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Thread: The Hardware / Audio / Video Thread.

  1. #11
    Senior Member banditwolf's Avatar
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    So looking at the Apple lossless format, will that work in other devices? It is an MP4 file so it would need to support that then to play?
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  2. #12
    Senior Member zborgerd's Avatar
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    Quote Originally Posted by banditwolf View Post
    So looking at the Apple lossless format, will that work in other devices? It is an MP4 file so it would need to support that then to play?
    On the random occasions where I've encountered ALAC or APE, I always convert it FLAC based upon principle alone. Each format has its advantages but FLAC is the most universally compatible. ALAC does work in some things other than Apple devices though. It's getting better.

    ALAC has had an Apache license now for many years, even though it had been reverse engineered. Most things should support it by now. It has an MPEG M4A container but is not an MPEG codec internally.

    Do you have any devices where you would need decoding? I use Google Music Manager to automatically encode my FLAC collection to max bitrate MP3 for playback while driving, skating in the garage, playing music while cooking / cutting the grass / shaving, and other such tasks. https://support.google.com/googlepla.../1100462?hl=en

    I can play my whole collection anywhere with the Google Play Music app, but I use my FLAC collection while at my desk.

    I wish that Google would use high quality VBR though, because 320k CBR is stupid. At any rate; I would pay Google or someone big $$$ to archive my music in FLAC rather than MP3. I suppose I could transfer it all to "cloud storage" but it would be nice to have a single archive that preserves the original integrity and can stream lossy versions for applications that demand it.

  3. #13
    Senior Member zborgerd's Avatar
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    Tempest-level ramble here.

    The loudness war just sucks. I hate how most music gets slammed against the ceiling to make everything seem loud for the flat / portable / crummy bluetooth devices out there.

    Trying to tweak hi-fi equipment and annoyed with how bad most recordings are. A ham-fisted approach is to apply light dynamic range compression for 44.1 kHz stuff that is too loud. I don't like that because it exacerbates the problem, but at least allows some level of equalization. There is also soft clipping which kinda works and changes the sound of the clipping. There isn't anything else that you can do when you have crap at the digital portion before it even gets to the DAC point. Basically the audio has already hit the brick wall in the digital realm. If you use digital equalization, it just creates clipping. Might be possible to use samplerate conversion pre-EQ and gain some headroom. I'm still dorking around with it.

    Everything is mastered these days so that every frequency is at clipping point and allows for little equalization. The point of >44.1 kHz isn't so much that it needs to be much higher; It's that everything is mastered by morons that slam the audio against the rooftops. Even loud 24 bit recordings (like Death Magnetic that I mentioned in the music thread) sound far better in 88.2 because the master isn't hitting the clipping point.

  4. #14
    Senior Member banditwolf's Avatar
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    Not very tempesty. I give it a 2/10. Try again. Needs more cold loads, spam and purell.
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  5. #15
    Senior Member Gypsy's Avatar
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    Your post made sense, therefore, not tempest level. Ruling is final.
    Bad media connoisseur and frequent avatar changer.

  6. #16
    Senior Member zborgerd's Avatar
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    Quote Originally Posted by Gypsy View Post
    Your post made sense, therefore, not tempest level. Ruling is final.
    To elaborate, it only recently occurred to me how much of a problem equalization is in the digital realm when you are processing modern music in a 16 bit / 44.1 kHz space.

    The issue is not that 16-bit depth / 44.1 kHz is not enough room for human hearing. It's just that any substantial digital equalization is going to push an already crushed song outside of the level of amplitude that can be sustained in the bitstream. The result is distortion from the clipping point. There is a minimum and maximum level in that digital realm, even before it hits the DAC and gets to your amp / speakers / and ears.

    The conundrum is processing and conversion, applying the EQ at the absolute right point.

    I think that, in theory, digital EQ for slammed recordings would be applied only after converting to 24 / 88.2 for CD audio. A lot of DACs don't support 88.2; Mine included. It's a perfect multiple of 44.1 kHz and is only slightly lower than 96 kHz.

    The tradeoff is trying to do it with minimal CPU resources to get the desired processing. Alternately, the gain on the EQ could be dropped to the max level of added EQ filters, but that would produce noise. The alternative is just to leave it all flat without any EQ, and then use an external analog EQ instead. That means that you can spit out raw 16 bit / 44.1 kHZ without slamming it any higher to clipping point, and let the analog processing fix it.
    Last edited by zborgerd; 11-28-2016 at 02:21 AM.

  7. #17
    Senior Member zborgerd's Avatar
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    Attenuating the EQ preamp seems to produce the least destructive results. Effectively, drop your EQ premap by X dB from your highest digital EQ filter gain. Basically, EQ for your speakers and amp, but drop the overall gain of the bitstream to fit it within the limits of the processing if you encounter clipping.

    I tried two separate EQ profiles;

    1. Highest filter is +5 dB. Preamp at -5 dB.
    2. Preamp at 0 dB. All filters are -5 dB from the above level.

    The audio is identical, so it seems safe to assume that most digital EQs probably just add / subtract X from every filter where X is the preamp value.

    I suppose that the point of a digital EQ is more about understanding attenuation rather than gain, though each is just the inverse of the other.

    I know it all seems obvious, but sometimes the obvious seems less than obvious until you consider the boundaries of the technology.

    Furthermore, I run Linux. After some consideration, I have this suspicion that a lot of proprietary Windows drivers do all of this automatically, under the hood, to prevent people from complaining about audio quality. They dynamically adjust gain of the entire audio bitstream to fit within the clipping boundaries of the digital EQ.
    Last edited by zborgerd; 11-28-2016 at 04:15 AM.

  8. #18
    Administrator vicireland's Avatar
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    Quote Originally Posted by zborgerd View Post
    To elaborate, it only recently occurred to me how much of a problem equalization is in the digital realm when you are processing modern music in a 16 bit / 44.1 kHz space.

    The issue is not that 16-bit depth / 44.1 kHz is not enough room for human hearing. It's just that any substantial digital equalization is going to push an already crushed song outside of the level of amplitude that can be sustained in the bitstream. The result is distortion from the clipping point. There is a minimum and maximum level in that digital realm, even before it hits the DAC and gets to your amp / speakers / and ears.

    The conundrum is processing and conversion, applying the EQ at the absolute right point.

    I think that, in theory, digital EQ for slammed recordings would be applied only after converting to 24 / 88.2 for CD audio. A lot of DACs don't support 88.2; Mine included. It's a perfect multiple of 44.1 kHz and is only slightly lower than 96 kHz.

    The tradeoff is trying to do it with minimal CPU resources to get the desired processing. Alternately, the gain on the EQ could be dropped to the max level of added EQ filters, but that would produce noise. The alternative is just to leave it all flat without any EQ, and then use an external analog EQ instead. That means that you can spit out raw 16 bit / 44.1 kHZ without slamming it any higher to clipping point, and let the analog processing fix it.
    It's pretty much an epidemic in music mastering. Summon Night 6's opening movie song is massively overcompressed. The waveform is one big log, with almost no variance. I know the original is not that way because we have the master recording. So, I've asked Japan to not kill the English version when it's integrated. All I can do is hope.

  9. #19
    Senior Member zborgerd's Avatar
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    Quote Originally Posted by vicireland View Post
    It's pretty much an epidemic in music mastering. Summon Night 6's opening movie song is massively overcompressed. The waveform is one big log, with almost no variance. I know the original is not that way because we have the master recording. So, I've asked Japan to not kill the English version when it's integrated. All I can do is hope.
    To be fair to the people that master these recordings, it's not entirely their fault. Since everything is recorded and mastered in a 24 bit depth, they have to do a bit of work to get it into an acceptable 16 bit space. They compress the range to fit within a 96 dB range (24 bit is 144 dB SN ratio) and dither to accommodate. I assume that far less time goes into this than initial mastering for their high end gear.

    My problem is probably mostly to do with my EQ flow. I'm using a software EQ in the player. I think it's EQing while the FLAC is still 16 bit. I speculate that I could convert to 24 bit and EQ elsewhere, closer to the OS and hardware, with an LADSPA plugin. I've considered this in the past but never liked the performance. Not so much of an issue on new hardware. Going to try later with an alternate EQ.
    Last edited by zborgerd; 11-28-2016 at 04:38 PM.

  10. #20
    Senior Member Lord of Pirates's Avatar
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    Quote Originally Posted by zborgerd View Post
    To be fair to the people that master these recordings, it's not entirely their fault. Since everything is recorded and mastered in a 24 bit depth, they have to do a bit of work to get it into an acceptable 16 bit space. They compress the range to fit within a 96 dB range (24 bit is 144 dB SN ratio) and dither to accommodate. I assume that far less time goes into this than initial mastering for their high end gear.

    My problem is probably mostly to do with my EQ flow. I'm using a software EQ in the player. I think it's EQing while the FLAC is still 16 bit. I speculate that I could convert to 24 bit and EQ elsewhere, closer to the OS and hardware, with an LADSPA plugin. I've considered this in the past but never liked the performance. Not so much of an issue on new hardware. Going to try later with an alternate EQ.
    See also: Gonna be dead shortly. Isn't recording at 24 (or higher) mostly to due with purposely giving themselves more room to work with? I was under the impression that 80 - 90dB was the standard target for maximum loudness regardless of bit-depth.

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