Automating effects levels in Logic 8

I'm trying to automatically start and fade an effect at the begining of a song. The plan is to have the effect at the begining of the song, then fade it a short time in. Is there an easy/simple way to do this?

Have a look at automation in the Logic manual. You can record the fade you want that way

Similar Messages

  • Volume levels in logic pro X keep jumping up to +5.2 and  +5.0 by itself?

    Volume levels in logic pro X keep jumping up to  5.2 and  5.0 by itself.There are no plugins nor automation activated on tracks? how can i stop the track levels from jumping?
    What do i do to avoid this?

    I and the same problem and neither of these fixes solved it. Defintely not a controller problem as guessed.
    The problem was solved by going to the step editor - similar to automation - but more hidden, and also taking place in the track at a point that I had trimmed out that heppened before the beginning of the composition.
    I solved the problem by going into the step editor of the track and deleting the bar in the volume track that kept setting it, and not showing up as an automation curve.
    I found this problem infuriating ,I really hope this solution helps.

  • Audio levels in logic

    Why is the recording level in Logic way lower then my preamp? If I send a signal into my analog preamp, it's way lower in logic and the wave for is way lower too.

    Hi Tegtours
    The Apogee interfaces have the ability to be calibrated to different operating levels. So it could be a case of re-calibrating or at least resetting to the "factory" conditions.
    The 1272 will probably have been racked by an independent tech, so that's a bit of a wild card in terms of how it's wired and Dan Alexander say's it' a line amp, I'll defer to him as he's a Neve expert.
    +WHAT IS A NEVE 1272?+
    +A 1272 is a line amp/summing buss amp from the circa 1970-76 era of neve consoles, such as 8048,8014,8016,and bcm10 portable consoles.+
    +A 1272 is not a mic pre, nor was it ever used by neve as such.+
    +However, when asked what he thought about people rewiring the 1272 as a mic pre, Ruperts' answer was" why not? It's all the same stuff in there".+
    +Inside of a 1272, one finds an input transformer, a line amp card, and an output transformer.+
    +Some enterprising techno gentlemen rewire these components and with the addition of appropriate volume and gain switches, build a preamp that emulates that found in a 1073( or other class a) input module.+
    +The only problem occurs when one attempts to get much over 50 db of gain, which is accomplished in a 1073 by the insertion of an additional amp card. When wringing that extra gain out of one amp card, one encounters frequency response and distortion anomalies. However, assuming that the wiring is done properly, a1272 can be made into a mic pre that exactly emulates the classic neve class a preamp 1073/1064/1066/1089/1084 type of circuit and of course , done right, that sounds wonderful....+
    http://www.danalexanderaudio.com/neverap.html
    If you've purchased a 1272 ( modified to be a mic amp ) from a reputable tech, I'm sure they'll be happy to show you how to interface it correctly.
    Which output are you using from the P2 and what kind of cabling ?
    James

  • Levels in Logic

    Thre has been a lot of discussion on whether or not it is importat to try adn keep channel levels from clipping (with Pre-fader metering) in a 32 bit float app like Logic as long as it is not clipping the output. I took the postition that it still is and some disagreed.
    The following is my exchange with Paul Frindle, who designed the Sony Oxford series of plug-ins and is an acknowledged expert in digital audio.
    i apologize for the length but I think it is important.
    PAUL WROTE:
    There are 2 reasons to record, process and master at less than flat out dBFS:
    1. To avoid hidden overs not shown on metering due to the reconstruction of the signal. This occurs mostly in D/As (variably depending on how they are designed) and to some extent within some plug-ins and digital processes. Error from this range from limiting all the way to loud 'splats' as values may fold over completely.
    A safe margin for normal programme to avoid most of this is -3dBFS. Although some artificially maximised programme can create more than this and some test material can create 6dB of over - so real safety is only gained at around -6dBFS.
    If you have a reconstruction meter you can monitor this effect yourself (for the mix output) and compensate manually, or if you have a suitably equipped limiting app (I.e. Oxford Limiter) you can correct these errors automatically. This does not help much with stuff within the channels of the mix itself - so prudence us still advisable.
    2. To create headroom, using lower levels within your mix allows you to avoid clipping signals every time you do anything - it frees you up to concentrate on sound rather than red lights and radically eases the mixing process. Some plugs may actually sound better because internal overs may be avoided.
    To do this you need to reduce levels to something sensible first thing in the playback channel - process at lower levels - end up with a mix at less than flat out - then make up the level at the very end of the mix. It sounds like you are doing this already But please note this is NOT to avoid math overs in the PT summing buss - as these are catered for already in the PT mixer
    Ok - you talk of making mixes that are suitably modified by the output limiter? Yes, this is common practice and in fact mixing with the limiter in place is a really good idea as you instinctively adjust the mix for the best final sound. But these days we have to watch it as the industry is obsessed with loudness at the expense of absolutely everything else - we produce 2 dimensional programme that has no dynamic range. Therefore it isn't possible for you to create real dynamics in this current environment (if you want to stay in business) - instead you are limited to trying to create the impression of dynamics from the extra artefacts and distortions the limiter generates.
    Is was with this in mind that I designed the Oxford Limiter - basically to create the impression of dynamic range when in fact there was none - and do it in a way that sounded as natural as possible. You can use this effect either to produce stuff that is loud as ever but sounds less artificial - or you can use it to produce stuff that is as bad as before but is even louder
    I hope this is helpful [/QUOTE]
    I THE ASKED: ul would you say this advice holds true for a 32 bit float app like Logic as well as fixed point apps like PT?
    PAUL RESPONDED:
    Yes I would.
    The intersample peaking reconstruction problem is the same at the output of the mix, as it must be represented in a fixed point output format anyway (i.e. CD or DVD).
    Whilst with float it's possible to accomodate internally numbers bigger than flat out, any process that has need to refer to actual real values might be at risk of overload (or unspecified behaviour). Why take the risk?
    From the point of the headroom issue, things might be different in that an entirely float system from start to finish might handle overs properly - however the meters will be calibrated to a fixed point reference (and will come on willy nilly, whether the signal is clipped or not). Some systems using expansion DSP pass and process signals in fixed point (PowerCore being one example) and may not have any of the float headroom and may mess up with overs.
    Again, with 140dB or so dynamic real range at your disposal, why bother to risk it?
    The most important thing to remember is that recording and processing at lower levels DOES NOT waste 'bits'. It doesn't work like that - all your 'bits' are there all the time at all levels

    Thanks for that Jay.
    Paul certainly knows his stuff - interestingly, I have been trying out the Oxford AU's of late, and reading the manuals - they are pretty dry, but an excellent source of technical material on this stuff.
    Everything he says in your post I agree with, but it doesn't really have anything to do with mixing with high channel levels. Let me take a few points:-
    To avoid hidden overs not shown on metering due to the
    reconstruction of the signal.
    This is absolutely true. In fact, it is possible to clip a D/A (ie go over 0dBFS) from sample values that are as low as 70% of 0dbFS. If people don't know about reconstructed waveforms, go and have a read of the Oxford limiter manual - you can download it from oxfordplugins.com - which explains this.
    However, this is from the master output onwards. As long as your master output is not clipping, including the reonstructed waveform, then this has nothing at all to do with mixing at high channel levels. So we can happily ignore this.
    Next:-
    2. To create headroom, using lower levels within your mix allows
    you to avoid clipping signals every time you do anything - it frees
    you up to concentrate on sound rather than red lights and radically
    eases the mixing process. Some plugs may actually sound better
    because internal overs may be avoided.
    A few points to make here. When tracking, it absolutely makes sense to track at lower levels and not try to reach 0dBFS, for exactly the reasons noted above. Obviously if you clip your recording at the A/D stage, your recording will always have distortion regardless of what happens to the audio from that point on - it's "burned in" to the recording.
    When mixing, the maths makes sure we are not clipping signals even if we add +500dB of gain. In effect, in a well-designed digital mixer, we always have a ton of headroom. So no need to worry about clipping channels in normal use - it simply doesn't happen. In short, in a 32f mixer, we are always mixing at fairly low levels - one of the benefits of this is so we can mix lots of signals together.
    Now, I'm not sure of the exact figures here, but someone with a more DSP bent willl doubtless chime in and state how many 24-bit 0dBFS signals you can mix together in a 32f mixer without clipping the entire signal (ie running out of maths headroom) but I seem to recall it's a lot. ie thousands - but again, don't quote me on that!
    Whilst with float it's possible to accomodate internally numbers
    bigger than flat out, any process that has need to refer to actual
    real values might be at risk of overload (or unspecified behaviour).
    Why take the risk?
    I'm not sure exactly what he's saying here, but I take it to mean "If a system is properly designed, high internal values are no problem. But if you've got a badly designed digital mixer and/or processing, you might get problems".
    Whilst that is true, it's also true of anything else - badly designed gear is badly designed gear. Now it could be that Logic's mixer is not designed well, or some plugins may not handle 32f values properly, and if so, then it is a very real cause for concern. but for the most part, I do not believe it to be true, although I cannot back that up with empirical evidence.
    However, some null tests would probably be able to verify whether the maths is working properly, both without processing and using various plugins or plugin chains.
    From the point of the headroom issue, things might be different in
    that an entirely float system from start to finish might handle overs
    properly - however the meters will be calibrated to a fixed point
    reference (and will come on willy nilly, whether the signal is clipped
    or not).
    I'm not sure what's he's saying here. Some plugins might get their metering wrong? That may be the case. I've always said that bad plugins can be a problem, but I don't believe there are that many out there. So let's leave aside the "crap plugins" issue and take that as read, and just concentrate of the straight 32f mixer as containined in Logic and other DAWs.
    Some systems using expansion DSP pass and process signals in fixed
    point (PowerCore being one example) and may not have any of the
    float headroom and may mess up with overs.
    Possibly - I think for those things, there are often some maths juggling going on to do with the DSP processors they use. I can't really speak on that, but again, it's not (as I understand it) the case in an ideally implemented 32f software mixer.
    The most important thing to remember is that recording and
    processing at lower levels DOES NOT waste 'bits'. It doesn't work
    like that - all your 'bits' are there all the time at all levels
    Also absolutely true.
    So as far as I can see - and do correct me if I'm understanding or interpreting things wrongly - but there isn't really anything here that suggests that mixing in a 32f environment at high-ish levels, then turning the master fader down to bring the sum combined mix down to below the fixed odBFS level for your convertor/file-format is bad.
    What he's basically saying if I'm understanding him correctly is that there is the potential for implementation errors in these systems, and those can makes things sound bad, and by mixing at very high levels you can possibly exxagerate those things to occur. That sounds plausible to me.
    But the maths alone quite happily supports mixing multiple signals at high levels - in fact, let's say +12dB over your individual channels 0dbFS point would be considered high - but with the maths, that's still a really small signal - that additional gain is tiny in relation to the overall 32f mixer's headroom.
    Now, if you are mixing channels together that are clipping the mixer internally (and again, I'm not sure of the values), but let's say you were adding +1000dB to every channel and adding 150 of these, then yes, I would fully expect your signal to be degraded, as the maths would not be able to retain the integrity of the full signal. But that's a wildly over-the-top example which would never ever happen in practice.
    So, my view still holds - given my interest in this, I see no reason that a well-designed system shouldn't be behaving properly, and the variety of tests performed by people infinitely smarter than me that I've seen over the past few years seem to bear out that things by and large are performing perfectly, at least in digital mixer's summing processing.
    I'm not saying I know for sure they are, or that I have real evidence at my fingertips to back it up, and as such I always remain open to investigation and findings. But I'm pretty sure people have added a bunch of signals and bounced the result, then added he same bunch of signals at +1000dB, reduced the master fader to compensate bring the mix down to the exact same levels as the previous test, bounced that, and then phase inverted and tested the files, to reveal they null out down to impossibly small values.
    As in all of these things, there are additional complications - when people say they hear a difference, then either they are mistaken or there is something going on - but I'm not (yet) convinced that that something is the native performance of a 32f mixer.
    Phew! Lots of typing, and I've probably forgotten things, but that'll do for this post anyway - my brain's hurting!
    PS Do go and read the Oxford manuals though - they contain some of the best technical info I've come across on DSP and signal processing. Great stuff! (And the plugs aren't to shabby, neither!
    Out.

  • Logical level for logical fact table sources

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    Hey everyone,
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    Not owning an Apogee Ensemble, maybe someone else who does has more definitive info can jump in, but do not confuse the lower level of the SPDIF input with "lesser quality". It's a digital connection, with no analog conversion.
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    Well, this was a great line of attack.  I always forget about trashing preference files.  But, I just spent the last couple of hours trashing preference files for Logic, for CueMix, MOTU and basically ALL the Apple Audio related preference files I can find.  Still the same issue.  I even delved deeper into CueMix than I thought I could to no avail.  I'm becoming more and more stumped as I go along here.
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    Ugh...
    Oh, and the Traveler firmware is current according to the MOTU website.  Granted the last firmware update was 3 years ago.  Also, upon closer inspection I realized that the CueMix software was not current it was 1.6.52xxx as opposed to the latest 1.6.55333.  (I guess I ignored the numbers after the 1.6 because they weren't followed by a dot)  So I installed that, trashed the preferences again.... and.... still the same issue!
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  • Setup help - low signal level in Logic

    I'm getting low input signal levels when recording in Logic (-17db to about -10db, on average), and am hoping someone here can point out an obvious oversight on my part.
    I'm recording mic'd guitars and vocals (one track at a time), with the mic(s)running into a Mackie 1402 mixer via XLR. Signals are strong and fine when I solo on the Mackie.
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    Hi,
    You wrote:
    I'm recording mic'd guitars and vocals (one track at
    a time), with the mic(s)running into a Mackie 1402
    mixer via XLR. Signals are strong and fine when I
    solo on the Mackie.
    The solo on the Mackie is probably louder,check to see if you have it AFL(after fader listen or PFL(pre fader level or "level set").
    The signal runs out of the Mackie via the Alt 3-4 bus
    into the first and second 1/4" inputs on the back of
    a Motu 828 mkII, then into Logic.
    The first thing to check is to see what kind of cables you are running into the Motu mkii.
    A. Unbalanced (ie "guitar-type" with two wires)
    B.Balanced (ie "stereo-type" with three wires)
    The second thing is to check what your inputs on your Motu are,there are two choices: +4dBu("pro") or -10dBV("consumer")
    Here's the catch:
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    The Mackie can output AUTOMATICALLY either -10dBV or +4dBu depending on what cable is hooked up to it.
    Even if I drive the signal to near clipping on the
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    Logic. This is true whether I'm using a condenser mic
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    I'm running Logic Pro 7.2.3 and OS X (10.4.9) on a G5
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    Is my signal path in need of some tweaking, or
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    828?
    Yes,a setting somewhere in the Motu is not set correctly.Check above for how to do that.
    Any help greatly appreciated to get this rookie back
    on track!
    G5 2.5 DP
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    Cheers

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  • Effects of Filter Logic on Performance/Load Time

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  • Convert effects from older Logic songs

    Hello,
    I've got a problem with my Logic songs created in Logic 4.8.
    If I open one of these songs all of the audio effects in the mixer
    are crossed out – also if similiar or equals are integrated but just
    have another name (for example: "SDly" in 4.8 is now in Express 7
    "Stereo Delay" with nearly the same interface).
    Have I got any chance, to convert these effects or read out the
    settings so that I can use them in the same way as before?
    Thank you for your help and excuse my maybe bad english...
    Regards – derkrebs
    G5 1.8 GHz Dual   Mac OS X (10.4.8)   Logic Express 7.23

    Thanks for all responses.. Don't think mine is a space problem, alot of these tracks have hardly any instruments on to open. I am using Logic Pro 6 if that makes a difference?

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