Importing audio files into Cubase

[gretschzildjian]

Question. Since some time I have been fooling around with Cubase VST32 on my home PC. I encounter this problem. Every time I import an audio file into a track (whether it's through the menu or by click and drag) the file's speed and pitch alter. That is, the sound is played in a faster tempo and the pitch is higher.

 

How do I prevent this?

 

Thanks,

Vaas

[Ace Of Bass]

you need to change the sample rate of the file to the same as what cubase is using.

if your files are 16-bit 44100khz, you need to set cubase to that, or upsample them to 24-bit 48000.

[Ace Of Bass]

I realize this isn't quite what your problem is, but i suspect it's the same root:

 

The following article was published in the October 2004 issue of EQ Magazine, and appears courtesy of EQ and Music Player Network. http://www.eqmag.com/ Sample Rate Mismatch October 2004 By Todd G. Tatnall DAW users often report this problem, "My song sounded fine when I tracked and mixed down. But when I burned an audio CD and played it back, the music was pitched down about a half-step and sounded slow. What happened?" THE SETUP Let's start by looking at how this happens. We'll use Digidesign Pro Tools/Digi002 and the Presonus DigiMax for this example; however the problem can occur with just about any software and digital hardware if the conditions are correct. The problem begins when a digital connection is made to the audio recording hardware. The DigiMax's eight mic preamps are sent digitally via ADAT lightpipe optical to the optical input on the Digi002. In order for the Digi002 to receive the digital audio signal properly, we need to set its "clock source" to "ADAT." This is necessary because each device has its own internal clock, which determines at what sample rate it should operate: 44.1, 48, 96kHz, and/or whatever. Just like two wristwatches almost never "tick" at the same instant, and often tick at slightly different speeds, digital audio sample clocks are seldom precisely the same. By setting the clock source of the Digi002 to ADAT, we are asking it to synchronize to the speed of the DigiMax internal clock through the optical ADAT cable. If we don't do this, we may experience noise, audio spikes, or lack of audio signal. THIS IS IT Because the Digi002 is now setting its clock speed by following the DigiMax, it's essential to make sure both items are set to the same sample rate. Here's where the problem occurs. Let's say the Digi002 and the Pro Tools session are set to operate at 44.1kHz while the DigiMax is operating at 48kHz. The Digi002 is synchronized to operate at the speed of the DigiMax clock; however, the Pro Tools software still thinks it's recording at 44.1k. Like most DAWs, Pro Tools records audio by creating individual audio files for each track. In addition to the audio data, the file contains information about how it was created, including the sample rate at which it was recorded. The result in this case is that files that were recorded at 48k are "labeled" as 44.1k files. Anyone remember records? Vinyl? Near the end of their era, records operated at either 33 or 45 rpm. Suppose a record that was meant to operate at 45 rpm accidentally got a label that said it was 33 rpm. If you set your turntable to 33rpm and played the record, the music would be very slow and low-pitched. This is essentially what's happening in our sample rate mismatch scenario. The file was recorded at one sample rate, but improperly labeled with another. Back in our digital recording world, the problem continues during mixdown. The engineer uses "Bounce to Disk" in Pro Tools to create a stereo mix of the audio tracks, keeping the Pro Tools sample rate at 44.1kHz for the bounce in order to comply with the Red Book audio CD standard. You can play this "mislabeled" stereo file back in Pro Tools (which is still using the Digimax 48k sample rate), and it sounds fine. (Remember that Pro Tools and the Digi002 think that they're operating at 44.1k while in reality the DigiMax is setting the speed of the Digi002 to 48k, making the file sound correct.) But when the file has been burned to an audio CD, the slow, "pitched down" sound will be heard. When you play the disc back in a CD player, it's now playing at 44.1 kHz - slower than it was originally recorded. WHAT TO DO, WHAT TO DO? So are our tracks ruined permanently? Fear not, there are some things you can do. In our example, we can fix this right inside Pro Tools. Start by setting Pro Tools' clock source to "internal." Create a new 44.1k session and import the problem file. Select the Time Compression-Expansion AudioSuite plug-in, and set the Ratio field to 0.919:1 (the ratio of 44.1 to 48). By shortening the file, we make it play faster - as fast as it was originally recorded. Then using the AudioSuite Pitch plug-in, set the Ratio to 1.088:1. This raises the pitch of our selection by a half-step to its original state. If you're not using Pro Tools, most DAWs/audio editors have similar tools for time compression/expansion and pitch shift. Unfortunately, these types of processes aren't always completely "clean" - you may hear some artifacts from the processing. But at least you're back at the right speed and pitch, and you've saved the files. PREVENTION IS THE BEST CURE Of course, the best way to fix sample rate mismatches is prevention. Always make sure that any devices connected to one another digitally (and the software that uses them) have the same sample rates. In our example, the Digi002 has sample rate lights on the hardware that indicate the current sample rate. If the Digi002 detects a large enough difference between the speed of its internal clock and the external clock (say 44.1 versus 48k), the sample rate light will blink. Watch for this sort of indication any time you set your clock source to external.

 

[fretless]

hmmm sounds odd , I use VST32 ,and you just drag and drop or import , that easy , maybe Ace Of Bass is right . I was thinking maybe the "master click" had something to do with it Did you try for help at http://forum.cubase.net/phpbb2/

[sneakyjesus]

yo G

 

go find a freeware version of goldwave.

It's a program I use to convert mp3s to wavs and it'll do any audio diital conversion at any sampeling rate. and it doesnt' try to rip you off for working with mp3s... like tools.

[Ace Of Bass]

 

Originally posted by sneakyjesus yo G go find a freeware version of goldwave. It's a program I use to convert mp3s to wavs and it'll do any audio diital conversion at any sampeling rate. and it doesnt' try to rip you off for working with mp3s... like tools.

 

 

Hey now...mp3 is a privately owned and licensed technology. if you wanna use the tech in your program, you gotta pay for it. ditto for wma. not the fault of the programmers--they gotta pay to use it, so it's hard for them to make freeware mp3 encoders.

The alternative is to go open-source, like OGG....or, develop a new technology that also encodes MP3s, and make that open source--that's what LAME did, in a kinda sneaky way around the legal issues, which is why new programs based on LAME are free.

 

MP3 technology is patented by the Fraunhofer Society.

[sneakyjesus]

can you explain this more? does this mean that thye make cash off of companies who allow you to export import use protools?

 

Or sites that sell mp3s?

 

where does legality begin?

 

 

I'd never heard of this =0

[Ace Of Bass]

If someone invents new technology, it's theirs. They can do what they want with it. Software usually gets copyrighted. new algorithims often get patented. Realaudio, mp3, wma, and whatever it is that mac uses were all developed by different companies.

I think it's just the encoding that is licensed, at least for mp3s. You have to pay the company to use their encoding software. Sometime you notice that in freeware programs that can encode to various formats or to WAV, but you have to pay a bit extra to rip to mp3. More often you don't notice this, because it's included in the cost of the software (Nero, or Windows if you use Media Player.)

I'm not sure if you have to pay for the license to put out a software program that pays mp3 files; i don't think you do.

 

Realnetworks, on the other hand, doesn't let anyone else make software that plays their realaudio/video files. Of course, their player puts a lot of spyware and other {censored} on your computer, so I just don't use Realaudio.

Then not to long ago, someone figured out how to play realaudio/video files without paying for the license (basically figured out the codec on their own) and made "realalternative" which is freeware.

Realnetworks has since released Helix, and open source version.

 

Microsoft's WMA format is much better than mp3 formats, and they're trying hard to license it out to all the mp3 players on the market to increase publicity.

Apple also has their own format, which they refuse to let anyone else use. So, if you want to play apple-format stuff on anything, it has to be either on an apple ipod, or on apple software (itunes.)

Similarly, they don't want to let mcirosoft have any cut in the market, so they won't accept any wma technology to be used on their products.

 

There are many, many pissed off people now because their mp3 players don't support WMA-DRM files, which are becoming the standard for many legal downloading sites and programs. Most mp3 companies are releasing models that will handle these files.

Apple isn't, because they want everyone to download from itunes--and they obviously don't use WMA.

 

It's just another format war, like with VHS Vs Beta, DVD-R vs DVD+R, and Blu Ray vs HD-DVD. Competition pushes technology forward, which is good, but it means half the people end up pissed off because they picked the wrong side.

[FreestyleIntruder]

 

Originally posted by Ace Of Bass DVD-R vs DVD+R

 

 

I haven't even bothered to begin to understand this one - can someone explain it to me, besides the obvious fact that the two are incompatible. Why are there two different standards here? Is the industry just being awkward?

[Ace Of Bass]

Originally posted by FreestyleIntruder I haven't even bothered to begin to understand this one - can someone explain it to me, besides the obvious fact that the two are incompatible. Why are there two different standards here? Is the industry just being awkward?

 

Well, there was no standard for recordable DVDs when DVDs were released. There was no technology. Someone had to invent the technology.

There was more than one way to do it.

Pioneer did it one way in 1997. (DVD-R), and got it to work with -RW in 1999.

Sony, Phillips, yamaha, dell, HP and a few others worked together to create another way (DVD+R). I think they kinda gave up in 1997, although to their credit they did have a DVD+RW by 1997, but it was only 2GB. They revived their efforts later to compete with Pioneer and brought out the 4.7GB version in 2001. They came out with the Dual Layer DVD+R in late 2003 .

Pioneer then followed with the dual layer DVD-R in 2004.

 

This format war wasn't ever won--manufacturer developed DVD players that would work with either one, so they're both on the market.

The main advantage is that DVD+RW can be written like a floppy drive, in stages, where as with DVD-RW needs to be clean wiped to re-record on it, like with CD-RWs. I think the DVD-RWs are cheaper though.

As far as regular DVD+-R goes, though, there's basically no difference nor any reason to use one or the other, since all major players are multi-format.

[FreestyleIntruder]

Aha! Muchos thankos

[SteveyDevey]

Originally posted by gretschzildjian Question. Since some time I have been fooling around with Cubase VST32 on my home PC. I encounter this problem. Every time I import an audio file into a track (whether it's through the menu or by click and drag) the file's speed and pitch alter. That is, the sound is played in a faster tempo and the pitch is higher. How do I prevent this? Thanks, Vaas

I don't have much experience with cubase, but it's my understanding that it's similar to Acid, and I'm pretty darned familiar with Acid... On that note, I've encountered a similar "problem" in acid. It's not a problem per se, but a setting. You may need to find something to tell it not to try to beatmatch and/or adjust it to match the rest of your song. It may thing it's a looped segment or something of that world, and it's trying to smoosh it into place with the rest of your tracks.