Ambient mic always causes phase issues with close mic on guitar cab

Discussion in 'Guitars' started by rectifryer, Jan 16, 2014.

  1. Boswell

    Boswell Moderator Distinguished Member

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    Some manufacturers are very lax about phase preservation through their gear. This can range from stupidity in simple audio interfaces (microphone and line inputs in-phase but DI input appears inverted at the output) to large-format mixers that invert if you use the insert loop. Inversion can usually be cured in a balanced system simply by specifying the output XLR connector pins 2 and 3 to be laid out the other way round.

    In the contract design work I do, I make sure that phase preservation is specifically stated in the contract even if the equipment company had not thought to put it in there. One of the other equipment design consultants I know told me that one of his careful designs was butchered by the production department of the particular company he was working for in order to save an op-amp per channel. The result was not only that the sonic character of the device changed audibly over the range of gains, but it produced an overall phase inversion. He protested both about the sonics and the inversion, but they didn't correct it because they didn't understand the need for phase preservation. Their justification was that the frequency response amplitude curves didn't show a problem, so they left it out of phase.
     
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  2. RemyRAD

    RemyRAD Member

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    Oh, I've absolutely noticed that some plug-ins, do in fact flip phase. I'm sure this has made for other problems, for others. I think we've heard it here? Especially when, all the kids go wacky, with plug-ins and come up with these god-awful sounding mixes. And where we know, our stuff doesn't sound like that because we aren't playing with too many plug-ins. At least I'm not. I mean plug-ins are great. When you really know what they do and how they do it.

    I have an old plug-in that I've been using for probably 10 years now by VERSONIX, the BBE Sonic Maximizer. It most definitely inverts phase. But then it also does this other gobbledygook, even when the controls are set to zero. But the waveform is most predominantly, inverted. And this can really affect how fat and punchy your drums will sound.

    Right now I'm just basing this on a pair of FOSTEX 6301, self powered monitors. Whose polarity is fixed and non-adjustable. So then you're at the mercy of the digital to analog converter. And that it's decoding the polarity correctly. And then will it be at the polarity they designed it to be or, the polarity that we want it to be? Yet I find this to be a huge factor, in monitoring, in control rooms and the way, the recordings are presented to others. I don't believe this is a subject that even comes up in recorded schools? Because some of this flies in the face of theory. And no one can answer the questions.

    It's all between the ears
    Mx. Remy Ann David
     
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  3. Kurt Foster

    Kurt Foster Distinguished Member

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    i think a lot of people these days don't think polarity is such a big deal anymore. in the days when records were cut phase was an issiue ... tv was mono ... am radio was mono and if there was too much bass the needle would pop out of the groove or you would have to widen the groove shortening the time per side.

    mp3 16 44.1 whatever now days doesn't care about too much bass or mono compatibility. it simply doesn't matter ... even most am radio playing music is in a stereo format. mono am is reserved for talk radio format. out of phase? doesn't matter.
     
  4. audiokid

    audiokid Chris Staff

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    Do we hear out of phase in the natural world?

    I still mix in mono and check everything regardless.
     
  5. paulears

    paulears Well-Known Member

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    I mentioned all this to a friend of mine who has perfect pitch, and he told me about his piano tuning friend who has perfect pitch - but even more so. In piano tuning circles there is a difference between B Sharp and C, which is just a few Hz, but he can hear that, and he also complains that some pianos don't sound 'right' - he wondered if this is the polarity reversal of a piano mic?
     
  6. Kurt Foster

    Kurt Foster Distinguished Member

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    it's called Doppler effect .... like a siren passing by ,,, higher note on approach, lower note as it passes.

    this is also how a Leslie speaker works ... phase shift.
     
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  7. audiokid

    audiokid Chris Staff

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    When you record it, how does it translate in comparison to emulating the effect? I've never recorded a siren passing me by.
    I get the whipping of the Leslie and the siren passing by but we still sum it between our ears better than the outcome you get shifting audio through the extreme measures some do to create the wide effect.
    Phase has always been a love hate thing to me.
    Thanks for explaining the the Doppler Effect.
     
  8. DonnyThompson

    DonnyThompson Distinguished Member

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    I think that the plethora of freeware and shareware plugs available these days is perhaps one of the major culprits involved in phase problems and overall smearing that I'm personally hearing more and more of.

    The best plugs I've ever used were either designed and coded with a huge amount of quality control, or, are the plugs that come available as "stock" in whichever DAW platform you are using.

    For example, I LOVE Fabfilter's EQ. It's the best VST EQ I've ever used... it sounds fantastic.... but it ain't cheap. It's obvious that the developers took the time to test it on a variety of DAW platforms and it shows in the final product.

    But, DAW users, for the most part, aren't gonna want to pay $300 for an EQ plug, when there are so many available for far less, or even free. The general - and mistaken consensus - is that "all EQ's sound the same, they all do the same thing"... but that's just not always true... in fact, it hardly ever is true. And that can also be said for cheap/free compressors, limiters, delays, reverbs, or any other type of processor or effect.

    When you start crowding your mixes with multiple uses of cheap or free plugs - like Antress plugs, for example - or other freeware VST's - the sound will suffer, because the coding on these plugs is so quick and dirty... and that's why they are free.

    They may have been tested to make sure they work on your platform - meaning that they open, they work, all parameters are functioning - but they sure haven't been tested as to how they sound when they are doing whatever it is they are supposed to be doing...and, they haven't been tested in scenarios where multiple uses of the same processor or effect are being used at the same time.

    On a recent mix I did, which I will soon post, I ended up using an old Lexicon Prime Time for delay on backing vocals. Why? Because it sounded better than any stock delay plug I had for that particular use and particular song.... and, to take it further, I didn't use it as a "send and return", either. I sent the backing vox out of one PC, then through the Lexi, and then printed that processed track to another DAW on another PC.
    I then saved that part to an external HD and imported the finished track back into the timeline of the original project. The result was a much nicer treatment, a better sound, less smearing and phase, than if I had simply inserted a Delay VST internally to the vocal track's FX bin.

    FWIW

    d/
     
  9. kmetal

    kmetal Kyle P. Gushue Distinguished Member

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    Love the mono button, use it all the time, great for balances, ever since I started using a console w one, I haven't stopped using turning it on and off.

    As far as plug-insgo, I've been wondering if certain formats, vst, au, aax, ect, exhibit less artifacts than others, when the same pluggin is used. Also, how the coding of the actual DAW plays into it, when using the same format and pluggin among different daws.

    I've recently encountered something concerning polarity, when as we are getting a trident 24 going. One of the interns said that it was wired out w reverse polarity, than how our panels and cables are. So his solution was to use the phase fli (polarity reverse) on every channel as a starting point?

    I have had to deal w flooding so I haven't been able to re visit this concept, but I wonder if this is true about the console, and if so what would be the best, most sonically integral way to deal w this?
     
  10. dvdhawk

    dvdhawk Well-Known Member

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    If I remember correctly, there were a handful of hardware manufacturers (and I think certain era Tridents were among them) who decided pin #3 should be wired hot, pin #2 inverted. Mostly British companies, again if memory servers.
     
  11. Boswell

    Boswell Moderator Distinguished Member

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    Even with the old British desks that used pin 3 hot on the XLRs, they were consistent (mostly), meaning that there was no apparent phase inversion going from an XLR mic or line input through to an XLR output. The problem came when the signal appeared on TRS jacks, where what you expected to be the +ve signal was on the ring contact. Some people used to say it was the jacks that were wired in a non-standard way, but a glance at the schematics showed that the designers chose pin 3 hot at the XLRs and used tip hot at the jacks.

    I don't know where the Trident 24 fits in this scheme of things. It would surprise me if the XLRs were wired with pin 3 hot, but one does encounter surprises in the audio field.
     
  12. Reverend Lucas

    Reverend Lucas Active Member

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    Interesting discussion, per usual. I think the slight difference between B sharp and the enharmonic C is a result of pianos using equal rather than just temperament. A boon to those with golden ears, but probably worth it to the rest of us who would like to play in more than one key.
     
  13. pcrecord

    pcrecord Quality recording seeker ! Distinguished Member

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    So true !! For years I've used voxengo instead of sonar's EQ and when I started to use the last version of Sonar x3, I took time to try the new designed EQ with spectrum analyser. Sonar's was good looking but not sounding that good. I went for Fabfilter's recently and it is even better than Voxengo ;)
     
  14. RemyRAD

    RemyRAD Member

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    Reverse phase absolutely happens in the natural world. This can be heard in all sorts of environments. It's the timing. It's all in the timing. Which is why you move microphones around if you hear a loss of low-end and/or comb filtering. It's all from the timing.

    Some of this timing can now be manipulated within audio software. As has been mentioned by aligning peaks. To pull things into a consistent and cohesive related phase/timing. Software has been a boon to be able to do this.

    I have done thousands of multiple announcer commercials were three guys are sitting at a table mere inches apart. And the microphones are in cardioid. And you hear huge phase timing problems with grotesque comb filtering if I wasn't downward expanding/ducking/gating, each separate announcer. And where then each announcer sounds bigger and more intimate. And it's still actually works when they are playing off of each other, with the threshold set, oh so carefully, on the downward expanders/gates. All of which are used to do with separate hardware devices. Today, I can accomplish the same thing in software, drawing my own downward expanders, from compression/limiting, built-in audio software drop downs.

    Doing this with software compressor/limiters, to create the downward expansion and/or noise gating, can also combine compression along with peak limiting. All from the single compressor drop-down one finds resident in most audio software's. Basically creating a highly tailored vocal processor. That can also be manipulated, to also include, a DE-ESSER. Along with any kind of frequency weighting you want on your RMS or peak sensing, processor, preset. Which can let the low frequencies and/or the high frequencies come through relatively unaffected. Whereas the mid-band, at the vocal frequencies, are processed. Of which all is available from early shareware programs like Cool Edit 96. But you can only accomplish that when you have the graphic display of the dynamic processing. And not something that looks like a cartoon version of an 1176 or LA-2. You need the graph that displays level versus ratio. The same graph you'd find in those 1176 manuals.

    Of course not all can be done from the single compressor/limiter, for specialized correction issues. Sometimes that requires a tailored effect on top of a different tailored effect of dynamics processing. Where you want different time constants of attack and release of the different processing aspects, individualized. Where most of the time you can get what you want from the single pass processing. Other times you might need 3 layers of slightly differently tweaked processing. That don't all share the same attack, release, ratios, threshold values necessary to do the job. For instance when you have some gigantic background ambient noise to deal with. And you just want to accentuate the dialogue and not the broadband background ambient noise. While at the same time, you can't completely gate that out without it sounding totally unnatural. But you can't duck it, dynamically.

    All of these methods may have been employed by the engineers of that particular program. And it may have been done on an Avid Icon, through ProTools, with every available third-party plug-in, available on the planet? So they could have even taken it further than what I generally do? And because it was also produced likely not in Billy Bob's basement studio on a Beringer? Where Hollywood has no real budgetary limits.

    Sometimes, even weird devices are used on dialogue. Such as the Dolby 361, A noise reduction units with the CAT-22, single ended processor controller. Sometimes even broadcast processors have been employed by CBS, Orban, CRL, Gates. All of which may have also been in-house modified? When you need certain things to really cut through that wouldn't ordinarily. And the good old APHEX exciters. The real ones. Not the fake ones we all have.

    I'm a fake but my audio and facility are not.
    Mx. Remy Ann David
     
  15. Josh Conley

    Josh Conley Active Member

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    another reason may be the hurting higher end eq's can put on your cpu.
    i remember when UA first put out the 1073/1084 eq's -we were still on pci xards ;) - and one 1073 instance took up an entire card.

    in comes the 1073 SE, same plugin, less cpu drain, but what is being sacrificed?
     
  16. Kurt Foster

    Kurt Foster Distinguished Member

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    i don't think Phil used gates. bleed can be your friend.
     
  17. dvdhawk

    dvdhawk Well-Known Member

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    I'm pretty sure to this day Klark-Teknik still ships as pin #3 = hot, at least on the DN series EQs like I have. [ manual in ZIP form ] Their budget-line purple gear looks to be standard pin #2 = hot.

    The only evidence I could turn up of Trident's inclination to use Pin 3 as hot was an old Fleximix manual / available online in pdf form. [Section B-2] Whether it applied to the whole Trident line, I would defer to your expertise.

    As you point out, if the device uses the same hot/cold wiring scheme at the XLRs going in and going out - the net result should be the same at the Main Outputs. Would that be true at the typical TRS Direct Outs?
     
  18. RemyRAD

    RemyRAD Member

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    Chris, there are those of us that grew up with making Mono recordings. Or we'd be sending it out to the disk mastering and pressing plant. And we know what we couldn't send. So I've always mused over any mixer or consoles, that don't have a MONO button on them.

    In fact I know a number of engineers with gold records on their walls, that don't give a damn if some of their stuff as instruments or tracks that are out of phase. Because they wanted it that way. Though I must admit, I did a big jazz festival, shortly after getting my Rupert Neve console. I had not yet restored the console and, the intermittents were killing me LOL. One of the acts included additional Latin percussion. And of the two microphones on the Latin percussion, the phase buttons were so screwed up on my Neve, I only had those two tracks out of phase. It was certainly confounding! But in the end LOL... it sounds absolutely awesome that way! I went back and changed it on the mix down, later. And I missed the Latin percussion, out of phase. It just surrounded you so much more. With the drum set at the front door. And the Latin percussion coming from behind. Luckily I also had plenty of high pass filtering so, if it was to go to vinyl? It still would have been fine given the transient nature of Latin percussion. Certainly not a bass guitar & drums, out of phase trying to be cut to vinyl. So I went back to the Latin percussion, out of phase. It works.

    The real trick for me came when I was cutting jingles at 22 years of age. All on a crappy Yamaha PM-1000 (not a recording console) along with an 8-track MCI JH-110 A. Along with its matching two track. From the Owner of this multimillion dollar, international syndicated advertising agency, to the Chrysler/Budweiser/Ford producer, I was told these would only be strictly mono. I even suggested that I recorded it with the intention of a later stereo release? To which I was overruled at the production meeting. So mono it is. There was only one problem.

    Once, when the over 50 strong sales staff, across the country, heard these new jingles. They all requested stereo mixes to help them sell their clients better with. Whoops! But I had to deliver, stereo jingles. Good thing I had separate overdubs on separate pieces of 8 track tape. But even then still... the basic rhythm section would be forever mono. Bass and drums. Oh well? So I pulled off stereo jingles mono centered voices. Another request came back for stereo voices. Whoops! And then to have to go back through that entire process again, well... I had to do it. So it was quite good that I had recorded for mono release.

    Here is a little example of all that I have left, after the fire at the duplication plant. These are from copies of my archived copies. The tape was so sticky. Almost impossible to hold azimuth and with supply-side tension virtually eliminated. So this is rudimentary recording. The only digital stuff was an Eventide, H-910 Harmonizer. Transferred through my Digidesign, Audio Media 3 card, back in 1996.

    The first half are the experimental stereo mixes. The last half hour of the Mono production mixes. Whatever vocals you hear, are actually dry on the production masters. This allows us to add the reverb, while bringing the lyrics in and out through the different phrasing, between the announcers. Without truncating the reverb. You had to be more dynamic with your Mono mixing. Than you had to be for your stereo mix.

    Back to Mono.
    Mx. Remy Ann David
     
  19. RemyRAD

    RemyRAD Member

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    Right... like I'm firing on all cylinders. I forgot to include the link?


    View: https://soundcloud.com/remyrad/remygolnick2
     
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  20. Boswell

    Boswell Moderator Distinguished Member

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    To the best of my knowledge, there was never a tip=cold, ring=hot convention for TRS jacks to balance the pin 2 = cold, pin 3 = hot XLR convention.

    For TRS inserts, if you used standard cabling into the channel's XLR inputs, you would see an inversion at the TRS insert send. Similarly, there would be an inversion from the TRS insert return to the XLR outputs when used with a standard cable. This would not normally be a problem, as the inserted device would simply be processing inverted signals.
     
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