Hi
 
I am working in X3 on digital audio from reel to reel tape recordings.
 
Some of these recordings off the original master tape are at half speed, some at double speed.  
 
It used to be very easy to double or half the playback speed and the pitch would just follow!
 
Can someone help guide me in how to do it in X3?
 
 

Hi and welcome to the forums. The simplest method is to slip-stretch the individual clips (hold CTRL and drag the right edge when see the proper cursor).
 
When completed, for stretches that large, be sure to right click and "Bounce to clips(s)" afterwards to render the audio. The original wav files will remain in your project audio folder if you should need to revisit them at "original speed" later (or you can manually delete them, as desired).
 
Edit: OMG, I really need to cut down on multi-tasking, sheesh... TAPE! See posts below... Adjusting playback sample rate on the file is the best plan of attack (Audacity is the first free one that comes to mind with this capability). More details below.

Well unfortunately there is no varispeed function right inside Sonar (and many people have been asking for this) but there are more manual ways to go about it.
 
First though.... why are these recordings at these speeds? Do you not have the playback device or did the person who handed these files to you not realize the speed was wrong when they digitized them? Ideally you would obviously want to digitize at the appropriate playback speed.
 
However one thing you may want to study up on is the Loop Constructor View. It's too in depth to fully explain in a single post (at least for me) but it has pitch and tempo functions and you can output high quality results (but you really want to do it correctly to acheive that).
 
The other thing that might work is screwing with the samplerate. As in when you change the samplerate of a project to something other than the samplerate of the audio clip it will act kind of like using different speed settings on a tape machine. In fact this may be your actual problem. The files may have been digitized at a different samplerate than the one you have your project set to. I had this problem with some old files of mine. They were playing all slow and low. I change the project to the correct samplerate and all was well.
 
Cheers.

mettelus
Hi and welcome to the forums. The simplest method is to slip-stretch the individual clips (hold CTRL and drag the right edge when see the proper cursor).
 
When completed, for stretches that large, be sure to right click and "Bounce to clips(s)" afterwards to render the audio. The original wav files will remain in your project audio folder if you should need to revisit them at "original speed" later (or you can manually delete them, as desired).

That would not deal with the pitch problem unfortunately but that in conjuction with the Transpose process (Process > Transpose) would work. That is of course really manipulating the audio which ideally SHOULD be avoided as much as possible but done correctly Sonar seems to handle quite well.
 
Edit: And just to add to that I think if Varispeed style functions do get introduced I would actually like to see it integrated in the stretch function. Kind of like hold down an extra modifier key when doing the time stretching of the clip and it lower pitch incrementally based on the stretch like slowing down a tape would. And maybe an extra/different modifier could do stuff like gradual speed down/up.
 
aaand just because I'm day dreaming a Pitch/Speed map type thing that you could draw these types of things into (on unison or independent type level) would be soooooper cool.

Slip streatch keeps the pitch constant while altering duration.

The easiest way to reproduce halfspeed/double speed playback is to record at half
/double the intended sample rate, then alter the header of the sample file to the intended sample rate.

In other words, if your intended sample rate is 48k and you want to reproduce a halfspeed effect, record at 96khz then use a wav editor (audacity for example) to change sample rate value the header in the wav file to 48khz. When you playback the modified file, it will have twice the duration and half the pitch of the original.

Going the other way needs an extra step. Assuming the intended rate is 48khz, recording at 24khz is a bad idea. In that case i would record at 48khz, the modify the rate in the header to 96khz. This modified file will playback at double the pitch and half the duration. The additional step is to downsample the modified file to 48khz to maintain the altered pitch and duration.

The key is modifying the wav file header sample rate directly. This changes the way the file contents are interpreted.

js516
Slip streatch keeps the pitch constant while altering duration.

The easiest way to reproduce halfspeed/double speed playback is to record at half
/double the intended sample rate, then alter the header of the sample file to the intended sample rate.

In other words, if you're intended sample rate is 48k and you want to reproduce a halfspeed effect, record at 96khz then use a wav editor (audacity for example) to change sample rate value the header in the wav file to 48khz. When you playback the modified file, it will have twice the duration and half the pitch of the original.

Going the other way needs an extra step. Assuming the intended rate is 48khz, recording at 24khz is a bad idea. In that case i would record at 48khz, the modify the rate in the header to 96khz. This modified file will playback at double the pitch and half the duration. The additional step is to downsample the modified file to 48khz to maintain the altered pitch and duration.

The key is modifying the wav file header sample rate directly. This changes the way the file contents are interpreted.

I did not know it could be done that way. Cool!

http://www-mmsp.ece.mcgil...formats/wave/wave.html

If you're interested in wav file hacking, here's a good definition of the file spec. However, you're better off using a wav file tool to manipulate the sample rate (audacity can do this) instead of a hex editor. ;)

I'm not familiar with slip-stretch etc ... but FYI ...
 
I was playing with a free Beta VST recently that had a few tricks up it's sleeve.
ToneCarver tcStretch for time stretching (ridiculous amounts), pitch shifting, and blurring.
You might be able to do something with the routing on that,

js516
http://www-mmsp.ece.mcgil...formats/wave/wave.html

If you're interested in wav file hacking, here's a good definition of the file spec. However, you're better off using a wav file tool to manipulate the sample rate (audacity can do this) instead of a hex editor. ;)

Thanks. I guess I really should learn exactly what a wav file is and how it is constructed considering I use and create the buggers all day. I kind of always left the bits and bytes mangling up to the computer/programs but I supposed if I want to really advance that's the type of crap one should know.
 
lol... I just wanted to play my guitar. Now i'm turning into a nerd.
 
It's a good thing. ;-)

Changing the wave file header with an editor will indeed change the way the wave file is interpreted. But the playback will likely not give the result you want. Tape is a physical medium with more or less continuous analog modulation. As the speed of the tape over the head is altered, it will produce a more or less continuously variable change in pitch/duration. A wave file is a collection of samples of a continuous analog signal. It can represent an analog signal via the sampling theorem, but changing the rate at which the samples are played back, alters the relationship between the original signal and the digital model. Re-sampling is what is required if you do not want to lose fidelity, and there is more to re-sampling than just changing the sample rate in the wave header. When changes as large as 2X are contemplated, the use of filters may be necessary to avoid aliasing. A good varispeed algorithm is going to require more than just a sample rate change.