PCM | Arduino library for simple playback

I have data on subject codes and grades for students. Each has to take 4 subjects out of which English is mandatory. It is represented by code 301.

Columns: SUB1, SUB2, and so on represent subject codes for other modules and the next column represents the marks.

Based on these codes I am trying to do two things:

First thing:

I am trying to create a course column consisting of PCB if the student has subject codes 42, 43, and 44. PCM if the student has subject codes 41, 42, and 43. Commerce if the codes are 55, 54, and 30.

The issue that I am facing is that the codes are spread across columns and I facing difficulty to standardize them.

Second thing:

Based on the identified course, I am trying to sum the grades obtained by each student in these subjects. However, I am want to add the English grade to it as well.

Example of the data:

I am trying to build a graphical audio spectrum analyzer on Linux. I run an FFT function on each buffer of PCM samples/frames fed to the audio hardware so I can see which frequencies are the most prevalent in the audio output. Everything works, except the results from the FFT function only allocate a few array elements (bins) to the lower and mid frequencies. I understand that audio is logarithmic, and the FFT works with linear data. But with so little allocation to low/mid frequencies, I'm not sure how I can separate things cleanly to show the frequency distribution graphically. I have tried with window sizes of 256 up to 1024 bytes, and while the larger windows give more resolution in the low/mid range, it's still not that much. I am also applying a Hann function to each chunk of data to smooth out the window boundaries.

For example, I test using a mono audio file that plays tones at 120, 440, 1000, 5000, 15000 and 20000 Hz. These should be somewhat evenly distributed throughout the spectrum when interpreting them logarithmically. However, since FFTW works linearly, with a 256 element or 1024 element array only about 10% of the return array actually holds values up to about 5 kHz. The remainder of the array from FFTW contains frequencies above 10-15 kHz.

Here's roughly the result I'm after:

But this is what I'm actually getting:

Again, I understand this is probably working as designed, but I still need a way to get more resolution in the bottom and mids so I can separate the frequencies better.

What can I do to make this work?

I am trying to record in a Beaglebone AI from a Circular Microphone Board TIDA-01454. I have checked that the CMB's output is correct with an I2S DAC(I can hear the mics sound). However, I cant record in the Beaglebone AI, this is what happens when I try to arecord:

arecord --device="hw:1,0" -c2 -f S32_LE test.wav

Recording WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 8000 Hz, Stereo

arecord: pcm_read:2145: read error: Input/output error

I think the problem must be somewhere in my DTS file or ALSA configuration. But I dont really know, so if more information is needed, just ask for it.

DTS(not complete):

I am trying to make a speech to text system using raspberry pi. There are many problems with VAD. I am using DeepCpeech's VAD script. Adafruit I2S MEMS microphone accepts only 32-bit PCM audio. So I modified the script to record 32-bit audio and then convert it to 16 bit for DeepSpeech's processing. Frames generation and conversation parts are below:

I am trying to plot two imshow and one plot above each other sharing their x-axis. The figure layout is set up using gridspec. Here is a MWE:

I'm working on a script that sends data from a microphone to Google Cloud Speech-to-Text API. I need to access gRPC API to produce live readings during recording. Once the recording is completed, I need to access REST API for more precise asynchronous recognition.

The live streaming part is working. It is based on the quickstart sample, but with python-sounddevice instead of pyAudio. The stream below records cffi_backend_buffer objects into a queue, a separate thread collects these objects, converts them to bytes, and feeds them to the API.

I am trying to record an audio using AudioRecord on android and seperate the right and left channel recordings into two different files and then convert it to wav to be able to play on the phone.But the recorded files have fast speed and it has high pitch.

I read all the samples and wrote this code but I am not sure which part is causing the problem.

This is my AudioRecord definition.

I am new to python so please bear with me with silly questions I have multiple xml in the following format and I would like to extract certain tags within those xmls and export them to a single csv file.

Here is an example of the xml (c:\xml\1.xml)

We have an iOS project that, when built with XCode 12.5, will error with something similar the following:

Module file '/Users/revolt/Library/Developer/Xcode/DerivedData/ModuleCache.noindex/J3JDK8UPXZ4K/Foundation-39SXNSRMYWPGZ.pcm' is out of date and needs to be rebuilt: signature mismatch

This does not happen on the first build, but after the cache has been generated and can be 'solved' by deleting the DerivedData folder.