In article <***@g14g2000cwa.googlegroups.com>,
***@d-and-d.com says...
>
>
>Chevdo wrote:
>
>> I listen to what I'm doing for reasons other than to prevent clipping - I
>> simply analyze the file after it is recorded to see if there was a clip.
The
>> meters aren't particularly reliable.
>
>And if there's a clip? Do you leave it for the sake of loudness? Fix
>it? Listen to it and make a decision?
Depends on how good the take was, if the person playing the instrument says
it's "THE" take, we'll manually remove the clip. If the person playing the
instrument says that take wasn't that great, I wouldn't mind trying it again
anyway, I'll lower the level a bit and try again. If the person says hey all
these takes are costing me too much money, I'll tell them I could give them
plenty of headroom so that clipping never happens, but I will also tell them
that the quality of their recording will suffer as a result. Often when I tell
them that it may be negligable they appear skeptical since they know I believe
it's important to get hot levels. Ultimately it's up to whoever is paying to
decide how to record. When I'm working on my own stuff I don't compromise, I
always re-take clipped tracks but my stuff is mostly sequenced and the
instruments I do play are simple parts I can easily reproduce, because I'm not
much of an instrumentalist so I don't try to do anything difficult.
>
>> I'll take as many bits as I can get in the DAW I mixdown with. As for
>> samples, I use 12bit akai s612s, 8bit casio sk1, and various sample rates,
>> not to mention romplers of varying sample resolution. When I record them
>> into the DAW I set their output volume levels at just under maximum
>
>Not saying this as a put-down, but with gear like that, you're
>obviously into a form of music that doesn't have any dynamic range to
>begin with.
No, that's completely false. If I record a track from an s612 with a hot
level, I may place it very low in the mix, so suddenly the mix has greater
dynamic range than the s612, because I am mixing tracks recorded at 24bit. The
final mix will demonstrate the best my 24bit/44.1khz board and my own ability
can produce. I find it hard to believe that you don't understand this.
> It's easy to set levels accurately when you're using synths
>and samplers because they play at the same level every time, and you're
>probably sequencing with uniform volume and velocity
No, I'm not. That would make for a very uninteresting mix, in my opinion.
>- essentially it's
>compressed and limited when going in, so there's no reason to expect
>unexpected peaks. You already know where (and at what level) the peaks
>are.
not really, I don't work that way. The tweaks on the s612, for example, can't
be recorded into a midi sequencer, so I have to tweak in realtime during
recording, so I have no idea where the levels are going to be until I've
recorded a track.
>If your DAW can properly mix a bunch of tracks at full level,
>there's no reason not to do so. But other that volume, which you can
>fix by turning up your monitor level, there's no reason to develop this
>bad habit. Some day you might find yourself recording a jazz or country
>band, or a classical concert. But I won't hold my breath.
The only bad habit is your bad habit of fantasizing about my habits.
>
>> Slam then back off a tiny bit on outputs. Inputs as hot as possible. You
>> can waste your bits if you want to, I'm not going to. All converters are
>> MOST efficient at the top end.
>
>"Efficiency" is a term that I've never heard used to describe the
>performance of an A/D converter.
Wow, you're learning something then, I guess?? Efficiency in this context
refers to how often the converter converts a bit accurately. They're always
most accurate at the top. I have a digital scale that, like all digital
scales, is more accurate when you put a load on it first then 'tare' it to zero
, before adding what you want to weigh. It's supposed to be accurate to .05 of
a gram, but if you try to weigh something that weighs under a gram it won't
even register. This analogy isn't exactly perfect, and possibly irrelevent,
but regardless, converters are designed to be most accurate at their maximum.
>I suspect that you like the grit
>that's added by your inaccurate converters because it adds emphasis to
>the music you're recording.
I suspect you are ignorant about how converters work. I'm curious as to who
started this myth and how it perpetuated, though. Is it relegated to this
group? Because I've never heard it before.
>> No that's not the only thing you do, you also raise the level of the signal
>> in proportion to the noisefloor. You're going to push that noise floor way
>> up in the end with L3 or something similar in the end anyway.
>
>Not me. Like I said, what kind of shit are you using? Any modern
>converter can stand to have its quiescent (with a shorted input) noise
>level boosted by 20 dB without getting the noise level up to the point
>where it's audible. If you are finding that your system doesn't allow
>you to do this (even if you don't want to) you really should consider
>upgrading. But I'm wondering if you actualy are having a noise problem
>when you record at conservative levels, or if you're just working on
>principle without listening to alternatives.
I have no noise problem, you have a comprehension problem. But that's ok
because it's not my problem.
>Fair enough. I've forgotten why we got to this discussion, so I won't
>comment on why your preferred system might or might not be best. In
>theory it's not the best way to go,
In a better theory, it is. Bit resolution does exist and even a 12bit sampler
with a high 'noise floor' is still producing a signal below that noise floor.
It's not a cut-off, it's just the noise of the circuitry. There's still signal
below it.
> but in this business, we often do
>things that would be considered mistakes for the sake of getting a
>different (and hopefully effective or attractive) sound. Why do you
>think people crank their guitar amplifiers to distortion, or why was
>the Fuzztone invented? Clearly not to get cleaner sound, but it works
>for a lot of people.
>
works for me, but when I record into my DAW I am going for the highest quality
I can get and filling up the bits is the way I do that. That's the way it was
always done when only 16bits were available, and that's the way it should be
done when 24bits are available. The availablility of 24bits simply reduced the
importance, but only because we are still mixing down to a 16bit CD product.
What are you going to do when we're mixing to 24bit in a new consumer format?
Stick with your low levels and revel in your 'headroom'. You were talking
about developing bad habits earlier...
>We don't compete in the same part of the industry. I guess I should
>feel thankful for small favors. ;)
>
Even jazz and folk bands want a loud recording these days. Your only
sanctuary, for now, is classical. And tests have shown it's gotten louder and
more compressed over the years.