ESP8266 analogRead() microphone Input into playable audio

My goal is to record audio using an electret microphone hooked into the analog pin of an esp8266 (12E) and then be able to play this audio on another device. My circuit is:

In order to check the output of the microphone I connected the circuit to the oscilloscope and got this:

In the "gif" above you can see the waves made by my voice when talking to microphone.
here is my code on esp8266:

void loop() {
  sensorValue = analogRead(sensorPin);
  Serial.print(sensorValue);
  Serial.print(" ");
}

I would like to play the audio on the "Audacity" software in order to have an understanding of the result. Therefore, I copied the numbers from the serial monitor and paste it into the python code that maps the data to (-1,1) interval:

def mapPoint(value, currentMin, currentMax, targetMin, targetMax):
    currentInterval = currentMax - currentMin
    targetInterval = targetMax - targetMin

    valueScaled = float(value - currentMin) / float(currentInterval)

    return round(targetMin + (valueScaled * targetInterval),5)

class mapper():
    def __init__(self,raws):
        self.raws=raws.split(" ")
        self.raws=[float(i) for i in self.raws]

    def mapAll(self):
        self.mappeds=[mapPoint(i,min(self.raws),max(self.raws),-1,1) for i in self.raws ]
        self.strmappeds=str(self.mappeds).replace(",","").replace("]","").replace("[","")
        return self.strmappeds

Which takes the string of numbers, map them on the target interval (-1 ,+1) and return a space (" ") separated string of data ready to import into Audacity software. (Tools>Sample Data Import and then select the text file including the data). The result of importing data from almost 5 seconds voice:

which is about half a second and when I play I hear unintelligible noise. I also tried lower frequencies but there was only noise there, too.

The suspected causes for the problem are:

1- Esp8266 has not the capability to read the analog pin fast enough to return meaningful data (which is probably not the case since it's clock speed is around 100MHz).

2- The way software is gathering the data and outputs it is not the most optimized way (In the loop, Serial.print, etc.)

3- The microphone circuit output is too noisy. (which might be, but as observed from the oscilloscope test, my voice has to make a difference in the output audio. Which was not audible from the audacity)

4- The way I mapped and prepared the data for the Audacity.

• Is there something else I could try?

• Are there similar projects out there? (which to my surprise I couldn't find anything which was done transparently!)

• What can be the right way to do this? (since it can be a very useful and economic method for recording, transmitting and analyzing audio.)

There are many issues with your project:

You do not set a bias voltage on A0. The ADC can only measure voltages between Ground and VCC. When removing the microphone from the circuit, the voltage at A0 should be close to VCC/2. This is usually achieved by adding a voltage divider between VCC and GND made of 2 resistors, and connected directly to A0. Between the cap and A0.

Also, your circuit looks weird... Is the 47uF cap connected directly to the 3.3V ? If that's the case, you should connect it to pin 2 of the microphone instead. This would also indicate that right now your ADC is only recording noise (no bias voltage will do that).

You do not pace you input, meaning that you do not have a constant sampling rate. That is a very important issue. I suggest you set yourself a realistic target that is well within the limits of the ADC, and the limits of your serial port. The transfer rate in bytes/sec of a serial port is usually equal to baud-rate / 8. For 9600 bauds, that's only about 1200 bytes/sec, which means that once converted to text, you max transfer rate drops to about 400 samples per second. This issue needs to be addressed and the max calculated before you begin, as the max attainable overall sample rate is the maximum of the sample rate from the ADC and the transfer rate of the serial port.

The way to grab samples depends a lot on your needs and what you are trying to do with this project, your audio bandwidth, resolution and audio quality requirements for the application and the amount of work you can put into it. Reading from a loop as you are doing now may work with a fast enough serial port, but the quality will always be poor.

The way that is usually done is with a timer interrupt starting the ADC measurement and an ADC interrupt grabbing the result and storing it in a small FIFO, while the main loop transfers from this ADC fifo to the serial port, along the other tasks assigned to the chip. This cannot be done directly with the Arduino libraries, as you need to control the ADC directly to do that.

Here a short checklist of things to do:

1. Get the full ESP8266 datasheet from Expressif. Look up the actual specs of the ADC, mainly: the sample rates and resolutions available with your oscillator, and also its electrical constraints, at least its input voltage range and input impedance.
2. Once you know these numbers, set yourself some target, the math needed for successful project need input numbers. What is your application? Do you want to record audio or just detect a nondescript noise? What are the minimum requirements needed for things to work?
3. Look up in the Arduino documentartion how to set up a timer interrupt and an ADC interrupt.
4. Look up in the datasheet which registers you'll need to access to configure and run the ADC.
5. Fix the voltage bias issue on the ADC input. Nothing can work before that's done, and you do not want to destroy your processor.
6. Make sure the input AC voltage (the 'swing' voltage) is large enough to give you the results you want. It is not unusual to have to amplify a mic signal (with an opamp or a transistor), just for impedance matching.
7. Then you can start writing code.

This may sound awfully complex for such a small task, but that's what the average day of an embedded programmer looks like.

[EDIT] Your circuit would work a lot better if you simply replaced the 47uF DC blocking capacitor by a series resistor. Its value should be in the 2.2k to 7.6k range, to keep the circuit impedance within the 10k Ohms or so needed for the ADC. This would insure that the input voltage to A0 is within the operating limits of the ADC (GND-3.3V on the NodeMCU board, 0-1V with bare chip).

The signal may still be too weak for your application, though. What is the amplitude of the signal on your scope? How many bits of resolution does that range cover once converted by the ADC? Example, for a .1V peak to peak signal (SIG = 0.1), an ADC range of 0-3.3V (RNG = 3.3) and 10 bits of resolution (RES = 1024), you'll have

binary-range = RES * (SIG / RNG) 
             = 1024 * (0.1 / 3.3)
             = 1024 * .03
             = 31.03

A range of 31, which means around Log2(31) (~= 5) useful bits of resolution, is that enough for your application ?

As an aside note: The ADC will give you positive values, with a DC offset, You will probably need to filter the digital output with a DC blocking filter before playback. https://manual.audacityteam.org/man/dc_offset.html