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SikCiv

Catching DirectSound Software mixing Output?

2 posts in this topic

Audio mixing works like image blending, I beleive. You can add two signals together by halving them both first. The halving is needed to keep the data from over flowing or distorting, but, admitedly, you lose volume because the peak differentials are halved.

a/2 + b/2 = c

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Anyone know how to catch (save to memory) the output of the primary sound buffer while directsound is mixing samples from the secondary buffers?

I guess its only possible in software mixing, as hardware mixing is done on the sound card.

Also, is it possible to dump the contents of the sound card's output into your own buffer?

What i'm trying to do is mix a few seperate samples into one wave file. But my problem is I cant find any code to mix samples using software mixing, so I am trying to find a way to use Directsound's mixing feature to do it.

I know there are audio applications to do this, but I need to implement this feature into my own application.


(-: !!!!help!!!! :-(

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yeh, but isn't mixing more complicated?

Ive seen some assembly source code to mix two 8bit mono samples, and it looked VERY compicated (yes I do understand ASM). I dont have the time or resources to make my own mixing routine.
My objective is to mix 'many' samples into onoe sample in CD quality! Why re-invent the wheel, when its already available.

Anyone have a cheap library or source code for CD quality mixing??
It doesnt have to be lightning fast, I just need it to mix good quality sound.


pls help.

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