Digital microphones: from specification to finished product. Digital Microphone Blue Microphones Raspberry Studio (Red) Digital Microphones

In recent years, digital MEMS microphones have entered the electronic component market. Their advantages include: high sensitivity, linearity of the frequency response in the operating frequency band, repeatability of parameters and small overall dimensions. The use of a digital MEMS microphone also eliminates the problems associated with analog circuit noise and allows direct connection of the microphone to the processor. We were interested in these advantages, and we tried to implement them in practice.

At the time of the start of work, the Second Laboratory LLC had several prototypes of ADMP421 microphones manufactured by Analog Devices. Then we got the SPM0405HD4H-WB digital MEMS microphones from Knowles Electronics. The results of work with the listed microphones became the basis for writing this article.

The digital microphone can be connected to an audio codec that has an appropriate interface [eg 8-10]. But we were interested in the possibility of directly connecting a digital microphone to the microcontroller. This solution made it possible to abandon the use of an audio codec, which reduced the overall dimensions and further reduced the price of the product. For a preliminary assessment of the expected values \u200b\u200bof the parameters (required performance of the microcontroller, power consumption, sensitivity, dynamic range, THD, bandwidth), a small R&D was performed. Based on its results, the final decision was made on the circuitry, software and the 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 available in sufficient volume on the manufacturers' websites and is described in detail by other authors. Typically, digital microphones have five pins, short description which are given in the table. The electrical and timing parameters of the microphone leads are given in their specifications.

Table. Pin Descriptions for Digital Microphones

Name
withdrawal
Short description
1 VDD Microphone power
2 GND "Earth"
3 CLK Input timing signal, synchronously with which
DATA line toggles its states
4 DATA During one half of the CLK cycle, this output
is in a high impedance state,
and during the second half serves as a conclusion
for reading data from the Σ-Δ modulator output
microphone
5 L / R_Sel This pin is used to control
by 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
CLK signal, the DATA pin is connected to the
Σ-Δ microphone modulator. In case L / R_Sel
connected to GND, the edges of the CLK signal, along which
DATA line toggles, change to
opposite

To assess the required performance of the microcontroller, we used an ADSP-BF538 EZ KIT Lite debug board from Analog Devices. Microphones could be connected to this board using SPI or SPORT interfaces. The first of the named interfaces is more common, and therefore we used this interface in Slave mode. The hardware timer available in the microcontroller was used to generate the CLK clock signal. To obtain output samples with a standard sampling rate of 16 kHz with a decimation factor of 128, the required CLK clock frequency must be 2.048 MHz. The development board used a 12.288 MHz oscillator as the clock source for the processor on the development board, providing the required clock rate for the digital microphone when divided by 6. To minimize the load on the processor when receiving the initial information from the microphones, the DMA transfer mechanism was used.

In the course of modeling, it was calculated and experimentally verified that for processing data from a microphone, the processor must have a performance of about 8 MIPS. Evaluation of the required performance allowed us to conclude that it is possible to use a simpler microcontroller with less power consumption. Of the three alternatives (ARM, PIC, MSP430), the MSP430F5418 microcontroller from Texas Instruments was chosen, which has a minimum power consumption (165 μA / MIPS). In the future, to check the power consumption and work off software we used the MSP-EXP430F5438 Experimenter Board from the same company.

In fig. 1 shows simplified diagrams for connecting digital microphones to the debug boards used for prototyping, which allow you to completely simulate devices for reading, playing or storing data from microphones.

Figure: 1.Diagram for connecting a digital microphone to the board: a) ADSP-BF538 EZ KIT Lite; b) MSP-EXP430F5438

Microphone input audio conversion process

Figure: 2. Simplified MEMS Microphone Model

Each digital MEMS microphone can be simplified as the model shown in Fig. 2. The input sound vibrations are converted by means of a MEMS membrane into a weak electrical signal, which is then fed to the input of amplifier A. Then the preamplified signal 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 Σ-Δ modulator of the 4th order, which converts the input analog signal into a one-bit digital stream. The bit rate of data 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

To carry out the measurements, the following equipment was used: a CENTER-325 sound level meter, a G3-118 low-frequency signal generator, a C6-11 nonlinear distortion meter, an emitter from Dialog M-881HV headphones and a PC.

Figure: 3.Frequency response of the ADMP421 microphone

In the time domain, the data from the output of the Σ-Δ modulator is a jumble of zeros and ones. However, if each high logic level of the microphone output is assigned a value of 1.0, and each low level is assigned a value of –1.0, and then performing the Fourier transform, we get 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 a 1 kHz sine wave input signal with 94 dB SPL. The measurements were carried out for three values \u200b\u200bof the CLK signal frequency - 512, 1024, and 2048 kHz. (To reduce the volume of the article being published, the materials for the frequency of 1024 kHz are not given.) The spectrograms were built on a sample length of 1281024 samples.

Figure: 4. Microphone frequency response SPM0405HD4H-WB

The spectrograms show that the quantization noise is shifted outside the audio frequency range and does not affect the input audio signal. In this case, the quantization noise is shifted the further into the high frequency region, the higher the sampling frequency of the microphones. The approximate cutoff frequency from which the noise level starts to increase can be defined as F clk /one hundred. Although in the specifications for microphones, the operating frequency is normalized approximately in the range from 1 to 3 MHz, but, as the spectrograms show, microphones work normally at lower clock frequencies. This can be very useful when it becomes necessary to reduce the number of calculations on the microcontroller, although, of course, this will also narrow the working sound bandwidth.

You can also observe that both microphones have a constant component in the output signal (in the latest versions of microphones this effect has been eliminated). Moreover, the level of the constant component is comparable in level with the measured signal. In addition, the DC component is at least dependent on the supply voltage. This property required the implementation in the microcontroller of a recursive algorithm that eliminates constant bias.

If we compare microphones in terms of introduced noise levels, it is easy to see that the ADMP421 microphone has a better signal-to-noise ratio compared to the SPM0405HD4H-WB microphone - by about 5-6 dB, as well as a lower level of quantization noise.

If we compare the levels of nonlinear distortion, it will be seen that only second harmonics are present in the spectrograms of both microphones, while 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 firms standardize only the maximum THD and only for a certain level of sound pressure. In reality, this data is often insufficient. For example, it is impossible to compare the actual THD values \u200b\u200bof different microphones. In addition, it is now generally accepted practice to normalize SOI to the line input of recording devices, without taking into account the distortions introduced by microphones.

Therefore, in order to assess the nature of the dependence of SOI on the sound pressure level, an experiment was set up, which included the following stages:

  1. Influencing the microphone input with a sinusoidal audio signal with a frequency of 1 kHz and recording one-bit data from the microphone output to the flash memory (the sound pressure of the input signal varies from 87.5 to 115 dB SPL in 2.5 dB SPL steps).
  2. Mathematical processing of one-bit data from a microphone using a digital low-pass filter in order to obtain a deterministic digital signal and cut off quantization noise.
  3. Reproduction of processed digital data on a PC and measurement of the THD signal from the output of a PC sound card using a C6-11 nonlinear distortion meter (nonlinear distortions introduced by the sound card itself do not exceed 0.1%).
  4. Registration of the readings of the C6-11 device for each value of the sound pressure of the input audio signal.

Figure: five.SOI microphones versus sound pressure level

The results of this experiment are shown in Fig. 5. From the above graph it follows that at a sound pressure of less than 97 dB SPL THD microphones ADMP421 and SPM0405HD4H-WB does not exceed 1% and 0.3%, respectively. At higher sound pressure, the THD of the ADMP421 microphone is significantly higher than that of the SPM0405HD4H-WB microphone, and above 110 dB SPL, both microphones experience a sharp increase in the level of harmonic distortion. In general, it can be concluded that the Knowles Electronics microphone is suitable for use over a wider sound pressure range. It should also be noted that the values \u200b\u200bof SOI microphones given in the documentation are normalized at the maximum sound pressure. The real THD values \u200b\u200bat lower sound pressure values \u200b\u200bare much lower, and microphones can be used for high quality audio recording.

However, the ADMP421 microphone has another advantage. This microphone model is practically insensitive to noise on the power bus, even if the latter reaches values \u200b\u200bof 200-300 mV. In fig. 6 shows a case when artificially introduced impulse noise is present in the microphone power bus. Such a case is possible if the audio device operates in a pulsed mode of consumption (for example, cyclic data recording from a microphone to flash memory when powered from a low-power source).

Figure: 6. Pulse noise in the microphone power supply circuit

Figure: 7. Timing diagram of a signal from microphones when exposed to impulse noise in the power circuit

In fig. 7 shows the output signal from microphones passed through a DSP filter with an amplitude-frequency characteristic shown in Fig. 9. No reference audio signal was used to register power disturbances during recording. In order to be able to estimate the amplitude of the noise from the microphone output, in the upper part of Fig. 7 shows an 80 dB SPL sinusoidal audio signal recorded in the absence of power supply noise.

Figure: 8.Simplified circuit of the digital signal converter Σ-Δ modulator

Figure: nine. Frequency response of a software decimator implemented on the ADSP-BF538F and MSP430F5438 processors

To eliminate the influence of noise in the power supply circuits, we had to use an RC anti-aliasing filter.

Data processing from digital microphone output

To isolate the audio band signal, the microphone data must be filtered and resampled at a reduced rate (typically 50 to 128 times the sampling rate of the Σ-Δ modulator). A digital low-pass filter filters out external noise and microphone intrinsic noise outside the operating band ( f >F clk /2M) in order to protect against aliasing, and also makes it possible to reduce the data repetition rate. In fig. 8 shows one of the possible options for processing a one-bit data stream from a microphone, implemented software on the DSP or hardware in audio codecs.

Shown in fig. 8 the sampling rate compression (compressor) circuit downsamples the sampling rate due to the fact that each M samples of the filtered signal w(mM) is discarded M–1 sample. The input and output of the converter shown in Fig. 8 are related by the following expression:

For software implementation of frequency converters, 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 production and increase it specifications... As a reference, we note that during the development of the "Soroka-1" and "Soroka-2" 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 characteristic of the digital low-pass filter included in the implemented decimator is shown in Fig. 9. For complete information on practical digital filtering issues, refer to chapters 6-9 of the book.

As a second option, you can use adapted audio codecs to convert data from the digital microphone output, which will significantly reduce product development time. For example, Analog Devices suggests using the ADAU1361 and ADAU1761 codecs, which are equally suitable for the ADMP421 and SPM0405HD4H microphones.

Measuring the frequency response with the required accuracy for the operating frequency band turned out to be a rather difficult task due to the absence in the laboratory of an acoustic radiator with a linear amplitude characteristic in terms of 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 the passband less than 1 dB.

MEMS microphones open up new possibilities for sound designers. The process of creating digital audio devices becomes simple in terms of hardware implementation and difficult 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.

DIGITAL MICROPHONE WITH
FAST AGC AND
ADJUSTING THE SENSITIVITY

VOICE MICROPHONE

STELBERRY M-50 is a completely new solution for sound 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.
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 control adjusts the sensitivity of the digital microphone to suit any application.

IP MICROPHONE

The STELBERRY M-50 digital microphone is ideal for connecting to the line input of IP cameras, perfectly transmitting the acoustic picture environment.
This application actually makes it a complete IP microphone.
Also, the undoubted advantage of this solution is the ability to install a digital microphone anywhere, regardless of where the IP camera is installed.

STELBERRY M Series Omni-Directional Microphone Model Comparison Chart

Omni-directional microphone characteristics and parameters
Fixed sensitivity value
Adjustable sensitivity
Sensitivity setting method Resistor Resistor Resistor Resistor Resistor Resistor Resistor Resistor Resistor Joystick Joystick
AGC - automatic gain control
The ability to change the AGC speed
Ability to disable AGC
Low impedance switchable output for audio inputs of a number of IP cameras
Maximum bandwidth (Hz) 100...6100 100...7200 100...8300 100...9200 270...4000 80...16000 80...16000 270...4000 270...4000 80...16000 80...16000
Bandwidth adjustable
Ability to cut a frequency selected from a set of frequencies
Signal to noise ratio (dB) 48 48 48 48 48 63 63 63 63 67 67
Acoustic range (meters) 8 10 10 12 20 20 20 20 20 25 25
Sound processing analog analog digital analog analog digital digital digital digital
Locking settings
Output signal 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
Consumption current (mA) 3 3 8 8 25 8 8 25 25 25 25
Detachable microphone cable connection
Vandal-proof housing

For reliable operation of the STELBERRY M-50 digital microphone, high-quality power supply with low ripple is required. The best solution is to use the STELBERRY MX-225 PoE loop-through 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 STELBERRY MX-225 miniature PoE splitter is installed in the cut of the cable that connects the IP camera and the switch and can be glued to any surface or hidden inside the cable duct. To connect the power supply of the STELBERRY M-50 digital microphone, the PoE splitter is equipped with self-tightening connectors that provide reliable contact.

FAST DIGITAL
SIGNAL PROCESSOR

A miniature digital signal processor (DSP) digitizes audio from a sound capsule with a sampling rate of 44100 Hz and 16-bit sampling.
A distinctive feature of the processor is the presence of 2-speed AGCs, providing lightning-fast automatic gain control, both at the input and at the output of the device.
6 digital filters of the processor process the signal so that only the speech range remains at the line output.
Precise built-in preamplifier ensures high signal-to-noise ratio.

CONTROL PROCESSOR
DIGITAL MICROPHONE

The central control processor of the STELBERRY M-50 digital microphone provides microphone gain control and signal processing parameters.
The processor guarantees a quick exit of the microphone to the working mode after power-up, thanks to the high-speed communication line with the signal processor.

DIGITAL MICROPHONE WIND PROTECTION
STELBERRY M-50

For perfect sound transmission, the digital microphone is equipped with a windscreen filter.
By eliminating the wind component, the filter made of acoustic material cuts out unwanted sounds arising from the collision of wind currents with the sensitive membrane, resulting in a crystal sound.
The wind shield allowed us to create an effective microphone for voice.

OPTIMIZATION OF 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 270 ... 4000 Hz.
Thanks to this bandwidth, excellent speech intelligibility is achieved regardless of extraneous noise sources.
The signal processing is carried out by six digital high-speed filters, which guarantees a high slope of the frequency response in the low and high frequencies.

DOUBLE AGC SYSTEM

The microphone is equipped with two digital high-speed Automatic Gain Controls (AGC).
The first AGC adjusts the gain at the microphone input, immediately after digitizing the signal from the capsule, and the speed of response to changes in sound level is less than 1/1000 of a second.
This allows you to react to any, even the smallest, changes in the sound environment of the environment.
The second AGC processes the signal at the microphone output, reliably maintaining a stable output level. The response speed of the output AGC system is also less than 1 / 1000th of a second.

COMPARISON OF DIGITAL AUTOMATIC GAIN CONTROL (AGC) with ANALOG AGC

Stelberry M-50 digital microphone with adjustable gain, built on a dedicated processor. Microphone operation is an analog-to-digital conversion of the microphone capsule signal, subsequent digital filtering of the received signal, and reverse digital-to-analog conversion. The digital filters of the sensitive microphone M-50 are tuned to the range of human speech. Sound frequencies outside the 270 ... 4000 Hz frequency range are significantly attenuated by the microphone. The very fast AGC (Automatic Gain Control) of the digital microphone allows it to be used comfortably 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 high-sensitivity microphone for video cameras and audio recorders that are sensitive to the input signal level, and do not have own funds filtering sound.

Sensitive microphone Stelberry M-50 is used as an external microphone for various CCTV 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.

Location of the digital microphone with AGC Stelberry M-50 in the room

When the M-50 microphone is placed in the corner of the room and the maximum microphone sensitivity is set, the comfortable listening area will correspond to a quarter circle area of \u200b\u200b50m². With further distance from the microphone, the level of its output signal will gradually weaken up to the acoustic audibility limit 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-in of the camcorder. Connecting a microphone to the camera is done in this way. The yellow wire of the M-50 microphone, to the camera input connector "Jack-3.5mm", is connected to the terminal (central) and ring contact of the connector (Specify in the manual for the camera.). If the camera or IP camera uses the RCA connector for audio input, then to the center pin of the RCA connector. The black wire of the M-50 digital microphone is connected to the common (body) contact of the Jack-3.5mm 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 diagram of a digital microphone with AGC and gain control Stelberry M-50

The digital microphone of the speech frequency range Stelberry M-50 is omnidirectional, and has a circular directivity pattern with a slight attenuation of the microphone sensitivity from the side of the sensitivity control. The radiation pattern is shown for the microphone capsule used in the microphone, taking into account the influence of the microphone body.

Microphones Stelberry





Description STELBERRY M-50

STELBERRY M-50 is a completely new solution for sound 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 control adjusts the sensitivity of the digital microphone to suit any application.

IP microphone

The digital microphone is ideal for connecting to the line input of IP cameras, perfectly conveying the acoustic image of the environment. This application actually makes it a complete IP microphone. Also, the undoubted advantage of this solution is the ability to install a digital microphone anywhere, regardless of where the IP camera is installed.

High-speed digital signal processor

A miniature digital signal processor (DSP) digitizes audio from a sound capsule at 44100 Hz sampling rate 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 at the output of the device. 6 digital filters of the processor process the signal so that only the speech range remains at the line output. Precise built-in preamplifier ensures high signal-to-noise ratio.

Digital Microphone Control Processor

The digital microphone's central control processor provides microphone gain control and signal processing parameters. The processor guarantees a quick exit of the microphone to the operating mode after power supply, thanks to the high-speed communication line with the signal processor.

Wind protection for digital microphone

For perfect sound transmission, the digital microphone is equipped with a windscreen filter. By eliminating the wind component, the filter made of acoustic material cuts out unwanted sounds arising from the collision of wind currents with the sensitive membrane, resulting in a crystal sound. The wind shield allowed us to create an effective microphone for voice.

Optimizing the microphone for the speech range

The bandwidth of the digital microphone is tuned to the frequency range of human speech and lies in the range of 270 ... 4000 Hz. Thanks to this bandwidth, excellent speech intelligibility is achieved regardless of extraneous noise sources. The signal processing is carried out by six digital high-speed filters, which guarantees a high slope of the frequency response in the low and high frequencies.

Dual AGC system

The microphone is equipped with two digital high-speed Automatic Gain Controls (AGC). The first AGC adjusts the gain at the microphone input, immediately after digitizing the signal from the capsule, and the speed of response to changes in sound level is less than 1/1000 of a second. This allows you to react to any, even the smallest, changes in the sound environment of the environment. The second AGC processes the signal at the microphone output, reliably maintaining a stable output level. The response speed of the output AGC system is also less than 1 / 1000th of a second.

Convenient adjustment

The convenient location of the gain control makes it easy to adjust the microphone gain. The peculiarity of a high-sensitivity microphone is that the gain adjustment occurs before the start of the AGC processing. This makes it easy to achieve the desired sound quality. The microphone's bandwidth is designed to pass voice frequencies, eliminating unwanted sounds from high frequency sources.

Specifications STELBERRY M-50

  • Unit of measurement: 1 piece
  • Dimensions (mm): 10x10x52
  • Weight (kg): 0.01
  • Acoustic range: up to 20 meters
  • Protection against electromagnetic interference: yes
  • Wind protection: Acoustic foam rubber
  • Bandwidth (after digital processing): 270 ... 4000 Hz
  • Line length: up to 300 meters
  • Gain adjustment range: 350 times
  • Number of digital AGCs: 2
  • "Angle of attack" of the input AGC: 0.7 ms
  • Output AGC "angle of attack": 0.7 msec
  • Digital Low Pass Filtering: 2 1st order filters
  • Digital High Pass Filtering: 3 2nd order filters
  • Signal to noise ratio: 38dB
  • Sampling: 16 bit
  • Sampling frequency: 44100 Hz
  • Body: aluminum
  • Power supply: 7.5 ... 16 Volts
  • Consumption: 20mA
  • Dimensions: Ø10x52 mm
  • Weight: 10 grams

 

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