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Why Don't They Also Make . . . ?


MikeRivers

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A "Normalize graphic waveform display" function for DAWs? Then I could stop telling people not to worry that their tracks aren't hot enough when the waveform doesn't fill the entire track area (or to worry, if it does). We'd get a lot fewer worried new users and probably cleaner mixes if they set the record level correctly and didn't worry about the size of the squiggles.

 

I really think this is a great idea. Anyone else?

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How about just educating these lumps of nescience, instead? I get so sick of "engineers" who can't think their way through the simplest logic problems. What on earth went wrong with the educational system that you end up with people who imagine themselves to be "engineers" yet who can't seem to figure out how to do simple troubleshooting?

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Yea its and issue if you dont understand that the maximum meter reading is 0db and if you pin the meter you're in the digital distortion zone. I rely on my meter peak holds more than what the waveform appears to be in either case. I can use the waveform display to identify if I had a low meter tracking and I can pretty much spot the general DB the tracks came in at. They are also handy for editing parts. But for actual mixing they can be very decieving.

 

I was thinking of doing some screen shots from something tracked poorly and have clips that identify the parts, then have shots and tracks recorded with the proper gain levels. It would be a very useful guide for those just getting started. It would likely require a version for different DAW programs though for best results.

 

I had a good example from some tracks I recorded with the band this weekend. I moved the drumset the previous week and I bumped the attenuator pot on the subwoofer and the signal was practically non existant.

It was at least 16db too low. I was able to add gain with Sonars gain adjustment but the RMS value sucked. It sounded thin in comparison to how it should have sounded. I did use Waves L2 limiter on it and it was tolerable, but its a perfect example of how not to do things. Analog gain does effect tone with most preamps.

 

I find if you can target your guitars tracks to come in at -6db, Bass about -4~5db and everything else in the -3db range, you can mix very well at those levels. You wind up using your sliders in the 50~60% range and not preaking the crap out of the meters. You also have enough headroom left to run mastering plugins without attenuating the mix to get them to function right.

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How about just educating these lumps of nescience, instead? I get so sick of "engineers" who can't think their way through the simplest logic problems. What on earth went wrong with the educational system that you end up with people who imagine themselves to be "engineers" yet who can't seem to figure out how to do simple troubleshooting?

 

 

What Mike mentions is what is called a "Poka Yoke" (idiot-proof) system.

There are many idiots -myself included-, so such systems are a need.

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How about just educating these lumps of nescience, instead?

 

If I had a dollar for every time I've posted the explanation, I could retire now.

 

Oh, wait a minute . . . I AM retired now.

 

There's so much stuff that's hidden from the user in these DAW things, why make something so worrisome so obvious? Why do you need to look at the waveform anyway? To locate the start of a phrase or to find an edit point, right? And for that it really helps to have it at maximum amplitude. You might still need to zoom in if you want to cut a chair creak out of a quiet portion of a recording, but most of the time if, at the end of the take, it just blew the maximum level up to full scale, you'd be comfortable with what you saw, knowing that you actually made a recording.

 

As an added benefit, since some people just think they need compression or limiting so they slap it on, looking at the difference between the full scale peaks and the eyeball average of the track will give you a clue. And that's something that's easier to explain and easier for a non-engineer to understand.

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The idea is not to help with levels, but rather, to remove the idea of analyzing level vs OdB info via the waveform.

 

Use the waveform for what it is good for. Navigating sound vs. time. Finding in and outs. Locating aural phenomenon with a visual aid.

 

They're not for getting levels. They're not meters. But by normalizing globally, you still retain the value of comparative level analysis track to track. Snare vs. Kick, etc.

 

I agree with Mike. Normalize the little peckers into something more girthy! And visible...

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Call me a nut, but I always thought that modern DAWs stored the audio in an IEEE 754-1984 single-precision 32-bit floating-point format. I always thought that modern plug-ins made use of the SSE instruction set, which is inherently floating point. Now if that is true then that would mean that the DAWs are very capable of processing hot signals that exceed 0dB, as well as wimpy signals that are way the {censored} down there.

 

If I understand this stuff correctly it would mean that clipping only becomes a problem when the signal exits the computer (i.e. when you play it through a sound card or when you try to save it in a 24-bit or 16-bit file). If I understand this stuff correctly then that would mean that when you normalize a track there are no real benefits, and that you only run the risk of losing information (specifically, the LSBs of the mantissa).

 

Is there anything I

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Call me a nut, but I always thought that modern DAWs stored the audio in an IEEE 754-1984 single-precision 32-bit floating-point format. I always thought that modern plug-ins made use of the SSE instruction set, which is inherently floating point. Now if that is true then that would mean that the DAWs are very capable of processing hot signals that exceed 0dB, as well as wimpy signals that are way the {censored} down there.

 

If I understand this stuff correctly it would mean that clipping only becomes a problem when the signal exits the computer (i.e. when you play it through a sound card or when you try to save it in a 24-bit or 16-bit file). If I understand this stuff correctly then that would mean that when you normalize a track there are no real benefits, and that you only run the risk of losing information (specifically, the LSBs of the mantissa).

 

Is there anything I

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Call me a nut, but I always thought that modern DAWs stored the audio in an IEEE 754-1984 single-precision 32-bit floating-point format. I always thought that modern plug-ins made use of the SSE instruction set, which is inherently floating point. Now if that is true then that would mean that the DAWs are very capable of processing hot signals that exceed 0dB, as well as wimpy signals that are way the {censored} down there.

 

If I understand this stuff correctly it would mean that clipping only becomes a problem when the signal exits the computer (i.e. when you play it through a sound card or when you try to save it in a 24-bit or 16-bit file). If I understand this stuff correctly then that would mean that when you normalize a track there are no real benefits, and that you only run the risk of losing information (specifically, the LSBs of the mantissa).

 

Is there anything I

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Call me a nut, but I always thought that modern DAWs stored the audio in an IEEE 754-1984 single-precision 32-bit floating-point format. I always thought that modern plug-ins made use of the SSE instruction set, which is inherently floating point. Now if that is true then that would mean that the DAWs are very capable of processing hot signals that exceed 0dB, as well as wimpy signals that are way the {censored} down there.

 

You're misunderstanding the problem that I think needs solving. The internal arithmetic isn't the issue. It's that the users want to push the level dangerously close to the clipping point of the A/D converter (which is real) in an attempt to getting a good looking waveform display for their recorded tracks.

 

If I understand this stuff correctly it would mean that clipping only becomes a problem when the signal exits the computer

 

That's the case with a properly designed DAW. In the early days, normalizing tracks that will be summed was considered a no-no because the internal arithmetic might not be able to handl the sum, but today most professional DAW programs, as well as modern hardware mixers, deal with this properly. But if the signal is clipped on the way in (which is the only place that you have real control) then it's clipped and will be a problem.

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A "Normalize graphic waveform display" function for DAWs? Then I could stop telling people not to worry that their tracks aren't hot enough when the waveform doesn't fill the entire track area (or to worry, if it does). We'd get a lot fewer worried new users and probably cleaner mixes if they set the record level correctly and didn't worry about the size of the squiggles.


I really think this is a great idea. Anyone else?

 

 

"ALT"> "arrow up" in Sonar.

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How about battery doors or lids that don't fall off and get lost?

 

I have a couple of devices that have hinged battery compartment doors, but the hinges feel so flimsy that I'm always afraid that I'll break the cover off if I'm not very careful. I've occasionally picked up a used piece that didn't have the battery compartment cover, so I guess they must go somewhere. Maybe they're with the coat hangers.

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