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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: jessedscott on June 16, 2005, 02:47:39 AM
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I was just talking to Skalinder about this. I resampled a show that I taped on the JB3 @ 48hz in SF 7.0 w/ the alias filter. Going back and listening to it, it seem that the tempo of the show is slower that the original. It also changed the length from 2hrs 36min to 2hrs 49 min. Brian suggested to change the header of the wav file. But didn't really know how to do it see as he doesn't use SF 7.0 I was thinking if I tracked it out first, then go back and change the sample rate of each track/file this might help. Listening to it at 48, it sounds fine, only when I resample does this happen. I usually tape at 44.1, and did this by accident. I have never had this happen, and I'm somewhat confused. Maybe trying to resample the whole thing at once is the problem. :hmmm:
discuss.....
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The time shift matches the 48/44.1 ratio. Sounds like you didn't actually resample it, you just told it to play back at 44.1 which would slow it down (that's the header change). You should also notice a pitch change between the two. Don't do any conversions track by track or they might not match up well.
Question - do you know the real duration of the show? What was the UA-5 set at when it was powered on? (changes to sampling rate on the UA-5 only occur after it's been power cycled). If you start your JB3 before the UA-5 is feeding an optical signal, the JB3 will write a 48kHz header on the file no matter what you feed it, even though it's actually running at 44.1. It will still record all the bits but mislabel the header.
When you're working, one way to spot resampling vs. header changes is the first takes processing time, the second is instantaneous.
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I was just talking to Skalinder about this. I resampled a show that I taped on the JB3 @ 48hz in SF 7.0 w/ the alias filter. Going back and listening to it, it seem that the tempo of the show is slower that the original. It also changed the length from 2hrs 36min to 2hrs 49 min. Brian suggested to change the header of the wav file. But didn't really know how to do it see as he doesn't use SF 7.0 I was thinking if I tracked it out first, then go back and change the sample rate of each track/file this might help. Listening to it at 48, it sounds fine, only when I resample does this happen. I usually tape at 44.1, and did this by accident. I have never had this happen, and I'm somewhat confused. Maybe trying to resample the whole thing at once is the problem. :hmmm:
discuss.....
Sounds like you didn't actually do Sample Rate Conversion but just changed the WAV header. 156 minutes * 60 seconds/minute * 48000 samples/second = 449280000 samples.
( 449280000 samples / 44100 samples/second ) * 1/60 minutes/second = 169 minutes == 2:49:00
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The time shift matches the 48/44.1 ratio. Sounds like you didn't actually resample it, you just told it to play back at 44.1 which would slow it down (that's the header change). You should also notice a pitch change between the two. Don't do any conversions track by track or they might not match up well.
Question - do you know the real duration of the show? What was the UA-5 set at when it was powered on? (changes to sampling rate on the UA-5 only occur after it's been power cycled). If you start your JB3 before the UA-5 is feeding an optical signal, the JB3 will write a 48kHz header on the file no matter what you feed it, even though it's actually running at 44.1. It will still record all the bits but mislabel the header.
When you're working, one way to spot resampling vs. header changes is the first takes processing time, the second is instantaneous.
The UA5 was set to 44.1 +T for the info
I was just talking to Skalinder about this. I resampled a show that I taped on the JB3 @ 48hz in SF 7.0 w/ the alias filter. Going back and listening to it, it seem that the tempo of the show is slower that the original. It also changed the length from 2hrs 36min to 2hrs 49 min. Brian suggested to change the header of the wav file. But didn't really know how to do it see as he doesn't use SF 7.0 I was thinking if I tracked it out first, then go back and change the sample rate of each track/file this might help. Listening to it at 48, it sounds fine, only when I resample does this happen. I usually tape at 44.1, and did this by accident. I have never had this happen, and I'm somewhat confused. Maybe trying to resample the whole thing at once is the problem. :hmmm:
discuss.....
Sounds like you didn't actually do Sample Rate Conversion but just changed the WAV header. 156 minutes * 60 seconds/minute * 48000 samples/second = 449280000 samples.
( 449280000 samples / 44100 samples/second ) * 1/60 minutes/second = 169 minutes == 2:49:00
You're right, I wasn't actually changing the sample rate, but the wav header.
In SF 7.0 I set the sample rate to 44.1 + using the anti alias filter. I thought I was doing this before, but I wasn't. I just made sure I was doing all of that. Everything has worked out. +t for the info....
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so its better to record in 48 then downsample to 44 right??
from the sounds of a reply you want to make sure the a/d, in my case a ad-20, is turned on before the jb3, correct??
what is the plus side to recording in 48 vs 44?
and does anyone use a specific MAC program to resample?
thanks
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I was just talking to Skalinder about this. I resampled a show that I taped on the JB3 @ 48hz in SF 7.0 w/ the alias filter. Going back and listening to it, it seem that the tempo of the show is slower that the original. It also changed the length from 2hrs 36min to 2hrs 49 min. Brian suggested to change the header of the wav file. But didn't really know how to do it see as he doesn't use SF 7.0 I was thinking if I tracked it out first, then go back and change the sample rate of each track/file this might help. Listening to it at 48, it sounds fine, only when I resample does this happen. I usually tape at 44.1, and did this by accident. I have never had this happen, and I'm somewhat confused. Maybe trying to resample the whole thing at once is the problem. :hmmm:
discuss.....
Sounds like you didn't actually do Sample Rate Conversion but just changed the WAV header. 156 minutes * 60 seconds/minute * 48000 samples/second = 449280000 samples.
( 449280000 samples / 44100 samples/second ) * 1/60 minutes/second = 169 minutes == 2:49:00
+T on behalf of your last math teacher ;D