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JCBigler said:
No. Recording at 48k and 44.1 won't make a difference in pitch. If it's recorded at 44.1k and you try to play back at 48k, you will not get a pitch change. What will happen is 1) your play back system won't play it at all because the sample rates are different, or 2) it will automatically convert the sample rate to the current sample rate and play back as it should.
yes, that is what is SUPPOSED to happen. not always the case though. for example, if i play an mp3 file and while it's playing, i open a soft synth and change the settings to a different sample rate, now the mp3 file is playing back at the wrong speed. it's not supposed to happen, but i've seen things played back at the wrong sample rate quite a few times. It can also happen if a program changes the sample rate, but fails to change it back on exit (or crash.)

sure, it might not be the most likely thing, but no more unheard of than accidently turning on a pitch shifting plugin.
 

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JPSaxMan- just for clarification, what is the first note supposed to be? and when you say it happens in both audition and the sound recorder, do you just mean that it plays back the file sharp in both, or that you actually tried to record in both?
 

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i stand by my original claim that it's a sample rate problem. The only other explanation that's even possible is that you have some sort of EAX type onboard soundcard effect enabled, which seems unlikely since the file plays back sharp for everyone. My guess is that if you uninstall and then reinstall your soundcard drivers, the problem will go away.

there are a very limited number of reasons why a digital file would record or playback at the wrong pitch, and i think they've all been covered.
 

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two things to try.... try recording something that's already on the computer: go into the recording properties of the soundcard volume control and change it so that it's recording "what you hear" instead of the mic. then play back something that sounds correct in one app like an audio cd, and record it in sound recorder.

also, try recording a metronome, and then see if the tempos match up when you play it back so that you know if it's just messing up the pitch, or the time as well. my guess is both.

i think we can eliminate some of the possibilities:

  • It can't be faulty compression if it happens in sound recorder, because it uses no compression.
  • it can't be a plugin because no such plugin exists for sound recorder.
  • i don't think the onboard audio on a dell has any kind of real time effects that can be enabled, so that's also unlikely.

it shouldn't be hard to download the audio drivers from dell and reinstall them. i bet it fixes your problem.
 

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JCBigler said:
This is NOT analog tape we're talking about here. The sample rate that is recorded has nothing to do with the play back pitch. Pitch shifting capabilities is a fairly advanced and complex process which requires lots of CPU horse power.
pitch shifting (changing pitch without changing tempo) is indeed a fairly complex process. but playing something back at the incorrect sample rate (changing both pitch and tempo) can be an extremly simple process and one that CAN get screwed up from time to time. and playing back something at the wrong sample rate without compensating for the time and pitch ends up with results EXACTLY like playing back analog tape does. obviously, this isn't supposed to happen. the differences between sample rates are supposed to be compensated for, but sometimes things get messed up.

you can keep telling me it never happens, but you are simply wrong. it does. i've seen it. If, for example, the files he is recording are being tagged with the incorrect sample rate (because of a problem with the sound card drivers most likely) they would then play back at the wrong pitch and tempo on any computer.

these types of problems are relatively rare, but a hell of a lot more plausible than accidently turning on a pitch shifting plugin in one app, and then accidently using a post process time stretch effect in another app (and by some huge coincidence having accidently set them to produce similar pitch differences.)

Ironically, the last scenario you describe, with the clock speed being set wrong, IS exactly the type of problem you keep telling me doesn't exist - a sample rate problem. it is, however, unlikely as it would not cause a speed or pitch issue on his computer (but it would if he tried to play it back on another computer.)
 

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Toobz said:
Sample rate issues would not create speed problems on playback.

Digital code is 16 bit 000's and 111's. (0011010110010111) it either IS
read properly or not at all. If speed was a problem during playback,
IT HAS to be on the analog side. This is not up for argument. There is
no such thing as FAST or FASTER 000's & 111"s (0011010110010111).
Once the digital code is converted to analog, that is where the speed
problem can manifest itself, and ONLY then . If digital settings are affecting
playback, it has to be some code that is tied into the analog control stages somewhere.
This is still an analog problem, and not a decoding one.
as others have pointed out... this is total BS. the sample rate is simply the number of samples per second. there absolutely IS such a thing as faster 0s and 1s. if a one minute sound clip at 96k/16-bit has a bigger file size than a one minute sound clip at 44.1k/16-bit, then clearly, since they both take a minute to play back, and the samples are the same size, then the 0s and 1s MUST be read faster in the 96k version of the file.

I deal with digital audio and video all day for a living, and i've seen these kinds of problems before. They can certainly happen on the digital side. In fact, it would be near impossible for this type of problem to occur after the digital to analog conversion seeing as it would require a TIME MACHINE in order to play the file faster than it was being coverted.

The only difference between files of different sample rates, is the header information. I could go in and manually edit the header of a 44.1k wav file and change it to 96k. the file would then playback much faster and higher in pitch. thus, if the header information was written incorrectly, the file could play back at the wrong speed.

you can say this isn't up for argument all you want, but you clearly have no idea what you're talking about. I'm sorry if I sound rude, but it's a major pet peeve of mine when uninformed people spread disinformation and negate the advice of those who actually do know what they are talking about and are trying to help.
 

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Giganova said:
Toobz is still right that sample rate issues would not create speed problems on playback. You all have to be more careful when you talk about these things: Statements like "it was recorded at 22.05k, and played back at 44.1k" don't make sense because a sample rate is in kHz, which is the unit for "samples PER SECOND".
Actually, it's the unit for "thousands of samples per second," but yes close enough.

Giganova said:
Therefore, a sample rate alone doesn't change pitch because a sample rate of is always referring to an amount of time (in units of seconds).
This is incredibly backwards logic. First of all time = pitch when talking about samples. faster samples = higher pitch. that's why pitch is also refered to as "frequency", as in how frequent the waveforms repeat. faster samples = faster waveform cycles. I suppose you could say that the sample rate refers to an amount of time... for example a sample in 44.1khz could be said to take up 1/44100 of a second... of course there's nothing in the actual sample data itself that has anything to do with frequency or sample rate. It is simply a string of numbers. The sample rate data is all in the header of the file, and changing it, will change how fast the computer plays back the samples. faster samples = faster tempo & higher pitch.

I think that some of the confusion here might come from the fact that your computer will atomatically convert sample rates in most instances. If my soundcard is set to 48khz and i try to play back a 44.1 khz file, it still plays back correctly, but that is only because the computer is compensating for the difference and converting the sample rate. This process is hidden from you, but it happens. If the computer didn't know that it was supposed to be a 44.1khz file and played it back just like any other 48k file, the pitch would be higher and the tempo would be faster.

FROM WIKIPEDIA: (the bold is mine)

SAMPLE RATE CONVERSION
Sample rate conversion is the process of converting a (usually digital) signal from one sampling rate to another, while changing the information carried by the signal as little as possible. When applied to an image, this process is often called scaling.

Sample rate conversion is needed because different systems use different sampling rates, for engineering, economic, or historical reasons. The physics of sampling merely sets minimum sampling rate (an analog signal can be sampled at any rate above twice the highest frequency contained in the signal, see Nyquist frequency), and so other factors determine the actual rates used. For example, different audio systems use different rates of 44.1, 48, and 96 kHz. As another example, American television, European television, and movies all use different numbers of frames per second. Users would like to transfer source material between these systems. Just replaying the existing data at the new rate will not normally work - it introduces large changes in pitch (for audio) and movement as well (for video), plus it cannot be done in real time. Hence sample rate conversion is required.
I really hope that settles this nonsense.
 
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