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  1. #51
    Senior Member Julf's Avatar
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    Quote Originally Posted by M-H View Post
    That is something I will explore too, It might not give me the OGG format I love, but the few extra bits of MP3 storage do not cost much anymore , so is not really an argument.
    Haven't looked at the code closely enough, but it should be pretty easy to make mp3fs to use whatever encoder/codec you want.

    Edit: Had a look. Not trivial.
    Last edited by Julf; 2017-07-10 at 01:38.
    "To try to judge the real from the false will always be hard. In this fast-growing art of 'high fidelity' the quackery will bear a solid gilt edge that will fool many people" - Paul W Klipsch, 1953

  2. #52
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    Quote Originally Posted by Julf View Post
    Definitely. They are basically an unnecessarily complicated way of storing 24-bit data in a 32-bit container. Floating point makes sense for data with a widely varying range, but not for well-constrained audio data.
    Tbh there are other reasons people want to do this; programming with a 32 bit word length on all your data is in some ways better and certainly intrinsically more efficient inside the CPU itself, though of course it doesn't add anything of any benefit to the content. We've been running pointlessly 32 bit graphics displays for a decade or two when there's only 24 bits of colour information.


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  3. #53
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    Quote Originally Posted by Julf View Post
    My solution is mp3fs that automatically synchronizes a compressed version of my uncompressed files.
    Heh, I just wrote a script. I wound up with three copies on disk because I then have another ogg set where I've force normalised the audio data (android doesn't support replaygain).


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  4. #54
    Senior Member Julf's Avatar
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    Quote Originally Posted by drmatt View Post
    Tbh there are other reasons people want to do this; programming with a 32 bit word length on all your data is in some ways better and certainly intrinsically more efficient inside the CPU itself, though of course it doesn't add anything of any benefit to the content.
    That definitely applies for 32-bit integers, but with floating point it depends on the FP capabilities of the processor - and going floating point requires care in handling rounding errors.
    "To try to judge the real from the false will always be hard. In this fast-growing art of 'high fidelity' the quackery will bear a solid gilt edge that will fool many people" - Paul W Klipsch, 1953

  5. #55
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    Yes and no, the inherent alignment on word/cache line boundaries can improve performance regardless of whether the data represents an FP or INT value. Subsequent processing in FP brings its own problems though, I agree.


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  6. #56
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    Quote Originally Posted by Julf View Post
    So you would argue that the downsampling (using a high quality algorithm) is audible?
    I wrote it isn't always a good idea to down-sample.

    I was actually assuming high quality down-sampling. My concern is really the quality of up-sampling in older DACs - a concern which is lessened with higher rate input. Newer DACs should have higher quality up-sampling, so down-sampling your files would be less of an issue with these.

    Since there is little up-side to it, my rule of thumb is not to down-sample.
    Check it, add to it! http://www.dr.loudness-war.info/

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