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changing file speed with Audacity (audible quality losss ?)

Discussion in 'Audio' started by bomber07, Sep 16, 2010.

  1. bomber07

    bomber07 Member

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    Hello,

    If I was to change the speed of a 3-4 minute song, either slowing it down or speeding it up by about 5% using Audacity, and then changed it back to it's original speed, would there be any "audible" quality loss ?
     
  2. Ripper

    Ripper Active member

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    Not if you save it at the same bitrate, I shouldn't think. Obviously the song would just sound slower or faster..
     
  3. Mez

    Mez Active member

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    The loss depends on the original bit rate. The higher the bit rate the less loss you have. Lossless does not degrade at all. HiFi lossy will, but you will not be able to notice the loss since it is negligible. At 128 you may be able to hear a difference maybe not. It will depend on your equipment and the music. Electronic music and voices do not degrade easily.

    Why would you change the speed of your music then reverse the change? Why not use the original when you are done?
     
    Last edited: Sep 16, 2010
  4. bomber07

    bomber07 Member

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    Well I'm refering to WAVE files and saving as WAVE.
    I sometimes do it with live music from old cassette masters that play the wrong speed when you try to transfer them to digital..., so I try to fix the speed.
    I never have to change them by more than 5% so that's why I asked in the 5% range.
    I know speed changing is tecnically a lossy process because it resamples, but was wondering if there is technically a definitive answer as to whether changing backwards and forwards in a 5% range would create audible quality loss ?
    No point keeping a cassette once it's gone to digital. So was wondering if I ever changed the speed around 5% again would it have audible quality loss ?
     
  5. Mez

    Mez Active member

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    Then do whatever the quality will not degrade. I suggest not keeping them as waves but as highly compressed flac or ape. They are still lossless but take up far less space.
     
  6. bomber07

    bomber07 Member

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    Yeah most of my stuff is FLAC, but for some reason I have it set to export as WAVE, the size doesn't matter as I usually then burn it to CD, DVD-A or whatever and delelte the file from my HD, but yeah FLAC is definately better when storing stuff on your HD!, I just did a big clean out of all the files taking up space last week !

    At first your comment "then do whatever the quality will not degrade" confused me, I must have read it wrong but after having read it again I take it you are saying if I change a file 5% then change it back to the original speed there will be no audible quality loss ?
    It's getting too late for this stuff tonight... haha
     
    Last edited: Sep 16, 2010
  7. k00ka

    k00ka Regular member

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    AFAIK, Audacity always re-encodes the output..So there will be additional quality loss..Will it be audible quality loss?..Since none of us share ears, equip, listening environments etc..only you can answer that..Try it and see-- err hear for yourself..
    Jm2cents..
     
  8. Mez

    Mez Active member

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    My opinion is, no loss if the source is lossless.
     
  9. davexnet

    davexnet Active member

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    High Quality software of this kind costs hundreds of $$$.
    http://www.prosoniq.com/editing-products/timefactory-2/

    It does the same thing, why is it so expensive?
    Because it uses an algorithm that preserves as much as possible
    the original sound quality.

    It's not even necessary to change it and change it back.
    The first change already lowers the quality - but as mentioned,
    it may be very subtle or it may be obvious, perhaps depending on the material.
     
  10. Mez

    Mez Active member

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    Dave, I am extremely suspicious of your claim that it will preserve more sound quality. It is common knowledge lossless to lossless does not lose quality. For one thing algorithms imply logic to intelligently alter values. A lossless to lossless conversion ought to mean a simple transfer of data into a new format not altering values. Why would you want to alter values? The only purpose of maintaining lossless is to preserve the exact values. Lossless is always some form of a wave file which is merely a string of digital values in a time index. Flac and ape files use lossless compression CDAs add check sums but in all cases the core is just a sting of data in a time index. The digital age relies on lossless compression as being lossless. We use compression to transmit most digital data. I hope you are not claiming that this does not work. Lossless compression is like zipping a wave file. Are you implying that you lose quality when you do this or when you unzip it?

    I think you buy that software because of the time saving features. If your time is money then it is smart to buy the software. Building software is expensive and if you do not expect to sell much you need to charge more money.
     
  11. k00ka

    k00ka Regular member

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    Agreed, of course Mez..My response was/is intended @ Audacity first uncompressing and then re-encoding a lossy source..Guess I should have read more carefully when the OP stated they were working with .wav
    And FWIW, I also prefer storing/archiving in Flac format..
    Cheers!!
     
  12. Mez

    Mez Active member

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    I expected that was the case. That is the point of lossless. You would not archive as lossless unless there was a good reason.

    My guess is you couldn't hear any difference if you used a 320 BR mp3 but there should be some degradation. I do not convert lossy music so I wouldn't know. I do convert lossy audio books to a more compressed version. I now compress to 16 BR m4a. I get better quality with m4a at that bit rate.

    Did you ever check out audio myths in the top sticky? There was a startling demonstration of degradation, but with sound cards. Pros use professional grade sound cards because they do capture sound more accurately. They ran a side by side sound clip taken at the same time index from both cards. They sounded the same. That was no surprise especially because the video was low quality audio. However when I couldn't hear and difference with the pro card and the clip captured buy the cheap card then played and captured, what 150 times I was shocked. I never went to check out the actual files since they wouldn't have posted them if you could hear a difference. There is a huge difference between what can be detected and what can be heard.
     
  13. davexnet

    davexnet Active member

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    Hi Mez,
    aren't we talking about a time stretch operation on the audio (either longer or shorter
    compared to original?) - that's not lossless, the data has been changed.

    If you save it again as WAV, it will be the same quality as it sounded in the
    audio editor after the stretch operation - however, it has degraded slightly
    compared to the original file because of the operation you did on it.

    I have Sound Forge 8 on my PC. Internally it works on uncompressed files.
    If I do a common operation in the editor, for example resample from 48000 to 44100,
    the control has an accuracy setting, low and high. The "high" setting takes 3 times longer than the "low" to complete the operation. The "low" result gives an audible
    loss of fidelity, while the "high" sounds virtually identical to the source .

    It's the same with any operation on the data. Better algorithms degrade the sound less.
    In the example above, the "low" setting would be analogous to the stretch in
    Audacity, while the "high" would equate to Prosonic TimeFactory.
     
  14. k00ka

    k00ka Regular member

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    @ Mez,
    Yeah I have a hard time ABXing a lame encoded mp3@ 320kbps CBR, from the original file, but frankly when I encode from my lossless flacs to lossy, I prefer lame -V2 ~190kbps, for my portable(s)..I could go lower but -V2 is a good setting and being transparent to my ears from the original..I just don't see the point in using 320Kbps CBR, when using VBR aims for a quality level and not targeting a specific bitrate..Might as well stick to lossless if using the highest MP3 setting (320),IMO..
    Lately I've been using ~256 kbps VBR/AAC/m4a, using Quicktime via foobar2k, and sometimes I use Nero/AAC/m4a..
     
    Last edited: Sep 17, 2010
  15. Mez

    Mez Active member

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    Good point Dave! I do not know enough about audacity to know exactly what it does in the edit mode. It must use an algorithm to alter the data. It would have to alter every point. Since I don't do anything like that, that sticky point completely slipped my mind. I leave hiss, pops, etc in my audio captures because when I thought about it, I didn't trust the editor. Plus, I can hear the track came from a vinyl source and it gives me the warm fuzzies (just like old times).

    Thanks for keeping this thread on the right track.
     
  16. Mez

    Mez Active member

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    You are probably doing it better than I. I rip to high VBR (0) then archive my original CDs. I then use the mp3s to listen to and make CDs. My quality audio devices play mp3s and the other players are crap. Since my wife plays her CDs so I must I archive them as lossless. It is a half ass process when it comes to CDs butit works for me. I can't tell the difference. I don't really see enough space savings to go lower but I bet I could fit more tunes on my 4G ipod if I did. I know I can go as low as 160 BR and not hear a quality loss for most music. I also have a 8 g view which I don't like as much especially for audio books.

    I don't like quick time because it loads as start up and 99% of my music is mp3. Have you heard any artifacts? I bet not. Hydrogen Audio has a long list of defects for AAC. I suspect you might never hear one. I often use the Helix encoder which also has defects but the defects are all very high and soft so the are really impossible to detect by ear and there are only a few know artifacts vs AAC. I use Helix/DNA/Guess because it is lightning fast if I am converting an archived lossless file. When ripping, speed is not so much an issue for me so I use Lame with no known defects.
     
    Last edited: Sep 17, 2010
  17. k00ka

    k00ka Regular member

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    Can't say I'm fond of Qucktime either..I don't use the media player, I actually just use the Quicktime encoder, (QuickTime 7.6.7 (qtaacenc) via foobar2k..This is using a true VBR bitrate encode..And my ipod luvs it, as do I..
    Got all the info over at HA, and have been using it ever since..Works for me thus far(knock on wood)..
     
  18. davexnet

    davexnet Active member

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    Mez, I think you're right, sometimes it *is* better to leave it alone.
    One thing I noticed when listening to captured needle drops, the digital sound
    had retained the warmth of the vinyl! It amazed me when I first heard it.

    In general though, any operation you do try, (be it noise reduction, EQ, compression,
    etc,etc) I found out that often less is more. It is so easy to make the sound worse
    by applying too much effect.

    This has been an interesting thread, pleased to have helped.
     
  19. Mez

    Mez Active member

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    K00ka, I prefer the concept of true VBR over CBR. VBR equals constant quality.

    Dave,
    Most of my vinyl is in good shape so there is not much noise and no reason to clean it up unless you need to listen to sterile music.

    I might have an interesting thought/theory to vinyl 'warmth'. Back before the age of CDs I knew an audiophile that bought 'single crystal' audio cables to link his turntable up to his preamp. The single crystal wires are supposed to have less impedance than regular wire. He was amazed at how crisp the sound was after putting in his cables. When he got his CD player for a crazy price, he remarked how close the CD sounded like his super duper turntable with the single crystal patch cords. I never heard his system.

    I am wondering if the vinyl warmth is due to a bit of a distortion due to such low power going through the wire that the wire can color the music. Like patina, the warmth is attractive what ever the cause.
     
  20. bomber07

    bomber07 Member

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    Some info from the "AUDACITY" forum specifically on how the programme performs this process, it seems to me that it's extremely unlikely I will hear a difference if I change the speed back and forth by just 5% a couple of times :
    ----------
    Yes, I do a lot of testing of Audacity. Yes Change Tempo and Change Pitch cause noticeable sound quality loss, particularly with large changes. Yes Change Speed causes a measurable distortion, but it is primarily "phase shift" which is virtually inaudible. There is also some slight degradation in "transients", but again it is virtually inaudible (to my ears, even using good monitoring equipment). The third thing is a loss of high frequencies - this becomes more noticeable as the amount that the speed is changed increases - for small changes (a few %) the difference is subtly noticeable with good monitoring equipment and good hearing. For larger changes (around 100%) the loss can be very obvious resulting in the sound becoming very dull.

    (All three of these symptom essentially come down to a loss of bandwidth)

    I believe that the difference between the Pitch/Tempo change and Speed Change is that the first two require "time stretching" (for which Audacity uses "SoundTouch") whereas for Change Speed all that is required is resampling. By default Audacity uses libresample, which is a high quality resampling library. If Audacity is built from the source code it can be built to use libsamplerate which is an even higher quality resampling library, but there are some licensing issues that prevent Audacity from being distributed with libsamplerate.
     
    Last edited: Sep 17, 2010

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