#1351: aliasing is heared with playback 44.1kHz files on 48kHz system sample rate -----------------------------+---------------------------------------------- Reporter: nutela | Owner: pulkomandy Type: bug | Status: assigned Priority: normal | Milestone: R1 Component: Kits/Media Kit | Version: R1/pre-alpha1 Resolution: | Keywords: Blocked By: | Has a Patch: 0 Platform: x86 | Blocking: -----------------------------+---------------------------------------------- Comment (by stargatefan): Replying to [comment:10 pulkomandy]: > You may have multiple streams at different rates playing at the same time, so the resampler is needed. > It would be nice to have some 'auto' setting for the soundcard output rate, but you may as well need a fixed one (spdif for example). > > So we can improve the situation, but we should fix the mixer anyway. that may very well be true but it would be in some very rare circumstances and generally would be handeled by the host application. typically what is more likely to occur is truncation and decimation to a lower sample rate IE 48khz to 44.1 as 100% of audio gear supports 16b 44.1khz. now where things get tricky in this situation is when you would be mixing audio streams in a Digital workstation program. but those renders should be handled by the host application not the media server as latency correction acros multiple tracks could lead to some very big problems later down the road. What should be allowable is a master bus output from those applications as well as application control over input sources. I would never want the Os to handle audio routing or encoding of the digital media for a rendered wav file in a audio application. If anything I want the Os out of the way as much as possiable.. It just flatly doesn't make sense and no audio software I am aware of does so. -- Ticket URL: <http://dev.haiku-os.org/ticket/1351#comment:12> Haiku <http://dev.haiku-os.org> Haiku - the operating system.