My planned application is Linux/Windows cross-platform, so I can’t
use any DirectSound-dependent functions.
I really need, at a minimum, a 3-band equalizer (low/midrange/high)
but writing iir DSP filters is something I really don’t want to do, if I
don’t absolutely have to. As this appears to be something a lot of people
have asked about, and a few appear to have achieved, I was wondering
if anyone has some code they could share on how this is done – I can’t
tell you how much I’d appreciate it!
I have found some sample code for doing this, but it’s implemented
for the fixed-point MAD mp3 library, which I’d consider using but would
lose support for Ogg Vorbis. Also, I’m kind of scratching my head
about whether different filter equations have to be done for each
output rate (48khz, 44khz, etc.) as well as different implementations
for the different stream data types (float, integer) I might encounter.
Don’t really care what language it’s written in – I write in c/c++ but
translation is a much smaller concern than writing this from scratch.
Feel free to email me at firstname.lastname@example.org
- Baudtender asked 15 years ago
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