问题
I want to increase a volume on buffer with voice data. The point is I'm using DirectSound and I have one primary and one secondary buffer - all streams mixing is done by hand. In a voice chat all participants can have independent volume levels. I multiply each stream data by a value (gain) and sum it to one buffer. Everything works fine but when I try to multiply data by a value greater than 1.0f - I hear some clipping or what.
I've tried using Audacity effect compressor but this doesn't help reducing the strange noise.
Probably I should modify gain in some other way? Or just use another post-processing algorithm?
UPDATE: Wow, I've just found out interesting thing! I've dumped audio before increasing volume and right after that.
Here is the pic
Sorry for the quality - I think that's how the sound is supposed to appear (I've drawn red line myself). Really looks like values exceed the sample data type. But I cannot understand WHY? My samplebuffer is BYTE but I access it only via short pointer. It is signed but clipping happens even when *ptr is about 15-20 thousand.
回答1:
For every sample - convert it to some larger data type - if you have 16 bit signed samples, they originally fit in SHORT - extract it from the stream, then cast to local double, then multiply, then CLIP, then cast back to SHORT.
It MUST work that way...
I can even provide code sample if needed.
EDIT:
Your picture is exact evidence that you didn't expand to larger type before multiplication - you can't 'capture' clipping condition on SHORT because it will wrap automatically.
short* sampleBuffer;
...
short sample=*sampleBuffer;
double dsample=(double)sample * gain;
if (dsample>32767.0) {dsample=32767.0;}
if (dsample<-32768.0) {dsample=-32768.0;}
*sampleBuffer=(short)dsample;
sampleBuffer++;
And one more EDIT:
if you have several voices - first put them all to double - then GAIN each one - then add them - and CLIP them as the last step.
One more EDIT (+1s are inspiring me):
If you have STEREO, same stuff will work also, just count all the samples x2 i.e.
number of shorts = number of samples * 2.
来源：https://stackoverflow.com/questions/4300995/modify-volume-gain-on-audio-sample-buffer