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Typical USB sound cards have an earphone jack and a microphone jack. Since it works with your microphone, it is set up to deliver power to the electret microphone - hence the 4.8 Volts when open. That is to say, the sound card is delivering DC Voltage to your microphone. The microphone input of a sound card can't supply much current, but it is enough for ...

3

After doing some research I've found that: MEMS microphones have typical diaphragms of 0.5 mm (I've checked Knowles which seems to be one of the world leaders in the area): ) Assuming that we are working with a MEMS microphone with a diameter of 0.5 mm and a frequency of 18 kHz, from the frequency and the speed of sound we can determine λ = speed/...

3

You need to provide 2 clocks - one is a division of the other. The main clock is the "bit clock", aka the Serial Clock. This is one clock per data bit transferred, and 32 bits of data per sample. The secondary clock, normally known as LRCLK is the left / right channel clock. This divides the data stream into pairs of samples, one for the left channel, ...

2

That "microphone" is actually a complete assembly that includes a amplifier. As the spec says, it puts out about ±200 mV when you talk normally into it from arms length. You want to connect this to the "line in" jack of your computer, not the microphone jack. The line in jack is meant for signals like this that are already at a roughly standard &...

2

The jacks are standard 3.5mm mini-phone plugs, otherwise known as tip-ring-sleeve jacks. They're not powered; as you observe the sleeve is ground, but the tip and ring are the left and right stereo channels. What you describe ought to work, assuming the breakout board you're using provides enough amplification. The critical question is what input levels the ...

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One option would be to buy a hot-air reflow station, e.g. Sparkfun. Chinese-manufactured ones are reasonably inexpensive, and some are of good enough quality for prototype/home use. You might also consider dead-bug prototyping with some 30-gauge wire or so and a good quality hand solder with a fine tip. This involves turning chips upside down and soldering ...

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The main point here is that the microphone will pick up all sounds at any distance. The problem, as has been already discussed, is that ambient noise plus electronic noise will limit your ability to distinguish them. Our ears (ears is important since we have two which helps the process), coupled with our brains, have significant signal processing capability ...

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A microphone can pick up sounds of any level. However, it also generates noise, as does the amplifier following it. Very quiet sounds will be indistinguishable able the noise. You therefore need to concentrate on the Signal to Noise ratio (SNR) of your microphone + amplifier system. I emphasise system, as there are many sources of noise, and you need to ...

2

I believe your chart choices are correct: the OPA344 was chosen for its low quiescent current, making it a good choice for battery-powered applications. The trade-off comes from your other desired areas of improvement. The OPA344 has a gain-bandwidth product of only 1 MHz, which makes it a poor choice for high-frequency amplification in your circuit. In ...

2

What are the advantages of using a stereo microphone on the quality of sound? The advantage of a stereo microphone (or two mono microphones) is that it is possible to record in stereo for reproduction of a stereophonic image with sound sources located in the audio panorama. There is no advantage in fidelity of signal or signal to noise ratio. If ...

2

Well that calculation works fine for sine wave audio being measured by the microphone but real audio has a significant crest factor and this means that the peak voltages will be much greater for the same acoustic power signal. If you want to avoid distortion/clipping consider crest-factor: - Picture from here. You should also consider that your ADC will ...

1

We nomrally can hear a 100 decible or dB range at 1 kilohertz (kHz) from perfect silence to a loud stereo at 1m. ( more can be painful) The typical (1933) Fletcher Munson curves are below for human hearing. This logarithmic pressure and frequency range in Sound Pressure Level (SPL) is plotted in dB SPL so 100dB spans 5 decades or 20dB per decade. How ...

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The sensitivity is -38 dBV for a 1 kHz sound pressure level of 94 dB (the data sheet tells you that) and -38 dBV is 12.6 mV RMS so you now know the signal level in volts, the frequency (1 kHz) and the sound pressure level (1 pascal = 94 dB SPL). If the sound pressure level decreased by 10 dB the voltage output would fall by 3.162 to 4 mV because 10 dB is a ...

1

The mems microphone produces a digital signal and when the incident RMS pressure is 94 dB SPL, the digital signal has an RMS level of -26 dBFS. If the positive full scale is (say) $2^{15}$ or 32,768 then the digital RMS level is 26 dB down on this at 1,642. This will be at 1 kHz. 1,642 is an RMS number so the full peak-to peak value (for a sinewave) is ...

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Excellentish PCM2912 Application Note here From the SMA100 datasheet note that Vdd absolute maximum is 4.0V !!!! Normal operating Vdd range is 1.5 to 3.6V IF the PCXM2912 is operated fom 3.6V or less then you can share Vdd/Vcc between the two ICs. IF PCM2912 Vcc is 5V you MUST NOT share Vdds. AT 5V Vdd on the CODEC the Mbias voltage is 5 x 75% = ...

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The MEMs mics used in the tutorial don't output data, they put out a plain old analog audio signal that is digitized by the USB Soundcard. The ones you are looking at (the ones whose datasheet you linked to) are designed for digital interfaces and CANNOT be used in place of the analog ones.

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It seems that microphones were damaged during PCB cleaning process. The producer puts spray on the PCB after production and small particules may enter inside the sound inlet during the process. When we changed the microphones, they started working. This explains the problem.

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One thing that is really key is you want to be sure you sample them together, but you keep all of their clocks rock steady without interruption. If you had four i2s channels carefully configured that would work. I wonder about an fpga design that would divide down a 4x clock from a micro, collect data from the microphones at that rate and pass it on at ...

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It appears to be a device with digital output using PDM (Pulse Density Modulation), which is the 1-bit oversampled output of a sigma-delta ADC-- it's similar to what's used in SACD. A microcontroller should be able to convert it to the more usual PCM format by digital filtering and resampling, provided it has enough processing power. Since the CS43L22 ...

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