Hardware vs. software reverb and effects

Discussion in 'rec.audio.pro' started by David Nobel, Aug 28, 2003.

  1. EggHd

    EggHd Guest

    << It's easy for a
    beginner to produce a bad mix. Many of the tones and instruments that
    beginning musicians play with are not very mix friendly either. >>

    Flectcher "summed" it up nicley in another thread.

    You can go out and buy a stratocaster but it isn't going to make you play like
    Eric Clapton (paraphrase)



    ---------------------------------------
    "I know enough to know I don't know enough"
  2. Jeff Chestek

    Jeff Chestek Guest

    Scott Dorsey wrote:
    > Ryan Mitchley <rmitchle@removethis.worldonline.co.za> wrote:
    >
    >>Unfortunately, most of the reverb plugins suck. Badly. And I've tried quite
    >>a few of them. Most of them seem to suffer from a ringy, comb-filtering kind
    >>of sound

    >
    >
    > What do you want from a PC? You just don't get that many taps on a reverb
    > if you want to do it realtime, unless you have dedicated hardware to do it.
    >
    > I'd still settle for a hardware unit that sounded like a real room, let
    > alone a plugin.
    > --scott
    >


    Have you tried the Sony S-777 sampling reverb? Comes pretty close to a
    speaker and a couple of mics in a real room.

    Jeff
  3. DN> Should have made myself clearer here. My point about staying at the 96
    DN> kHz sample rate after the round trip conversion to the analog reverb
    DN> was contingent on proceeding to do other processing--EQ, compression,
    DN> stereo enhancement, etc.--in the digital environment. As mentioned,
    DN> theoretically, the finished CD should benefit sonically from having
    DN> done this digital work at the 96 kHz vs. 44.1 or 48 kHz sample
    DN> rates--even if the original recording was done at the lower rates.

    I think the above posters knew what you were saying. The point is that 96kHz
    gives you better bandwidth, i.e. reproduction of frequencies debatably
    considered ultrasonic. The 96kHz sampling rate won't do anything for
    frequencies < 22kHz *AT ALL* (although Bob was putting forward a rather
    tenuous argument how this *might* occur). Working at a higher sample rate
    won't buy you higher resolution for processing. Increasing the bit depth
    *will*. Whether this is necessary depends on the noisefloors of your
    equipment and the gains applied at each stage.

    Ryan
  4. Ethan Winer

    Ethan Winer Guest

    Monte,

    > Truncation and low resolution math. <


    Okay, maybe. I never owned ProTools nor would I want to. I started with SAW
    and now use Sonar. Both mix internally at high resolution and it's never
    been a problem for me. Whenever I am not satisfied with a mix - which is
    pretty often - I know for a fact that it's me and not the tools.

    > One other theory I have as to why some people like analog summing is that

    most folks' DACs are pretty poor quality, and can't handle a complex mix all
    that well. <

    I don't know why that would be the case. The real issue is handling fast
    changes in level, which happens with fast transients and complex mixes, but
    also with a single snare or tambourine hit. Of the modern (last few years)
    semi-pro and pro sound cards I've played with, all are pretty darn
    transparent.

    > It's easy for a beginner to produce a bad mix. <


    This is the very crux of it. A newbie makes a mix that sucks and doesn't
    understand why. The gear stores and magazine writers are happy to tell them
    to buy yet more gear, that the problem is surely their mike pre / sound card
    / word clock / you name it. And snobby pros jump in too saying they'll never
    get good results with their toy system.

    > kids love to use distortion boxes that sound great soloed but absolutely

    wreck a mix <

    Yes, and that also aligns with my theory which pertains to all instruments
    and tones. Often things that sound great alone don't blend well because of
    masking.

    --Ethan
  5. Ethan Winer

    Ethan Winer Guest

    Monte,

    > if you use an outboard processor with digital IO and can save the settings

    to a preset (or do a MIDI sysex dump), you can use outboard hardware and
    also have complete resettability. <

    Yes, but I'm too lazy for even that. :->)

    Aside from reverb, I have no complaints about the quality of the plug-ins I
    use (UltraFunk). And their reverb is still better than most other plug-ins.

    --Ethan
  6. In article <-42dndgN6ucT0c6iXTWJkQ@giganews.com>,
    Ethan Winer <ethan at ethanwiner dot com> wrote:
    >Monte,
    >> One other theory I have as to why some people like analog summing is that

    >most folks' DACs are pretty poor quality, and can't handle a complex mix all
    >that well. <
    >
    >I don't know why that would be the case. The real issue is handling fast
    >changes in level, which happens with fast transients and complex mixes, but
    >also with a single snare or tambourine hit. Of the modern (last few years)
    >semi-pro and pro sound cards I've played with, all are pretty darn
    >transparent.


    That single tambourine hit might get slightly mangled by a bad DAC,
    but it couldn't cross modulate the rest of the mix if it's being
    played out solo, as in an analog mix environment. I still find that
    even some fairly pricey modern DACs aren't all that good with complex
    mixes, for example the Apogees. Maybe some soundcards are better, but
    maybe not.


    Regards,

    Monte McGuire
    mcguire@theworld.com
  7. Ethan Winer

    Ethan Winer Guest

    Monte,

    > That single tambourine hit ... cross modulate the rest of the mix <


    Okay, good point, though a tambourine does have many components and so can
    create IM products. But to keep things in perspective, what kind of IM
    distortion figures do you have in mind? To my ears, anything 60 or probably
    even 40 dB below the program, and masked by the program, is basically
    inaudible. I think all/most modern digital gear is way better than that, but
    I admit I only really know the gear I personally own.

    And to REALLY keep things in perspective, why are miniscule amounts of
    degradation considered so damaging in a digital system, when IM and noise
    etc. 100 times worse on an analog tape deck are so cool and desirable? :->)

    --Ethan
  8. mrivers@d-and-d.com (Mike Rivers) wrote in message

    > The flaw with that procedure is that you have to decide at the time
    > that you're recording the vocal just what settings to use on the
    > reverb unit...........


    Yep, that's the 'rub'. And I agree with you that it would be better to
    "print" the effects close to mixdown time ......Plus if you print lots
    of effects along the way, the files become huge.

    But the title of this thread, "Hardware vs. software reverb and
    effects", presents the problem as an either-or situation. Why not use
    both, and this is a simple effective solution.

    With computer recording, people tend to see a closed system and keep
    their heads in the software sand, but don't we need to think outside
    of the box a bit? We should learn to commonly intergrate some of our
    favoite hardware with our computer recording systems, instead of
    replacing then. Can you really software simulate the sound of tube
    gear or other classic sounds?........ I wish a few soundcard and
    hardware manufacturers would consider adding more input/output access
    such as inserts, busses, etc.... in a few of their products.

    Maybe I'm old-fashioned, but for me, it's worked out well to 'print'
    some of my favorite reverbs and processors to tape..... ah, I mean
    disc.

    J_West
  9. "Ethan Winer" <ethan at ethanwiner dot com> wrote in message
    news:c_idnc2j2ZcIOs2iU-KYuA@giganews.com...
    > Ricky and all,
    >
    > > On the best digital systems (which I have not used) it might be OK but

    at
    > my level of equipment it sounds too 2 dimensional if done completely in

    the
    > DAW. <
    >
    > Of course I've heard these comments, but I just don't buy them because
    > logically they make no sense to me. Depth and space are ambience and
    > left/right timing issues. How can adding a bunch of numbers digitally

    affect
    > image width and depth? And why wouldn't the same thing happen with analog
    > summing? I guess my real question is, "What TECHNICALLY AND
    > SPECIFICALLY occurs in the digital realm that could cause such a

    phenomenon?
    >
    > Here are my two theories as to why some people prefer analog tape and why
    > some people blame mix busses:
    >
    > People that prefer analog tape refer to its natural compression, gentle
    > suppression of transients, a small but desirable increase in distortion,

    and
    > a noticeably wider image than digital recording offers.


    Yes. This is definitely the reason I like analog. The more I thought about
    it, two dimensional wasn't really the right word but I just prefer the way
    analog makes it "gel" and sound "whole" instead of "flat" . Of course, if my
    chops were better I might could do that in digital but for now the analog
    tape does a nice job.
  10. Ethan Winer wrote:

    > > I'm not sure "staying completely inside the DAW" makes for quality mixes.

    >
    > I've never experienced quality problems mixing entirely in a DAW, and I know
    > many others who also use DAWs successfully. What kind of quality degradation
    > do you mean, and what specifically / technically would cause it?


    I honestly have no idea what causes it, but I have come up with a mixer
    design that maintains total DAW control over the mix and eliminates
    essentially all possible analog electronics variables except one. That
    one remaining variable is rather elusive, but we've been able to
    conduct rather controlled tests that suggest the "DAW summing"
    phenomenon really does exist, even if I can't figure out the cause of
    it. I've sold a few of these mixers, and the users are figuring out
    more and more all the time how an outboard mix differs from an internal
    mix, even with no faders or mixer electronics to skew the results.

    ulysses
  11. David Nobel <dnobel@sympatico.ca> wrote:

    > > Why do you want to run 96 ksamp/sec? I'd pick a 44.1 system that sounded
    > > good over a 96 system that sounded bad, any day.
    > > --scott

    >
    > That's a good point. However, I have good-sounding 96 k and happen to
    > be in the camp that believes that working with acoustic music at
    > higher sample rates before burning can improve the quality of the
    > audio CD. It irks me to contemplate not being able to use that
    > capability.
    >
    > The chamber is not practical for me now, but something else which
    > occurred to me is running a round trip analog conversion solely to
    > post process to an older, affordable, classic analog hardware reverb
    > unit. Some folks believe in running digital tracks out to analog
    > boards for mixing, so why not for reverb? This would of course require
    > good ADA converters. So is it a stupid idea?


    It's a fine idea. It's just reverb. It's completely reasonable to
    keep your tracks at 96k all the way through, and either downsample your
    reverb send to fit it into a digital box, or simply convert to analog
    and send it out that way. Whichever makes you feel better about the
    theoretical fidelity. There's absolutely nothing going on in the
    reverb that requires the higher sampling rate, so don't worry about it.
    But if you're satisfied with the quality of your 96k A/D and D/A
    converters for use on your dry tracks, then they'll be perfectly
    adequate for connecting to and from whatever reverb you want to use.

    ulysses
  12. David Nobel <dnobel@sympatico.ca> wrote:

    > > Why do you want to run 96 ksamp/sec? I'd pick a 44.1 system that sounded
    > > good over a 96 system that sounded bad, any day.
    > > --scott

    >
    > That's a good point. However, I have good-sounding 96 k and happen to
    > be in the camp that believes that working with acoustic music at
    > higher sample rates before burning can improve the quality of the
    > audio CD. It irks me to contemplate not being able to use that
    > capability.
    >
    > The chamber is not practical for me now, but something else which
    > occurred to me is running a round trip analog conversion solely to
    > post process to an older, affordable, classic analog hardware reverb
    > unit. Some folks believe in running digital tracks out to analog
    > boards for mixing, so why not for reverb? This would of course require
    > good ADA converters. So is it a stupid idea?


    It's a fine idea. It's just reverb. It's completely reasonable to
    keep your tracks at 96k all the way through, and either downsample your
    reverb send to fit it into a digital box, or simply convert to analog
    and send it out that way. Whichever makes you feel better about the
    theoretical fidelity. There's absolutely nothing going on in the
    reverb that requires the higher sampling rate, so don't worry about it.
    But if you're satisfied with the quality of your 96k A/D and D/A
    converters for use on your dry tracks, then they'll be perfectly
    adequate for connecting to and from whatever reverb you want to use.

    ulysses
  13. Arny Krueger

    Arny Krueger Guest

    "Ethan Winer" <ethan at ethanwiner dot com> wrote in message
    news:uuOcnU4AZOU4F86iXTWJiQ@giganews.com

    >> That single tambourine hit ... cross modulate the rest of the mix <


    > Okay, good point, though a tambourine does have many components and
    > so can create IM products. But to keep things in perspective, what
    > kind of IM distortion figures do you have in mind? To my ears,
    > anything 60 or probably even 40 dB below the program, and masked by
    > the program, is basically inaudible. I think all/most modern digital
    > gear is way better than that, but I admit I only really know the gear
    > I personally own.


    The 60 dB number is significant. I've found that downward conversion via IM
    is the nonlinear distortion mode that the ear is most sensitive to, and -60
    dB is the range of the audible limit under ideal conditions. Ideal
    conditions are fairly rare.

    Tambourines aren't the worst case sounds to use to listen for distortion
    because they create a fair amount of output at lower frequencies that can
    mask a good chunk of any distortion products that are created.

    The biggest roadblock to hearing nonlinear distortion products is IME
    self-masking.

    IOW, the musical instrument or ensemble produces sounds in the range where
    the most likely distortion products tend to land. If the distortion products
    are correlated with the masking sound, then you are trying to hear a change
    in timbre. If they are not correlated, then you have a traditional
    concurrent masking situation.

    Either way, the ear can't resolve really small differences. IME when
    concurrent masking is involved hearing thresholds fall into the -30 dB
    range, if that.

    If you want to hear really small amounts of distortion, you need dead
    silence in the range where the distortion products tend to fall.

    The worst case (quasi) natural sound that I've found for detecting high
    frequency IM is keys jangling, particularly if you aggressively high pass
    them at 4 KHz. This will get you to -60 dB, but only as a solo instrument.

    IMO listening to recorded music from the 20th century tells me that
    consciously or unconsciously people were arranging music to incidentally
    mask the kinds and levels of distortion that were common at the time the
    recording was made. By the end of the 20th century, we had recording and
    playback equipment that was so good that just about anything could in some
    sense work.

    > And to REALLY keep things in perspective, why are miniscule amounts of
    > degradation considered so damaging in a digital system, when IM and
    > noise etc. 100 times worse on an analog tape deck are so cool and
    > desirable? :->)


    Nonlinear distortion created by tape sounds cool partially because the
    threshold is so gentle. There seem to be some overall compression effects,
    as well. But, if you hit tape with sounds that need serious amounts of
    clarity, forget it. Analog tape can't be sonically transparent, but digital
    done right can.

    This puts us looking squarely at mics, speakers, positioning them ideally,
    and rooms at the start of the 21st century.
  14. David Nobel

    David Nobel Guest

    "Ryan Mitchley" <rmitchle@removethis.worldonline.co.za> wrote in message news:<3f5303e1$0$3358$a32e20b9@news.nntpservers.com>...
    DN> theoretically, the finished CD should benefit sonically from
    having
    > DN> done this digital work at the 96 kHz vs. 44.1 or 48 kHz sample
    > DN> rates--even if the original recording was done at the lower rates.
    >
    >The 96kHz sampling rate won't do anything for
    > frequencies < 22kHz *AT ALL* (although Bob was putting forward a rather
    > tenuous argument how this *might* occur).


    You may find this a "tenuous argument," but it certainly goes a long
    way toward explaining why music can sound better when non-linear
    processing occurs at higher sample rates, even when originally tracked
    at 44.1 and 48 kHz. Some highly respected mastering engineers claim
    that processing at the higher rates can reduce distortion in the
    audible band by as much as 3 db, through the reduction of artifacts.

    >Working at a higher sample rate
    > won't buy you higher resolution for processing. Increasing the bit depth
    > *will*. Whether this is necessary depends on the noisefloors of your
    > equipment and the gains applied at each stage.


    Agreed.


    -- David
  15. Bob Cain

    Bob Cain Guest

    David Nobel wrote:
    >
    > "Ryan Mitchley" <rmitchle@removethis.worldonline.co.za> wrote in message news:<3f5303e1$0$3358$a32e20b9@news.nntpservers.com>...
    > DN> theoretically, the finished CD should benefit sonically from
    > having
    > > DN> done this digital work at the 96 kHz vs. 44.1 or 48 kHz sample
    > > DN> rates--even if the original recording was done at the lower rates.
    > >
    > >The 96kHz sampling rate won't do anything for
    > > frequencies < 22kHz *AT ALL* (although Bob was putting forward a rather
    > > tenuous argument how this *might* occur).

    >
    > You may find this a "tenuous argument," but it certainly goes a long
    > way toward explaining why music can sound better when non-linear
    > processing occurs at higher sample rates, even when originally tracked
    > at 44.1 and 48 kHz. Some highly respected mastering engineers claim
    > that processing at the higher rates can reduce distortion in the
    > audible band by as much as 3 db, through the reduction of artifacts.


    It would be easy enough to test the tenuousness of the
    hypothesis with a program containing a spectral analyzer
    like Audition (CEP.) Just run the process or processes on
    something resampled upwards from 48 to 96 and see what
    appears in the 24 to 48 kHz region after applying them.


    Bob
    --

    "Things should be described as simply as possible, but no
    simpler."

    A. Einstein

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