Master told me an audio synthesis algorithm, thank you

douyafei

I want to write a Filter synthesis sound with two input pins, the problem is the core algorithm: put two Sample
Synthesized algorithm. I hope you have any enlightenment

爱萱儿

The easiest way to add up and divide by 2

douyafei

Put two audio DSample data?

爱萱儿

Yep!
However, if the sampling rate and sampling accuracy are not the same, it may be a bit troublesome!
Do something else!

douyafei

Oh, thanks. I also want to ask how the data is organized in the Sample. I want to figure it out, something went wrong for a reason

douyafei

I haven't figured out how to add memory blocks. Is there an algorithm? Come me one thanks

douyafei

Oh. I know this question.
But the type of data collected is: {This is the data collected by the audio
             FWaveFormat.wFormatTag: = 1;
             FWaveFormat.nChannels: = 2; // mono
             FWaveFormat.nSamplesPerSec: = 44100;
             FWaveFormat.nAvgBytesPerSec: = 176400;
             FWaveFormat.nBlockAlign: = 4;
             FWaveFormat.wbitspersample: = 16;
             FWaveFormat.cbSize: = 0;} The size of each Sample is 176400/2, and each sampling point needs 16 bits, so the number of points I collected is calculated as follows: 8200/2 (channel) / 16 (bit) What is the integer

爱萱儿

16bits = 2 bytes
faint!
 FWaveFormat.nAvgBytesPerSec: = 176400;
See clearly, people are bytes per second

atanihao

a + b-a * b / (2 ^ 16)

This is an algorithm. a, b represent two samples.

There are many algorithms, the key is to ensure that there is no overflow after superposition

douyafei

thank you