Steve,Thanks so much. By the time I saw your reply, I already had debugger tripping on division by 0 line, but of course was nowhere close to solving the problem. I've already tried some different compiler settings, and also MS vs. Intel (VS.NET 2008). Will keep you posted.P.S.: Post edited to comply with TOS #8.
Steve,There's one small annoyance I would like to try fixing. Any time playback is started, there's one single "pop" - it sounds like buffer is not empty, and sample(s) that's left there are being played for a fraction of second. If that's so, then buffer needs to be forcibly "flushed" when playback is stopped in Winamp. I guess it could be either input or output buffer.What do you think about that possibility? Where in the code would you suggest to look if that's the case, and how would you suggest to empty the buffer?
Definitely something can be done, since ASIO plugin, for instance, doesn't behave the same way.
Is there a reason why it's done like this? Would it be safe to just remove this "if" statement, so it continues the same as if gapless isn't enabled? If not - what would be best solution?
From a cold boot, the first track played in Winamp doesn't exhibit this problemAdjacent tracks in the playlist of the same format (i.e. sample rate, word length and number of channels) don't exhibit this problem between themThe problem only occurs if you manually stop (not pause) the player while a track is still playing and then manually start a track playing - may also occur if you quit Winamp mid track and restart it or if the format changes - the important point being that audible sound was being output when the plugin was stopped
When i play a normal (16bit) mp3 file
Sorry for interrupting your dialogue, but since i was not able to find any e-mail, blog, forum, etc. on the Kernel Streaming for Winamp's webpage i'm disturbing you here Winamp 5.541 + Kernel Audio 3.63 + SB Audigy 2SE (24bit) + Latest official drivers. When i play a normal (16bit) mp3 file, Kernel streaming always says: Sample size: 24 bits. When i play 16bit FLAC file - it says Sample size: 16 bits. When i play 24 bit 96khz file - it's displayed correctly.Is it normal for FLAC and mp3 files to be streamed differently?I have a small utility attached to Creative's sound driver. I can select PCM sample rate(44.1,48,96) and "Enable Bit-perfect Playback" and some Dolby Digital options.Does Kernel Streaming go around these options or it obeys them too?Thanks!
after closing the box and setting kernel streaming as the output, no devices are shown as being available.
I don't know if you ever solved the filters-error problem, but "no devices are shown" is a problem that I see with ks363 (on 64 bit WinXP). The problem is strictly cosmetic