Development
DIGITAL MICROPHONE WITH
FAST AGC AND
SENSITIVITY ADJUSTMENT
MICROPHONE FOR VOICE
The STELBERRY M-50 is a completely new solution for audio recording systems and the best voice microphone in its class. High-speed digital signal processing effectively isolates the speech range, significantly reducing unnecessary sounds in the low and high frequencies.
The STELBERRY M-50 is equipped with a dual digital Automatic Gain Control system with a response speed of less than one thousandth of a second.
An external regulator allows you to adjust the sensitivity of the digital microphone for any operating conditions.
IP MICROPHONE The STELBERRY M-50 digital microphone is ideal for connecting to the line input of IP cameras, ideally conveying the acoustic picture.
environment
This application actually makes it a full-fledged IP microphone. Also, a definite plus this decision
, is the ability to install a digital microphone anywhere, regardless of the location of the IP camera.
| Comparison table of models of omnidirectional microphones of the STELBERRY M series | |||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|
| Characteristics and parameters of omnidirectional microphones | ✔ | ➖ | ✔ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ |
| Fixed sensitivity value | ➖ | ✔ | ➖ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ |
| Adjustable sensitivity | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Sensitivity setting method | Resistor | Resistor |
| Joystick | ➖ | ➖ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ |
| AGC - automatic gain control | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ |
| Ability to change AGC speed | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ |
| Possibility to disable AGC | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ | ✔ | ✔ | ✔ | ➖ | ➖ |
| Switchable low impedance output for audio inputs of a range of IP cameras | 100...6100 | 100...7200 | 100...8300 | 100...9200 | 270...4000 | 80...16000 | 80...16000 | 270...4000 | 270...4000 | 80...16000 | 80...16000 |
| Maximum Bandwidth (Hz) | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ | ✔ |
| Bandwidth adjustable | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ |
| Ability to cut a frequency selected from a set of frequencies | 48 | 48 | 48 | 48 | 48 | 63 | 63 | 63 | 63 | 67 | 67 |
| Signal to noise ratio (dB) | 8 | 10 | 10 | 12 | 20 | 20 | 20 | 20 | 20 | 25 | 25 |
| Acoustic range (meters) | ➖ | ➖ | Audio processing | Audio processing | digital | Audio processing | Audio processing | digital | digital | digital | digital |
| Lock settings | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ | ✔ |
| Output level (V) | 1 | 1 | 1 | 1 | 1 | 1 | 1 | 1 | 1 | 1 | 1 |
| Maximum line length (meters) | 300 | 300 | 300 | 300 | 300 | 300 | 300 | 300 | 300 | 300 | 300 |
| Rated supply voltage (V) | 12 | 12 | 12 | 12 | 12 | 12 | 12 | 12 | 12 | 12 | 12 |
| Current consumption (mA) | 3 | 3 | 8 | 8 | 25 | 8 | 8 | 25 | 25 | 25 | 25 |
| Detachable cable connection with microphone | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ | ✔ | ✔ | ✔ | ✔ | ✔ |
| Anti-vandal housing | ➖ | ➖ | ➖ | ➖ | ➖ | ➖ | ✔ | ➖ | ✔ | ➖ | ➖ |
For reliable operation of the STELBERRY M-50 digital microphone, high-quality power supply with a low ripple level is required. The best solution is to use the STELBERRY MX-225 pass-through PoE splitter, which has an output voltage filtering system. Also, STELBERRY MX-225 has built-in protection against short circuit at the output or exceeding the maximum permissible current.
The miniature pass-through PoE splitter STELBERRY MX-225 is installed in the cable cut that connects the IP camera and the switch and can be glued to any surface or hidden inside the box through which the cable is laid. To connect power to the STELBERRY M-50 digital microphone, the PoE splitter is equipped with self-clamping connectors that ensure reliable contact.
FAST DIGITAL
SIGNAL PROCESSOR
A miniature digital signal processor (DSP) digitizes the audio signal from the audio capsule at a sampling rate of 44,100 Hz and 16-bit sampling.
Distinctive feature The processor is the presence of 2-speed AGC, providing lightning-fast automatic gain control, both at the input and output of the device.
6 digital filters of the processor process the signal in such a way that only the speech range remains at the linear output.
A precision built-in preamplifier guarantees a high signal-to-noise ratio.
CONTROL PROCESSOR
DIGITAL MICROPHONE
The central control processor of the STELBERRY M-50 digital microphone provides microphone gain adjustment and control of signal processing parameters.
The processor guarantees that the microphone quickly returns to operating mode after power is applied, thanks to a high-speed exchange line with the signal processor.
WIND PROTECTION FOR DIGITAL MICROPHONE
STELBERRY M-50
For ideal sound transmission, the digital microphone is equipped with a wind filter.
By eliminating the wind component, a filter made of acoustic material cuts off unwanted sounds that arise when wind flows collide with a sensitive membrane, resulting in crystal-clear sound.
The presence of wind protection allowed us to create an effective microphone for voice.
OPTIMIZING THE MICROPHONE UNDER SPEECH
RANGE
The bandwidth of the STELBERRY M-50 digital microphone is tuned to the frequency range of human speech and lies within the range of 270...4000 Hz.
This bandwidth ensures excellent speech intelligibility, regardless of extraneous noise sources.
Signal processing is carried out by six digital high-speed filters, which guarantees a high slope of the amplitude-frequency response in the low and high frequency range.
DOUBLE AGC SYSTEM
The microphone is equipped with two digital high-speed Automatic Gain Controls (AGC).
The first AGC controls the gain at the microphone input, immediately after the signal from the capsule is digitized, and the response speed to changes in sound level is less than 1/1000 of a second.
This allows you to react to any, even the most minor changes in the sound environment.
The second AGC processes the signal at the microphone output, reliably maintaining a stable output signal level. The response speed of the AGC output system is also less than 1/1000 of a second.
COMPARISON OF DIGITAL AUTOMATIC GAIN CONTROL (AGC) WITH ANALOG AGC
STELBERRY M-50 is a completely new solution for audio recording systems and the best voice microphone in its class. High-speed digital signal processing effectively isolates the speech range, significantly reducing unnecessary sounds in the low and high frequencies. The microphone is equipped with a dual digital Automatic Gain Control system with a response speed of less than one thousandth of a second. An external regulator allows you to adjust the sensitivity of the digital microphone for any operating conditions. IP microphone The digital microphone is ideal for connecting to the line input of IP cameras, ideally conveying the acoustic picture of the environment. This application actually makes it a full-fledged IP microphone. Also, an undoubted advantage of this solution is the ability to install a digital microphone anywhere, regardless of the location of the IP camera. Fast Digital Signal Processor A miniature digital signal processor (DSP) digitizes the audio signal from the audio capsule at a sampling rate of 44,100 Hz and 16-bit sampling. A distinctive feature of the processor is the presence of 2-speed AGC, providing lightning-fast automatic gain control, both at the input and output of the device. 6 digital filters of the processor process the signal in such a way that only the speech range remains at the linear output. Precise built-in preamp guarantees high attitude signal/noise Digital microphone control processor The digital microphone's central control processor provides microphone gain adjustment and control of signal processing parameters. The processor guarantees that the microphone quickly returns to operating mode after power is applied, thanks to a high-speed exchange line with the signal processor. Wind protection for digital microphone For ideal sound transmission, the digital microphone is equipped with a wind filter. By eliminating the wind component, a filter made of acoustic material cuts off unwanted sounds that arise when wind flows collide with a sensitive membrane, resulting in crystal-clear sound. The presence of wind protection allowed us to create an effective microphone for voice. Optimizing the microphone for the speech range The bandwidth of the digital microphone is adjusted to the frequency range of human speech and lies within the range of 270...4000 Hz. This bandwidth ensures excellent speech intelligibility, regardless of extraneous noise sources. Signal processing is carried out by six digital high-speed filters, which guarantees a high slope of the amplitude-frequency response in the low and high frequency range. Dual AGC system The microphone is equipped with two digital, high-speed Automatic Gain Controls (AGC). The first AGC controls the gain at the microphone input, immediately after the signal from the capsule is digitized, and the response speed to changes in sound level is less than 1/1000 of a second. This allows you to react to any, even the most minor changes in the sound environment. The second AGC processes the signal at the microphone output, reliably maintaining a stable output signal level. The response speed of the AGC output system is also less than 1/1000 of a second. Convenient adjustment The convenient location of the sensitivity adjustment makes it easy to adjust the microphone gain. A feature of a highly sensitive microphone is that gain adjustment occurs before AGC processing begins. This makes it easy to achieve the desired sound quality. The microphone's bandwidth is selected to allow voice frequencies to pass through, eliminating unwanted sounds from high-frequency sources.Description STELBERRY M-50
Technical characteristics of STELBERRY M-50
Digital microphone Stelberry M-50 with adjustable gain, built on a specialized processor. The microphone operation process consists of analog-to-digital conversion of the microphone capsule signal, subsequent digital filtering of the received signal, and reverse digital-to-analog conversion. The M-50's sensitive microphone has digital filters tuned to the range of human speech. Sound frequencies outside the frequency range 270...4000 Hz are significantly attenuated by the microphone. The very fast AGC (automatic gain control) of the digital microphone allows you to comfortably use it in a room with sudden changes in the volume of sound or human speech.
The M-50 digital microphone is well suited as a voice recording microphone for projects that focus on recording conversations. Ideal as an external highly sensitive microphone for video cameras and audio recorders that are sensitive to the input signal level and do not have own funds sound filtering.
The sensitive microphone Stelberry M-50 is used as an external microphone for various video surveillance cameras, including IP cameras, for audio monitoring of premises, as a highly sensitive microphone for voice recording in call recording systems and speech recognition systems.
Placement of a digital microphone with AGC Stelberry M-50 indoors
When placing the M-50 microphone in the corner of the room and setting the maximum sensitivity of the microphone, the comfortable listening zone will correspond to a quarter circle area of 50 m². With further distance from the microphone, the level of its output signal will gradually weaken down to the limit of acoustic audibility of 20 meters.
Connecting a digital microphone with AGC STELBERRY M-50 to an IP camera
The M-50 digital microphone connects directly to the audio line input of the video camera. Connecting a microphone to the camera is done in this way. The yellow wire of the M-50 microphone, to the camera’s “Jack-3.5mm” input connector, is connected to the end (central) and ring contact of the connector (Check in the camera manual.). If a camera or IP camera uses an RCA (“tulip”) connector for audio input, then go to the central contact of the RCA connector. The black wire of the M-50 digital microphone is connected to the common (body) contact of the 3.5mm Jack connector (or to the ring external contact of the RCA connector), and to the negative common wire of the stabilized power supply. The red wire of the microphone is connected to the “positive” wire of the stabilized power supply.
Directional pattern of a digital microphone with AGC and gain control Stelberry M-50
The digital speech microphone Stelberry M-50 is omnidirectional and has pie chart directionality with a slight weakening of the microphone sensitivity on the side of the sensitivity control. The polar pattern is based on the microphone capsule used in the microphone, taking into account the influence of the microphone body.
Microphones Stelberry
In recent years, digital MEMS microphones have appeared on the electronic components market. Their advantages include: high sensitivity, linearity of the frequency response in the operating frequency band, repeatability of parameters and small overall dimensions. Using a digital MEMS microphone also eliminates the problems associated with analog circuit noise and makes it possible to directly connect the microphone to the processor. These advantages interested us, and we tried to put them into practice.
At the time of the start of work, Second Laboratory LLC had several prototypes of ADMP421 microphones manufactured by Analog Devices. Then we had the SPM0405HD4H-WB digital MEMS microphones from Knowles Electronics. The results of work with the listed microphones became the basis for writing this article.
A digital microphone can be connected to an audio codec that has an appropriate interface [for example, 8–10]. But we were interested in the possibility of directly connecting a digital microphone to a microcontroller. This solution made it possible to abandon the use of an audio codec, which reduced overall dimensions and further reduced the price of the product. To make a preliminary assessment of the expected parameter values (required microcontroller performance, power consumption, sensitivity, dynamic range, SOI, operating frequency band), a small development work was performed. Based on its results, a final decision was made on the circuit design, software and element base used.
Connecting digital microphones to microcontrollers
The interface between the microcontroller and the digital microphone is simple, and information on its implementation is sufficiently posted on the manufacturers’ websites and described in detail by other authors. Typically, digital microphones have five terminals, short description which are given in the table. Electrical and timing parameters of microphone outputs are given in their specifications.
Table. Description of digital microphone pins
| № | Name output |
Short description |
| 1 | VDD | Microphone power |
| 2 | GND | "Earth" |
| 3 | CLK | Input clock signal, synchronous with which the DATA line switches its states |
| 4 | DATA | During one half of the CLK cycle, this pin is in a state of high impedance, and during the second half serves as a conclusion for reading data from the output of the Σ-Δ modulator microphone |
| 5 | L/R_Sel | This pin is used to control switching the DATA line. If L/R_Sel connected to VDD, then some time after detecting the rising edge of the CLK signal DATA pin goes high impedance, and after the arrival of the falling edge signal CLK pin DATA is connected to the output Σ-Δ microphone modulator. If L/R_Sel connected to GND, the edges of the CLK signal, along which the DATA line switches, changes to opposite |
To evaluate the required performance of the microcontroller, the ADSP-BF538 EZ KIT Lite development board from Analog Devices was used. Microphones could be connected to this board using SPI or SPORT interfaces. The first of these interfaces is more common, and therefore we used this interface in Slave mode. To generate the CLK clock signal, the hardware timer available in the microcontroller was used. To obtain output samples at a standard sampling rate of 16 kHz at a decimation factor of 128, the required CLK clock frequency must be 2.048 MHz. As a clock source for the processor on the development board, a generator with a frequency of 12.288 MHz was used, which, when divided by 6, provided the required clock frequency for a digital microphone. To minimize the load on the processor when receiving initial information from microphones, the DMA transfer mechanism was used.
During the modeling process, it was calculated and experimentally verified that to process data from a microphone, the processor must have a performance of about 8 MIPS. An assessment of the required performance allowed us to conclude that it was possible to use a simpler microcontroller with less power consumption. Of the three alternative options (ARM, PIC, MSP430), the MSP430F5418 microcontroller manufactured by Texas Instruments was selected, which has minimal power consumption (165 μA/MIPS). In the future, to check power consumption and test software The MSP-EXP430F5438 Experimenter Board from the same company was used.
In Fig. Figure 1 shows simplified diagrams for connecting digital microphones to the debug boards used in prototyping, allowing you to fully simulate devices for reading, playing or storing data from microphones.
Rice. 1. Diagram for connecting a digital microphone to the board: a) ADSP-BF538 EZ KIT Lite; b) MSP-EXP430F5438
The process of converting the input audio signal in a microphone
Rice. 2. Simplified model of a MEMS microphone
Each digital MEMS microphone can be simplified into the model shown in Fig. 2. Input sound vibrations are converted through a MEMS membrane into a weak electrical signal, which is then fed to the input of amplifier A. The pre-amplified signal then passes through an analog low-pass filter, which is necessary to protect against aliasing. The final element of signal processing in the microphone is a 4th order Σ-Δ modulator, which converts the input analog signal into a one-bit digital stream.
The frequency of data bits from the output of the Σ-Δ modulator is equal to the frequency of the input clock signal CLK and, as a rule, lies in the range from 1 to 4 MHz.
Measuring Digital Microphones
The following equipment was used to carry out the measurements: sound level meter CENTER-325, low-frequency signal generator G3-118, nonlinear distortion meter S6-11, headphone emitter Dialog M-881HV and PC. Rice. 3.
ADMP421 Microphone Frequency Response
In the time domain, the output of a Σ-Δ modulator is a jumbled collection of ones and zeros. However, if we assign a value of 1.0 to each high logic level of the microphone output, and a value of –1.0 to each low logic level, and then perform a Fourier transform, we will obtain a spectrogram of the output data from the microphone. In Fig. Figures 3 and 4 show the responses of the ADMP421 and SPM0405HD4H-WB microphones to an input sine wave audio signal with a frequency of 1 kHz and a level of 94 dB SPL. The measurements were carried out for three values of the CLK signal frequency - 512, 1024 and 2048 kHz. (To reduce the length of the published article, materials for the frequency of 1024 kHz are not given.) The spectrograms were constructed using a sample length of 128–1024 samples. Rice. 4.
Frequency response of microphone SPM0405HD4H-WB Judging by the spectrograms, quantization noise is shifted outside the audio frequency range and does not affect the input audio signal. In this case, quantization noise shifts further into the high-frequency region, the higher the sampling frequency of the microphones. Approximately the cutoff frequency from which the noise level begins to increase can be determined as F clk
You can also observe that both microphones contain a constant component in the output signal (this effect has been eliminated in the latest modifications of microphones). Moreover, the level of the constant component is comparable in level to the measured signal. In addition, the value of the constant component at least depends on the supply voltage. This property required the implementation of a recursive algorithm in the microcontroller that eliminates the constant offset.
If you compare microphones in terms of noise levels, it is easy to see that the ADMP421 microphone has best attitude signal to noise compared to the SPM0405HD4H-WB microphone is approximately 5–6 dB, as well as a lower level of quantization noise.
If we compare the levels of nonlinear distortion, we will see that the spectrograms of both microphones contain only second harmonics, despite the fact that the amplitude of the second harmonic of the Knowles Electronics microphone is significantly lower than that of the Analog Devices microphone. This fact is of particular interest, since both companies standardize only the maximum SOI and only for a certain level sound pressure. In reality, this data is not enough. For example, it is impossible to compare the actual THD values of different microphones. In addition, it is currently common practice to normalize SOI to the linear input of recording devices, without taking into account distortions introduced by microphones.
Therefore, in order to assess the nature of the dependence of SOI on sound pressure level, an experiment was carried out, which included the following steps:
- Exposing the microphone input to a sinusoidal audio signal with a frequency of 1 kHz and recording one-bit data from the microphone output to flash memory (the sound pressure of the input signal varies from 87.5 to 115 dB SPL in steps of 2.5 dB SPL).
- Mathematical processing of one-bit microphone data using a digital low-pass filter to obtain a deterministic digital signal and cut off quantization noise.
- Reproduction of processed digital data on a PC and measurement of the SOI signal from the output of a PC sound card using a nonlinear distortion meter S6-11 (nonlinear distortions introduced by the sound card itself do not exceed 0.1%).
- Registration of readings from the S6-11 device for each sound pressure value of the input audio signal.
Rice. 5. Dependence of SOI of microphones on sound pressure level
The results of the experiment are presented in Fig. 5. From the above graph it follows that at a sound pressure of less than 97 dB, the SPL THD of the ADMP421 and SPM0405HD4H-WB microphones does not exceed 1% and 0.3%, respectively. At higher sound pressures, the THD of the ADMP421 microphone is significantly higher than that of the SPM0405HD4H-WB microphone, and at pressures above 110 dB SPL, both microphones experience a sharp increase in the level of nonlinear distortion. In general, we can conclude that the Knowles Electronics microphone is suitable for use over a wider sound pressure range. It should also be noted that the SOI values of microphones given in the documentation are normalized at maximum sound pressure.
Actual THD values at lower sound pressure levels are much lower, and microphones can be used for high-quality audio recording.
However, the ADMP421 microphone has another advantage. This model of microphones is practically insensitive to noise on the power bus, even if the latter reaches values of 200–300 mV. In Fig. Figure 6 shows the case when artificially introduced impulse noise is present in the microphone power bus. This case is possible if the audio device operates in pulsed consumption mode (for example, cyclic recording of data from a microphone to flash memory when powered from a low-power source). Rice. 6.
Pulse noise in the microphone power supply circuit Rice. 7.
Time diagram of a signal from microphones when exposed to pulsed noise in the power circuit
In Fig. Figure 7 shows the output signal from microphones, passed through a digital low-pass filter with the amplitude-frequency response shown in Fig. 9. No reference audio signal was used to detect power interference during the recording process. To be able to estimate the amplitude of interference from the microphone output, in the upper part of Fig. Figure 7 shows an 80 dB SPL sine wave audio signal recorded in the absence of power supply interference. Rice. 8.
Simplified circuit of a digital signal converter Σ-Δ modulator Rice. 9.
Frequency response of a software decimator implemented on ADSP-BF538F and MSP430F5438 processors
To eliminate the influence of noise on the power supply circuits, we had to use an anti-aliasing RC filter.
To isolate the audio frequency band signal, the data from the microphone must be filtered and resampled at a reduced frequency (typically 50 to 128 times the sampling rate of the Σ-Δ modulator). A digital low-pass filter filters out external noise and the microphone's own noise outside the operating band ( f >Judging by the spectrograms, quantization noise is shifted outside the audio frequency range and does not affect the input audio signal. In this case, quantization noise shifts further into the high-frequency region, the higher the sampling frequency of the microphones. Approximately the cutoff frequency from which the noise level begins to increase can be determined as /2M) to protect against aliasing, and also makes it possible to reduce the data repetition rate. In Fig. 8 one of the possible options processing a one-bit data stream from a microphone, implemented in software on a DSP or in hardware in audio codecs.
Shown in Fig. 8, the sampling frequency compression circuit (compressor) lowers the sampling frequency due to the fact that from each M filtered signal samples w(mM) is discarded M–1 sample. The input and output of the converter shown in Fig. 8 are related by the following expression:
When implementing frequency converters in software, both FIR and IIR filters can be used as a digital low-pass filter. Developers should be very careful when choosing the type of filter, its length and bit depth, since the performance of the entire system as a whole directly depends on this. A correctly calculated and implemented decimator (frequency converter) in some cases will significantly reduce the cost of products and increase it specifications. As a reference, we note that during the development of the Soroka-1 and Soroka-2 voice recorders, software decimators that reduce the frequency by 64 times (from 1.024 MHz to 16 kHz) were successfully implemented both on the high-performance ADSP-BF538F processor and and on the MSP430F5438 microcontroller with an operating clock frequency of 12.288 MHz. The amplitude-frequency response of the digital low-pass filter included in the implemented decimator is shown in Fig. 9. For complete information on practical issues of digital filtering, please refer to chapters 6–9 of the book.
As a second option, audio codecs adapted for this can be used to convert data from the output of a digital microphone, which will significantly reduce product development time. For example, Analog Devices suggests using ADAU1361 and ADAU1761 codecs, which are equally suitable for ADMP421 and SPM0405HD4H microphones.
Measuring the frequency response for the operating frequency band with the required accuracy turned out to be quite a difficult task due to the lack in the laboratory of an acoustic emitter with a linear amplitude response to sound pressure. Estimates of the resulting frequency response show its linearity in the operating frequency band with an error of about ±4 dB. Therefore, when assessing the linearity of the frequency response, we considered it correct to rely on the declared characteristics of the manufacturers and the calculated characteristics of low-frequency filters with ripple in a passband of less than 1 dB.
MEMS microphones open up new possibilities for audio equipment developers. The process of creating digital audio devices becomes simple in terms of hardware implementation and complex in terms of writing programs for the microcontrollers used. We hope that the information on methods and parameters provided in this article will be of interest to many engineers.
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