forms without departing from the spirit or basic features of the utility model. Therefore, no matter from which point of view, the embodiments should be regarded as exemplary and non-restrictive. The scope of the utility model is limited by the attached claims rather than the above description, and it is intended to include all changes within the meaning and scope of the equivalent elements of the claims. Any figure mark in the claims should not be regarded as limiting the claims involved.

[0057] In addition, it should be understood that although the present specification is described according to implementation modes, not every implementation mode contains only one independent technical solution. This description of the specification is only for the sake of clarity. Those skilled in the art should regard the specification as a whole. The technical solutions in each embodiment may also be appropriately combined to form other implementation modes that can be understood by those skilled in the art.

Claims

1. An audio acquisition device for a digital intercom, wherein the intercom (1) is provided with an acquisition device (a), characterized in that: The collection device (a) comprises: A microphone array (a1) is composed of at least 3+2n microphones, where n is any one of 1, 2, and 3, wherein a plurality of microphones are arranged in a ring around a microphone at equal intervals, and is used to receive sound signals; A low-pass filter (a2), connected to the microphone array (a1), is used to filter high-frequency noise in the sound signal; an amplifier (a3) ​​connected to the low-pass filter (a2) and having an adjustable gain ranging from at least 10 dB to 60 dB for increasing the amplitude of the filtered sound signal; an analog-to-digital converter (a4), connected to the amplifier (a3), having a sampling rate of not less than 48 kHz and a bit number of not less than 24 bits, and used for converting the amplified analog sound signal into a digital sound signal; A digital signal processor (a5), connected to the analog-to-digital converter (a4), having a main frequency of at least 100 MHz, for further processing the digital sound signal, including noise suppression and signal enhancement; An adaptive gain controller (a6), connected to the digital signal processor (a5), for automatically adjusting the gain level according to the strength of the sound signal; The output adjustment module (a7) is connected to the adaptive gain controller (a6) and is used to make final adjustments and optimizations to the audio signal after gain.

2. The audio acquisition device for a digital intercom according to claim 1, characterized in that: The acquisition device (a) also includes an interface and power management device for providing power and managing the connection and communication between the microphone array (a1), the low-pass filter (a2), the amplifier (a3), the analog-to-digital converter (a4), the digital signal processor (a5), the adaptive gain controller (a6), and the output adjustment module (a7).

3. The audio acquisition device for a digital intercom according to claim 1, characterized in that: The frequency response range of the microphone array (a1) is 20Hz-20kHz, and the signal-to-noise ratio is ≥70dB.

4. The audio acquisition device for a digital intercom according to claim 1, characterized in that: The low pass filter (A2) has an adjustable cut-off frequency range of 300 Hz to 3.4 kHz.

5. The audio acquisition device for a digital intercom according to claim 2, characterized in that: The interface in the interface and power management device is one of I²S or PDM in the digital audio interface, and the power management device in the interface and power management device is a TPS65023 power management module, which has 3 DC / DC converters, 3 LDOs, I2C interface and power management of DVS technology.