Pitch dropping on files ??

This has happened a few times now recently...  I record some audio for a project in Sonar, and it sounds fine, but then if I open it in Adobe Audition 3, at some point when I come back to it in Sonar (and in AA) the pitch has dropped.  The timing is still the same, but the pitch of the audio is now noticeably lower.  And I can't seem to figure out what's causing that or how to fix it or change it back to the right pitch.  Is it some kind of conflict between the two programs?  Somehow the sample rate is changing, I presume.  Best I can tell, everything is configured correctly (I'm using an Echo Mia card with a Win 7 computer).  Both programs say that they are sampling at 44100.

Okay, it doesn't have anything to do with AA.  I have a project open in Sonar... sounds fine... then just listened to some sound sample effects from a website, and then went back to Sonar and the pitch is low.  If I exit out of Sonar and go back in, it's okay again.  So, anyway...

So, anyway...

I was about to suggest that another app had altered your soundcard sample rate, when you sort-of worked it out for yourself! Moral, therefore solution is obvious, I hope.

No, still trying to figure it out, but it's clearly more of a Sonar settings issue than AA.

Well, let's put it like this: if you have one program running your sound device at 44.1k and, let's say, another one running at 48k and you switch between the two, then the sound device is likely, when you switch back and try to play the file from the first one without reloading it, to play it at the last rate the device was set at - which means that it could either be about a semitone lower, or higher, depending on which way you go.

It can potentially get worse with ASIO, I think - although I'd have to try this to prove the hypothesis. If you have different channels attached to different programs (which is allowable), then I don't think that it's beyond the bounds of possibility that if you start a new file in one whilst the other's playing, that it will alter the rate during playback of the first one! Now, any sensible driver ought to trap that out, but you never know...

As far as I can tell from the tests I've been doing recently with Windows 7, the whole subject is a can of worms.

In general AA behaves impeccably on its own, switching between sample rates depending on the sample rate of files and handling sessions correctly.

Where it becomes impossible to predict is when different manufacturer's drivers behave differently in whether they reflect settings  through from different applications and whether asio settings can affect the wdm settings and vice versa. Asio may bypass the Windows audio, but it doesn't necessarily bypass the Windows settings.

The basic problem is that Windows' audio handling has become so potentially complex that it is becoming very difficult to advise people on how to set up general purpose but occasionally pro-audio machines.

The rule seems to be - switch off all Windows Sounds, decide on a sample rate that you will always use and set all defaults to that sample rate. Keep all internet audio etc on the on-board audio interface if possible and all the serious stuff on the proper soundcard.

Just read the whole thread.

Now it make sense to me. Because I'd been having trouble with my inter phase. Having read this I tried to do and try it now I am able to figure it out!

Thanks!

We've had similar problems, but we think i t may be more related to the echo card. There is a setting where you can "lock" the sample rate for the card, so we're going to try that next.

This business of different sources hi-jacking the sound card's clocking rate is nothing new - it's been going on for as long as I can remember - so I think it's unfair to lay the blame at Win 7's doorstep.  In fact, although it pains me to say it, it's not really a fault of the OS itself at all, but the way different applications are written.

This business of different sources hi-jacking the sound card's clocking rate is nothing new - it's been going on for as long as I can remember - so I think it's unfair to lay the blame at Win 7's doorstep.  In fact, although it pains me to say it, it's not really a fault of the OS itself at all, but the way different applications are written.

Graeme, I am not concerned so much with, say, an 'email arrived' ping from the OS affecting the sample rate because in Windows 7 with on-board sound or a professional card selected as the default audio device, that is covered normally by the built-in Win 7 sample rate converter.

Where it gets complicated and different with the different usb audio interfaces I've been testing is when the usb audio interface is set up in the Windows 7 control panel and mixer to have a default sample rate, and then the asio part of the driver is used for some proper recording or playback. With many audio interfaces I believe this is the cause of the "asio sample rate not supported" errors that can be seen so frequently on the message boards. When using two applications, it is quite possible that one is picking up settings from asio drivers and the other using the Windows audio settings. Some audio devices trigger the error message, others don't. Some drivers seem to cross-pollinate settings between sections of the driver, others don't. In all this some interfaces advertise multi-client drivers without specifying what that actually means

My problem with Windows 7 and audio is simply that there are so many options and places where an error can be made that a "normal" person will have great difficulty in being really confident that everything is set correctly. Of course, Windows 7 having a sample rate converter that can cause serious aliasing doesn't help at all.

I haven't really collected all my notes on this to produce a sensible definitive statement because that would take a major effort and I get exhausted just thinking about it.

My problem with Windows 7 and audio is simply that there are so many options and places where an error can be made that a "normal" person will have great difficulty in being really confident that everything is set correctly.

While I take your point, 'normal' people couldn't give a stuff about clocking rates. As long as they hear a sound, they will be happy.

Which, I suppose, is another way of saying none of us here is normal .