>> In our case, such "solution" would have to affect the whole buffer, meaning >> we would need 3 (or better yet 4) bytes per sample, which would eventually get >> reduced back to 2 bytes on the way out to the sound card. This seems neither >> elegant nor memory efficient but would work, and also solves the "a)" problem >> because we don't need to saturate so an atomic add can be performed on each >> sample. > >Yes, this solution is good. I've though about it, too. Unfortunately, it >adds additional transfers including saturation from the "sum" ring buffer >to the DMA buffer of hardware.
Hmmm... Not exactly. This is not a problem. First of all: it is way better to saturate once (i.e. just before the transfer) since this is a costly operation involving a conditional jump (unless you optimize for mmx) than do it for every channel individually. If you're mixing 4 channels you do it once, not 4 times. Just because you need to store the result in a different buffer, rather than putting it in it's original place seems hardly a big difference (except for cache hits maybe). Also, if you insist on sparing memory (the buffer is not *that* big is it?) you can lay it out as two separate (ring) buffers, one holding upper words, the other holding lower words. Now instead of shifting the samples right n-bits before adding to the buffer, you shift them left 16-n. In effect you will get a buffer (the upper part) which can be directly sent to the audio hw, and which was summed and divided without losing precision. The drawback is you lose the atomic add. If you don't shift, you can still saturate into the "upper" buffer and DMA from there. -------------- Fycio (J.Sobierski) [EMAIL PROTECTED] ------------------------------------------------------- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf _______________________________________________ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
