byter | 8-bit computer built from 7400 series ICs

I'm trying to write one function that can deinterleave 8/16/24/32 bit audio data, given that the audio data naturally arrives in an 8 bit buffer.

I have this working for 8 bit, and it works for 16/24/32, but only for the first channel (channel 0). I have tried so many + and * and other operators that I'm just guessing at this point. I cannot find the magic formula. I am using C++ but would also accept a memcpy into the vector if that's easiest.

Check out the code. If you change the demux call to another bitrate you will see the problem. There is an easy math solution here I am sure, I just cannot get it.

I am trying to write to wav by taking data from the microphone input, and other headers, and putting that into the wav file. I do that, but it still says corrupted file. One note about the code is that in the struct with the headers, it's not in the correct order. In the WriteToWav function I entered it in the correct order based on the chunk and sub chunks. Here's the code:

I am working on a project where I need to take input from a microphone in c++, and create a wav file that stores the recorded microphone. Right now, I made a simple function that just reads from the microphone, and dumps it into a bin file. I can open it in audacity and hear what I said, but I wish to convert it to sound through my program in lets say a function. Is there anyway I can create a function to convert a binary file into a wav format? It doesn't have to be wav, it can be any popular sound format such as mp3. This is my function for getting the sound --

I'm trying to create a custom BodyPublisher that would deserialize my JSON object. I could just deserialize the JSON when I'm creating the request and use the ofByteArray method of BodyPublishers but I would rather use a custom publisher.

I am using wavesurfer.js to create a multitrack player online and want to export a remixed version of the combined tracks with levels panning etc.

First I have an array of audioFiles and use this to create an array of wavesurfer elements.

I'm trying to create a software that stores the metadata of a wav file.

Here's an MWE:

I am looking for a fast, preferably standard library mechanism to determine the bit-depth of wav file e.g. '16-bit' or '24-bit'.

I am using a subprocess call to Sox to get a plethora of audio metadata but a subprocess call is very slow and the only information I can only currently get reliably from Sox is the bit-depth.

The built in wave module does not have a function like "getbitdepth()" and is also not compatible with 24bit wav files - I could use a 'try except' to access the files metadata using the wave module (if it works, manually record that it is 16bit) then on except call sox instead (where sox will perform the analysis to accurately record its bitdepth). My concern is that that this approach feels like guess work. What if a an 8bit file is read? I would be manually assigning 16-bit when it is not.

SciPy.io.wavefile also is not compatible with 24bit audio so creates a similar issue.

This tutorial is really interesting and even includes some really low level (low level for Python at least) scripting examples to extract information from the wav files headers - unfortunately these scripts don't work for 16-bit audio.

Is there any way to simply (and without calling sox) determine what bit-depth the wav file I'm checking has?

The wave header parser script I'm using is as follows:

Recently I started to code using Scapy. Basing on the example from:

Scapy Sniffing with Custom Actions

I capture multicast UDP datagrams, but besides the amount of captured packets per second, I would like to store the size(in bytes) of packets captured each second(by that I'd multiply the result by 8 and I'd have bitrate). The problem is that capturedPacketsSize seems to be undefined nevertheless I defined it before def custom_action().

I tried to define capturedPacketsSize in different places for e.g. before sniffing in while 1 loop. The same result.

I am testing an application in C# that receives a live audio stream and then saves it to a WAV file. The audio stream has these characteristics: frequency or sampling rate: 16000, channels: 1, frame Samples Per Channel: 320, play Delay in Ms: 200. The audio frames come as floats, and I am collecting the float frames and storing them into a Memorystream with Binarywriter. After that, I convert the content of the Memorystream into an array, and that array then is converted to a Float array again. With the float array, I start the process to assemble the WAV file.

I have compared the float frames values received with the ones inside the float array that I am using to build the WAV file and are the same. I am having trouble processing the float array to assemble the WAV file. I am not sure if I am doing the data conversion wrong with the ConvertAndWrite() method, or if the WAV header is not well formatted according to the characteristics of the audio stream.

I can see the WAV file being created, but there is no content inside apart from the header I think. Any guidance will be much appreciated. I put together this sample code for you to test what I am doing: