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Recording advantages of 44.1kHz vs. 48kHz revisited

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  • Recording advantages of 44.1kHz vs. 48kHz revisited

    My friend sitting here brought up a good point. We've been conditioned to record 44.1kHz due to CDs being 44.1kHz.



    But since CDs are disappearing, and we use audio for videos more commonly, including uploading 'em to YouTube or playing back MP3s, is there any disadvantage in recording at 48kHz?



    What are your thoughts?
    Ken Lee on 500px / Ken's Photo Store / Ken Lee Photography Facebook Website / Blueberry Buddha Studios / Ajanta Palace Houseboat - Kashmir / Hotel Green View - Kashmir / Eleven Shadows website / Ken Lee Photography Blog / Akai 12-track tape transfers / MY NEW ALBUM! The Mercury Seven

  • #2
    No disadvantage that I know of. I've been tracking at 48k since about 2004 or so.



    Here's something that I can't figure out...



    If you're ever planning on providing hd tracks to something resembling a "market", you better track at 96k or 192 .. like now.



    I always figured I could port stuff or upsample to 96 on existing tracks but you know what?... there are guys in high fidelity land that know when you do that. They can tell from scopes and screens and things.



    How do they know that? That really throws me. If I do a real-time port of a 48k 24bit song into a second daw at 96k... those guys ALWAYS know that the source was 48.



    And then they call foul.



    How do they do that?????? I'm going to have to find out more from those expert high-fidelity nerds.



    Anyway, I'm probably going to ramble on over to 96k pretty soon anyway for tracking.



    There's really no reason to avoid it now I guess.



    Bought my first cell phone this year so anything's possible.

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    • #3
      I
      <><><><><><><><><><><><><><><><><><><><><>

      “Music is well said to be the speech of angels... nothing among the utterances allowed to man is felt to be so divine."

      ~Thomas Carlyle

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      • #4






        Quote Originally Posted by Beck
        View Post

        I

        Transfer to reel-to-reel half-track analog tape first, or at least resample through an analog interface. That way you can
        --
        "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
        Drop by http://mikeriversaudio.wordpress.com now and then

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        • #5
          The only disadvantage to 48K is slightly bigger files, and a very slight (to me inaudible) degradation when converting to 44.1K for CD. The latter is minimized with good software; all are not alike.



          Regarding folks being able to detect upsampling: Just a guess, but I bet if you upsampled all the individual tracks and then mixed at 88K, the upsampling would be obscured. Keep in mind that FX would run at ful 88K, so for any nontrivial FX, it wouldn't be the same as upsampling the result.



          I have a friend who swears that doing this (recording at 48, upsampling to 88 before rendering) and then downsampling made the mix sound better, even though they weren't using any fancy FX, just a bit of reverbs and delays. He says it was a blind study and all the bandmates preferred the 88K mix. I'm skeptical of the claim, but he's a very intelligent and reliable guy. Of course, that's one data point and may not be significant.



          I'm skeptical because upsampling adds noise, downsampling adds noise, and linear FX like delays and reverbs (especially convolutions, but maybe not all reverbs) are *linear* so there's no time-domain improvment in running at a higher rate. Nonlinear stuff like chorus or pitch shift doubling could very easily sound better at higher rates, especially if the algorithms aren't as careful as they should be about anti-aliasing.
          learjeff.net

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          • #6
            I've been recording lately at 96/32-float. I like the sound--- it seems to me that vst/dx filters I apply exert less cumulative damage/noise to the audiofile at those rates. A 44/16 file seems to get awfully distorted by the time you've treated it with various dx/vst filters. I could be wrong. You guys understand these things much better than I.
            Every paint-stroke takes you farther and farther away from your initial concept. And you have to be thankful for that. Wayne Thiebaud


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            • #7
              I've also been recording at 32-bit float.



              In light of what my friend suggested the other evening and what you guys are writing here, I think I'll probably switch back to 48kHz for any new projects.



              Not sure about 96kHz yet. File size, etc. My system probably won't run so well, so I think that will have to wait.
              Ken Lee on 500px / Ken's Photo Store / Ken Lee Photography Facebook Website / Blueberry Buddha Studios / Ajanta Palace Houseboat - Kashmir / Hotel Green View - Kashmir / Eleven Shadows website / Ken Lee Photography Blog / Akai 12-track tape transfers / MY NEW ALBUM! The Mercury Seven

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              • #8






                Quote Originally Posted by MikeRivers
                View Post

                Why would you want to do that? You might obscure the trail, but make the sound worse in the process...




                I mix and master to analog anyway. It doesn
                <><><><><><><><><><><><><><><><><><><><><>

                “Music is well said to be the speech of angels... nothing among the utterances allowed to man is felt to be so divine."

                ~Thomas Carlyle

                Comment


                • #9
                  Storing recorded tracks as 32-bit float wastes 1 byte per sample. There is no additional information in the file; your soundcard gets 20 bits of significant audio data, adds 4 bits of white noise (which is a good thing), and delivers a 24-bit sample. Converting it to 32 bits before storing to disk just wastes space and disk I/O. But if you're not hitting any space or disk bandwidth limitations, then fine. It saves a little CPU time on playback, since the samples don't need to be converted from fixed to float.



                  However, 32-bit float has a huge advantage for mixes that will have any further processing (e.g., mastering). In 32-bit float format, there is no clipping. If you go over 0 dB "FS" (sort of a misnomer, because it's full-scale only in fixed formats), there's no loss in precision, no lopping off of peaks, etc.



                  The only possible harm to going over 0dB in float format is that some DSP algorithms might not be optimized to handle values over 0dBFS well. (DSP algorithms involve a complex set of tradeoffs; in some cases the high-level matrix math looks simple, but in practice intermediates will hold near-infinitesimals or near-infinites, and so the algos have to be modified to avoid these. BTW, this is about the only reason why 64-bit float might sound better than 32-bit float.) But that's far better than clipping. The solution is the same in either case (turn down the gain somewhere, or use compression/limiting, or whatever your favorite technique is for taming rogue peaks. With a fixed format, you have to do it before mixdown. With float, you can fix it after mixdown, and if you're going to do it using master-channel methods, there's really no difference (between fixing it pre- vs post- mixdown).



                  Oh, the only other harm is that you might just play the clip, and it'll sound bad if there's much content over 0dBFS.
                  learjeff.net

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                  • #10






                    Quote Originally Posted by Beck
                    View Post

                    I

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                    • #11






                      Quote Originally Posted by philbo
                      View Post

                      You wouldn't need to go through tape (though you could, and it sounds nice). Just loop the outputs to the inputs with a cable. The filtering on the D/A output will fill in the gaps just fine.




                      That would be easy with two computers. It might also be possible with two DAWs that can run concurrently, and two soundcards.



                      But I don't know of a DAW that can record inputs at one rate while playing to outputs at a different rate. I also doubt that most soundcards can run different ports at different rates concurrently.



                      Plus you wouldn't get that warm tape compression. ;-)
                      learjeff.net

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                      • #12






                        Quote Originally Posted by UstadKhanAli
                        View Post

                        I've also been recording at 32-bit float.




                        You understand that the 32-bit float(ing point) doesn't have anything to do with the sampling. The audio is still sampled to a word length of 24 bits. What's 32 bits is the arithmetic used for processing the data. Since the word length grows with just about every operation, using 32 bit floating point arithmetic prevents the audio data from being mangled by truncating or rounding to fit a 24-bit word.
                        --
                        "Today's production equipment is IT-based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson, Resolution Magazine, October 2006
                        Drop by http://mikeriversaudio.wordpress.com now and then

                        Comment


                        • #13






                          Quote Originally Posted by MikeRivers
                          View Post

                          You understand that the 32-bit float(ing point) doesn't have anything to do with the sampling. The audio is still sampled to a word length of 24 bits. What's 32 bits is the arithmetic used for processing the data. Since the word length grows with just about every operation, using 32 bit floating point arithmetic prevents the audio data from being mangled by truncating or rounding to fit a 24-bit word.






                          I believe Cubase samples at 32 float when set to do so..
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                          • #14






                            Quote Originally Posted by Bookumdano2
                            View Post

                            I always figured I could port stuff or upsample to 96 on existing tracks but you know what?... there are guys in high fidelity land that know when you do that. They can tell from scopes and screens and things.



                            How do they know that? That really throws me.




                            I was thinking it was obvious--maybe they just notice the complete absence of frequencies over 24k. But I've never actually put a high sample rate recording on a frequency analyzer to see if there is any information there at all.



                            This blog is as good an article as any I've seen on the topic: http://people.xiph.org/~xiphmont/demo/neil-young.html

                            Comment


                            • #15






                              Quote Originally Posted by learjeff
                              View Post

                              Storing recorded tracks as 32-bit float wastes 1 byte per sample. There is no additional information in the file; your soundcard gets 20 bits of significant audio data, adds 4 bits of white noise (which is a good thing), and delivers a 24-bit sample. Converting it to 32 bits before storing to disk just wastes space and disk I/O. But if you're not hitting any space or disk bandwidth limitations, then fine. It saves a little CPU time on playback, since the samples don't need to be converted from fixed to float.



                              However, 32-bit float has a huge advantage for mixes that will have any further processing (e.g., mastering). In 32-bit float format, there is no clipping. If you go over 0 dB "FS" (sort of a misnomer, because it's full-scale only in fixed formats), there's no loss in precision, no lopping off of peaks, etc.



                              The only possible harm to going over 0dB in float format is that some DSP algorithms might not be optimized to handle values over 0dBFS well. (DSP algorithms involve a complex set of tradeoffs; in some cases the high-level matrix math looks simple, but in practice intermediates will hold near-infinitesimals or near-infinites, and so the algos have to be modified to avoid these. BTW, this is about the only reason why 64-bit float might sound better than 32-bit float.) But that's far better than clipping. The solution is the same in either case (turn down the gain somewhere, or use compression/limiting, or whatever your favorite technique is for taming rogue peaks. With a fixed format, you have to do it before mixdown. With float, you can fix it after mixdown, and if you're going to do it using master-channel methods, there's really no difference (between fixing it pre- vs post- mixdown).



                              Oh, the only other harm is that you might just play the clip, and it'll sound bad if there's much content over 0dBFS.




                              If you don't have gain staging mastered, you don't need to be worrying about 32 bit floating point vs 24 bit fixed point. You need to be worrying about gain staging. My 2c.
                              Silk City Music Factory: A Connecticut Recording Studio

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