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Master told me an audio synthesis algorithm, thank you

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Post time: 2020-1-28 21:40:01
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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
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Post time: 2020-3-2 15:30:01
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The easiest way to add up and divide by 2
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 Author| Post time: 2020-3-4 02:00:01
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Put two audio DSample data?
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Post time: 2020-3-7 21:15:01
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Yep!
However, if the sampling rate and sampling accuracy are not the same, it may be a bit troublesome!
Do something else!
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 Author| Post time: 2020-3-22 22:30:02
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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
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 Author| Post time: 2020-3-26 14:15:01
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I haven't figured out how to add memory blocks. Is there an algorithm? Come me one thanks
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 Author| Post time: 2020-3-31 20:15:02
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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
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Post time: 2020-4-2 09:30:01
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16bits = 2 bytes
faint!
 FWaveFormat.nAvgBytesPerSec: = 176400;
See clearly, people are bytes per second
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Post time: 2020-5-11 22:15:01
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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
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 Author| Post time: 2020-5-15 15:00:02
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thank you
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