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princeganon said:
i stand corrected. still, the chances of this happening in both apps is near zero, unless he has a faulty mp3 codec.
Which is why I draw the conclusion I stated above, given what we know so far.

I guess the original poster doesn't understand those two essential questions. If he wants us to solve this problem he'll have to address them.
 

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How new?

Something I had not thought of until now. This is not to suggest that you should not seek the answers to Al Stevens and princeganon's questions - that can only help you.

My questions are these: how old is the laptop? Is it still under warranty? Have you ever had in-tune recordings on this laptop?

If the laptop is under warranty then our work here may be done (though I don't envy you for the number of hours you may spend on the phone with someone much less helpful than the people on this forum).

I would make the argument that if you did not mess with your bios to begin with and you have default settings in the Windows Media Player/Recorder then it's a problem with either the hardware you have or the way the computer was initially set up. Whether Dell will buy that or not, I'm not sure but you should at the least write down the checks you have made on your system to this point so you can rattle them off to their "support" team.

Just a thought. Good luck!
 

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So, I'm thinking that the OP found a problem, a simple solution and doesn't want to poke his head above the sand for fear of embarrassment.

Still, I'm curious what the problem was...
 
G

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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.
 

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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 .
What if you are converting 24 recording bit to 16 bit?

Sounds to me like there's some kind of signal processing math going on with this though...
 
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hakukani said:
What if you are converting 24 recording bit to 16 bit?

Sounds to me like there's some kind of signal processing math going on with this though...
Still the same basic issue. It either reads the group of code, or it does not. 16 bit, 20 bit, 24 bit, is all Analog to digital encoding. Once it's digitized, it is either a 0 or a 1 in however many bits. The sound we hear is after the digital to analog convertor, up until then we have no sound. No speed, tempo, tone , NADA ! Just digital code, or as you put it , math.
 

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hakukani said:
What if you are converting 24 recording bit to 16 bit?

Sounds to me like there's some kind of signal processing math going on with this though...
If you are loading a 24-bit recording into a 16-bit session, the DAW will drop the last eight bits of the source code.

I stand by my original diagnosis: that it's a user error somewhere, and not a sample rate issue. But, since the OP hasn't responded in several days, we can only assume that he figured out what he did wrong and doesn't want to fess up to being a n00b...
 

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JCBigler said:
If you are loading a 24-bit recording into a 16-bit session, the DAW will drop the last eight bits of the source code.
That's what I figured.

JCBigler said:
I stand by my original diagnosis: that it's a user error somewhere, and not a sample rate issue. But, since the OP hasn't responded in several days, we can only assume that he figured out what he did wrong and doesn't want to fess up to being a n00b...
Yeah, that's sort of what I meant by signal processing...something in the code that says 'add thirty cents to the whole thing'.
 

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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.
Not necessarily true. The sample rate is the number of samples per second that were recorded and that, of course, must be played back. So, if you record something at, say, 22050 hz and then tell the playback mechanism that it is to be played back at, say, 44100 hz, well, you can figure out what it will sound like. All you have to change in a .wav file is one value to make that happen. All the samples will remain as recorded, i.e. the 16 bit words of ones and zeros that you mentioned, but playback will be kind of, well, off speed and off pitch. A lot.

So, if a bum codec diddles with the sample rate either when encoding or decoding and either on input or output, that is clearly a digital problem not an analog one, and that will definitely generate funny audio.

I'm not arguing that this it what happened here, but without more evidence, and given the OP's assurance that he did everything right, it is the most likely answer.

I doubt that it is an analog problem because the OP's mp3 file plays back on my system with the pitch problems that he complains about. It's a digital problem somewhere in his system. Or else he really plays sharp.

Of course, these mysteries are always less fascinating once we learn the real culprit. We might find that he recorded to a cassette on one deck and ripped it into an mp3 on another, and one of the decks is out of calibration, in which case you would be right. Stranger things have happened.
 

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Discussion Starter · #31 ·
You guys are making my head spin with all this digital sample rate BS. It's very simple...no sample rates were changed when this anomaly happened. I just pressed record in Audition...and this happened. I shut down Audition and tried basic Sound Recorder...same thing happened. I restarted...did a defrag and anti-virus to ensure there were no clogging issues. Same thing happened.

Now the only thing someone mentioned before that might cure the problem, is that I was running off battery power when this happened. Maybe switching over to AC will help? Also, I think (THINK) another variable that could come into play (though I've never had this problem) was that I had a ceiling fan running at the time of the recording. But I used to record on my old Gateway desktop with no problem of pitch change just because of a stupid ceiling fan.

Does anyone have a REAL solution for me, other than hypothetical holy apparition changing of sample rates that I have no idea how to change back? I might want to try recording rite now with AC and see if that fixes it. If not...re-installing the drivers for my soundcard? What else is there?
 

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Discussion Starter · #32 ·
vermontsax said:
Something I had not thought of until now. This is not to suggest that you should not seek the answers to Al Stevens and princeganon's questions - that can only help you.

My questions are these: how old is the laptop? Is it still under warranty? Have you ever had in-tune recordings on this laptop?

If the laptop is under warranty then our work here may be done (though I don't envy you for the number of hours you may spend on the phone with someone much less helpful than the people on this forum).

I would make the argument that if you did not mess with your bios to begin with and you have default settings in the Windows Media Player/Recorder then it's a problem with either the hardware you have or the way the computer was initially set up. Whether Dell will buy that or not, I'm not sure but you should at the least write down the checks you have made on your system to this point so you can rattle them off to their "support" team.

Just a thought. Good luck!
The laptop was purchased in November/December of 2006 for me as a Christmas present (therefore I received it in late December of '06). It's virtually new as I had stated. All recordings done on this laptop since the day I bought it were in-tune, 100%. There was only one recording where I myself recorded sharp. I knew what I did wrong, and fixed it. Now, a month or so later, after use otherwise, I try to record and this freak accident happens.

Guys, I'm not afraid to admit anything I might have done wrong, I just don't know what, if what (JC Bigler apparently wants me to admit that I'm some [email protected]$$ of some sort) I did wrong. I'm not understanding exactly how to check all the sample rate stuff...in Audition it's the usual 44.1K/32 bit. In Sound Recorder, I have no friggin' clue how to check the rates there. Either way...something got messed up, either in the software/drivers (sample rates), or hardware (worst case scenario!). So I'm at my wit's end...
 

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Discussion Starter · #33 ·
Oh, and umm...for the comments about me being a n00b, I dunno if they were said out of fun or malice, but I've been playing my axe for almost a decade, going into college for music ed, and have been a recording artist for almost two years. I'm no n00b to studio stuff...I just don't have acts of the devil happening to my computer on a regular basis, nor do I know how to fix them. I hadn't responded in a while due to my busy schedule as a high school senior (spring concert, preparing for finals) so...lay off, mmk?
 

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JPSaxMan said:
Does anyone have a REAL solution for me, other than hypothetical holy apparition changing of sample rates that I have no idea how to change back?
If you can't figure that out then, you shouldn't be using a digital audio workstation, especially one as advanced as Adobe Audition.

Several people have asked you several times for more information. SUCH AS: when you play back the sharp recording, DOES IT PLAY BACK FASTER THEN THE ORIGINAL?

You have as yet, not answered that question, which is critical to find an answer to your problem.

A fragmented hard drive, or virus infection will not cause this kind of issue (has any ever heard of a virus that infects you computer and automatically adjusts the pitch of your digital recordings?).

Also, you said in your original post that you played along with it on your keyboard. Did you tune to your keyboard also? Did someone change the pitch control on your keyboard? Have you checked YOUR tuning, YOUR KEYBOARD'S tuning with a properly calibrated tuner?
 

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Discussion Starter · #35 ·
Ok...sorry about not answering that critical question. No, the tempo was not changed at all. I know how to change sample rates, just not by digging "deep" into DAW (maybe that's just some fancy word that I don't know). I had to change sample rates, as I might have stated before, when I recorded on an older desktop. It couldn't handle the high sample rate of 44.1K, so I had to lower it to 16K, then convert when the mixdown was made later. Again, I'm no n00b to audio workstations. Just not a techie.

My tuning on saxophone was checked with a tuner prior to the recording made. I actually played along with the recording, and found the horrendous sharp-ness without a tuner. Same thing with a keyboard...but I am going in a bit to record the keyboard with AC power to see if this solves any problems.
 

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JPSaxMan said:
It couldn't handle the high sample rate of 44.1K, so I had to lower it to 16K, then convert when the mixdown was made later.
You have a Core Duo (dual core processor) laptop, and it couldn't handle a 44.1k recording?

Even so, a 16k recording is less quality than a standard phone line (which is about 22k, if I remember correctly). At 16k, you are recording at about 36% of CD quality audio (which is 16-bit/44.1k).

Methinks you did something weird when you converted your recording "at mixdown".

Uninstall Audition, reboot, and reinstall. Start a NEW session, and rerecord. Trash the old session that was recorded sharp.
 

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Discussion Starter · #37 ·
Nah, I did the sample rate conversions on an old Gateway desktop (as I stated before). I've never had to do sample rate conversions on this beauty!
 

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Discussion Starter · #39 ·
Hmm...ok...so I was wrong. I just put a metronome against my piano recording when I recorded. I went back and tried to match the same tempo against it, and it kinda is like when you have two windshield wipers and one is going faster than the other...the two start off in rhythm, then separate, then they come back together, and drift apart. So...the recording is being sped up!

And just for sakes of entertainment, I also disconnected the preamp from the computer and tried just doing a line-in with the keyboard...same thing happened, so it's not the preamp...in case that crossed anyone's mind by chance.
 

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JPSaxMan said:
Ok...sorry about not answering that critical question. No, the tempo was not changed at all.
Not even a tiny little bit? You wouldn't really notice the tempo change you get with that small of a pitch change unless you really listen for it and can compare it to the original.

Now, please answer the other question about the wav file.

And if you really think what we are saying is BS, then you really don't want the help of people who know about these things. So, if you want us to stop trying to help you, tell us again that what we are saying is BS, and I for one will gladly and quietly slip away and let you work it out on your own.

And, no, it can't be because you are running on battery power. It's all DC by the time it gets to the board irrespective of where it originates.

Unless, of course, you are using a turntable and use 60 cps where 50 is expected. :D
 
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