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recorded pro tools 48khz session with clock at 44.1 khz

Discussion in 'Pro Tools' started by monster, Sep 15, 2004.

  1. monster

    monster Guest

    i recorded a session the other day in pro tools at 48 khz. the problem was, the clock was set to 44.1 khz and now i have 44.1 khz files that play back at 48 khz (so they sound sped up - faster and higher in pitch). i can bump the clock back to 44.1 khz (with a 48 khz session) and play it back at the desired speed and pitch, but when i bounce the tracks into a stereo file, i still get the higher tempo file (the bounce happens in the sound card and the clock is external).

    is there a way to turn a 48 khz aiff into a 44.1 without resampling?

    if all else fails, i'll just play back tracks into a DAT after I mix them, then import them into pro tools from the DAT, but there's got to be an easier way, doesn't there?

    PLEASE HELP! :-?
  2. ghellquist

    ghellquist Guest

    do a search on DUC, I tend to remember someone asking about this before. As my faint memory recalls, there is some utility you can use to change the header of the files to read as the correct speed.

  3. tomtom

    tomtom Guest

    i think that if you recreate a new session at 44.1kHz, clock PT at 44.1 and import all the tracks in this new session, It will play at the right speed, display the right timings in PT. PT takes the session sampling rate in account for all calculations (bounces, etc) no matter what the fs it is forced to. That's why you get the higher tempo and pitch even if you bounce. It thinks it is running at 48kHz. With the new session at 41kHz, it will do it right.
    You can then bounce or export your mix at 48 kHz.
  4. monster

    monster Guest

    "if you recreate a new session at 44.1kHz, clock PT at 44.1 and import all the tracks in this new session, It will play at the right speed"...

    unfortunately, pro tools automatically resamples the audio tracks if their sample rate doesn't match the session file. i was hoping there was a way to tell pro tools not to resample when you bring in the files, but there isn't...

  5. monster

    monster Guest

    no dice


    i tried to find a utility on that site and came up pretty empty.

    anywhere else you would recommend?

    it seems like it should be a very simple thing (as i'm sure its probably numbers that need to be changed in the header).
  6. ghellquist

    ghellquist Guest

    Had to search a bit for it. Here is the thread.

    Digidesign thread

    Sorry, no expert on this myself.
  7. monster

    monster Guest


    actually, that is the thread i started over there after you referred me... :D

    i got that issue resolved, now the problem is the quality of the time compression... to make up for the tempo difference (because even though the metronome was set to, 60 bpm for example, the metronome was beating slower than it was set - 55 for example). using the time compression on pro tools to compensate for the difference (approximately .919:1 calculated by the ratio 44.1/48:1) creates all sorts of digital gnarliness.

    i'll have to make a choice soon as to whether or not i'll actually use the pitched up version (and beef up the sound via reverb and eq a bit to compensate for the pinched resonance) or to shoot myself in the head.

    i didn't think the .919:1 would cause that much problem (as i've heard the sound quality doesn't degrade until around 0.8:1 or 1.2:1). any ideas on why i'm getting so much popping at .919:1?
  8. ghellquist

    ghellquist Guest

    Hi monster, I must admit I cannot really follow you. So I´ll try to reiterate.

    1) from some reason the recorded file was marked as 48kHz but it really was recorded at 44.1 (these things can happen when using an external clock).
    2) you have fixed that now, so the file now sounds at the right pitch.
    3) but you now want the playback to go faster (all due to the mistake in number 1).

    Right so far?

    I believe there is a number of programs that allow you to play modify a file to play back faster without changing the pitch. It depends a bit on which kind of material you have recorded what kind of effect you get on that.

    -- percussion type material, for example drums. Perhaps you should do "beat slice" processing. These programs work by "cutting" up the file in small beat segments. Each segment sounds exactly as before, the program simply plays the next segment a bit ahead of its original position.
    -- continuos music type material. I cannot say how a program does this, but it somehow manages.

    So just perhaps, the algorithm in PT LE is not the best in the world for your usage.

    Could you somehow place a snippet of the wav file somewhere for us to look at, and perhaps we could do something from there. Or perhaps describe the type of music.

    I have a mail adress you can mail to if you want to reach me privately : ghellquist (at) yahoo and then .se (mail snoopers steal adresses all the time it seems. It might that I could use some of the programs I have to get you where you want, that´s what friends are for.


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