Sure, but my personal experience is irrelevant beyond having to build circuits and instrumentation and try to figure out when things go sideways. That's the point--those are all unreliable anecdotes (and the plural STILL doesn't make data), things I could make up whole cloth. I don't want to pretend to make an emotional connection and impress some feeling upon you about a 8-legged epoxy-mounted dirty bit of sand.
I like my system. Surely a lot could be done to make it better, and hopefully I'll indulge myself the fun of design challenges and fabrication. Whatever comes of that, so be it!
As for pandering to your ignorant questions about basic circuits, yes, a community college class is clearly a first step for you. If not, then please answer your own questions, I'm not here to do your dirty work. Before you start playing hypotheses about this or that audible effect, surely you clearly understand the circuit theory and operation underlying those differences? Right? You wouldn't dare be asking because you're trying to lead someone down a carefully constructed path to your way of thinking?
Surely you have a selective memory. I said that any changes between most modern opamps (and here comes the caveat that they be working properly) are going to have SMALL electrical differences. This is doubly so within most audio-related uses of an opamp. Then you have to stop and question audibility of SMALL differences.
Stop and think about that for a moment. Again, I never say it's impossible or inaudible. I'm very much saying that changes in a circuit by changing the opamp are either gross (due to an opamp misbehaving) or pretty small. Bounds our expectations.
You really want to exclude people from not having the same viewpoint as you, no? Time and again, whenever someone runs contrary your opinion you wish to shut them out of the discussion. That's pretty pugnacious and obstinate. I'll let Mark run wild with his prognoses here. 😛
As for solidity, please ask yourself, in a brutally honest way, what you contribute to this forum. Beyond sniping and making caricatures out of other members? I'll be honest, content-wise, I provide very little. Ergo why I stay mostly in the lounge area. When was the last time you presented a circuit, a solidly formed idea about a test you ran, with all the details? Or helped another member troubleshoot an issue? Or, even, help him or her brainstorm an idea? Surely I've missed it. Perhaps I make you uncomfortable because the very things you're accusing me of not offering are yours to own?
* I don't give a hoot about your system unless it's to help you troubleshoot an issue, in which case schematics and repair manuals are in order, no?
When a circuit is measured and it behaves, it doesn't mean that it will always behave in the long run or with different auxiliary. To make sure that we use objective term when we talk about stability, no other way, we have to specify e.g. the minimum phase margin of the circuit. I think 45deg has been considered as "stable" but may be not sufficient.
Yes I like them inverting too, so the "main" stage will be inverting. This usually requires extra buffer. If any of the buffer has to be inverting (to preserve the absolute phase) I choose the output buffer not the input.
I'm not talking about the lowest noise power supply possible. Just to follow the high slew rate philosophy the power supply must have current ready at high speed to allow an op amp able to use it's slew rate spec. How is this done? Lets look backwards from the power supply pin of the op amp. First the best film bypass cap(s) you can muster as close to the opamp power pin as possible. 2nd a current source pulling 2x (min) the idle current of the op amp positioned so that it pulls the current past the pin of the op amp from the pass element VR source. 3rd a pass element that is vastly overrated for the amount of idle current it has to deliver.
This matters little though with the op amp models you guys are bandying about. Those "this is good enough for the audio guys" semiconductor EE's have fed you a load of bull sh**. Go read between the lines of the likes of Pease on attitudes about audio and understand that this is pervasive and deeply embedded in the psyche of the guys that make the decisions on what op amp products to label "audio" and feed to everyone.
I'd like to thank Jim Williams of Audio Upgrades for making me see the light.
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Barely greater than 0 degrees (at unity gain bandwidth) is theoretically stable, but we like to usually have a bit more than that. 🙂 (60 degrees is nice)
Even then, when you move past simple loop closures, these answers to "is it stable" becomes quickly "it depends". And what works in small-signal may go utterly sideways as you approach (and broach) clipping, where recovery to "normal" operation needs to be addressed. A lot of older opamps would invert when clipping.
http://www.analog.com/media/en/training-seminars/tutorials/MT-036.pdf
(Sorry for contributing relevant information about limitations to opamps, whether monolithic or discrete, since I'm evidently not supposed to post here)
🙂 I think Mark is not kidding. But ime, it is not about "detail" versus "not detail". Those are only "side effect"...
"Out of tune" is the clue. We can use any means to mask a "bad reality". That's valid (but not hi-fi said the objectivists). The problem is, is the reality that bad? I mean, is it from the source/recording, or rather, from the speaker or amplifier inability to produce the reality?
Yes, I saw that. I read and replied posts starting from the first post I haven't read, without seeing the later posts first 🙂
So, you and Mark are talking about the process in recording. You suggest to make a perfect recording by changing bad instruments with a better one. But you are also aware that sometimes the engineers just don't have the energy to do their homework. This will lead to bad recording material...
Now, how bad is a bad recording? May be you have an experience in a recording studio to listen to the bad sounding master files using STAX headphone and the likes and know if it is so bad that you really need to MASK that reality with distortion...?
Imo, the problem with masking is that you can fix certain problems but you create more problems. For example, you will hear better vocal but the violin will get worse. Or, rock songs will sound better but classical will suffer...
BTW, imo, the best masking trick is probably to roll of speaker response below 20kHz. HF contents are not conducive.
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That would depend on the available test protocols. As long as the protocols are mentioned or accessible, that should be fine.
Toole/Olive tried to find conditions that will allow human to hear the smallest difference. There were differences from individual to individual but in general it was said that pink noise is the easiest. Also better if the frequency is high (or at critical band) so 5kHz was chosen. High Q (Q=1 was chosen) was also believed to improve human ability to detect smallest difference.
Literature mentioned that for each individual, there is different comfortable level, so it is up to the listeners, but I believe that literature was correct saying that higher than 70dB is mandatory for ease of detection. Too high will hurt the ears of course.
morinix,
I would assume that typically you would have the tambourine or other percussion instruments on a separate track so you wouldn't have to apply the same distortion or other techniques on the whole recording but just where needed.
I would assume that typically you would have the tambourine or other percussion instruments on a separate track so you wouldn't have to apply the same distortion or other techniques on the whole recording but just where needed.
It's not the quality of the tambourine that was the problem. It already sounded fine, in a room, and if one listened with both ears. When you close mic something with a directional mic, it sounds different than you might expect. And often not better. An easy experiment is to stick a finger or ear plug in one ear, and stick your other ear right up to something that makes noise. A nice, musical noise would be best. What you find when you get very close is there is more HF detail, and not all of it is all that musical. By using one ear you find that a few room reflections received in stereo can sound a lot better than a dry monaural sound usually does.
That being the case, why close mic in monaural? I liked the suggestion to read Tape Op. It's good advice to start learning about recording. But the basic answer in this case is that for a mix to sound good and uncluttered, and un-muddied, each instrument must be allocated some "space" in the mix. The word space as used here means one or more range or ranges of frequencies for semi-exclusive use by an instrument, per stereo channel. The allotment of some frequency space in a channel can be dynamic, but to hear an instrument clearly, it can't be competing too much with other instruments at the same frequency. Particularly so in in the lower midrange band. Also, if you mic something in stereo, then it takes up space in both channels, at the expense of less room for other sounds. What one finds is that in general what makes an instrument sound best when playing solo, and what makes it sound best in a busy mix, are not the same things.
That being the case, why close mic in monaural? I liked the suggestion to read Tape Op. It's good advice to start learning about recording. But the basic answer in this case is that for a mix to sound good and uncluttered, and un-muddied, each instrument must be allocated some "space" in the mix. The word space as used here means one or more range or ranges of frequencies for semi-exclusive use by an instrument, per stereo channel. The allotment of some frequency space in a channel can be dynamic, but to hear an instrument clearly, it can't be competing too much with other instruments at the same frequency. Particularly so in in the lower midrange band. Also, if you mic something in stereo, then it takes up space in both channels, at the expense of less room for other sounds. What one finds is that in general what makes an instrument sound best when playing solo, and what makes it sound best in a busy mix, are not the same things.
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Thanks for the suggestion and recommendation. I cannot say that I like reading, but I read a lot, every time and everywhere. It is a "planned" reading activity. There is main objective, priorities and work break-down structure.
This should answer some people inability to understand my hesitation to waste my time analyzing Foobar files, troubleshooting/confirming Foobar bugs etc.
Mark and morinix,
Kind of grew up in a recording studio so I just postulated that you would only clean up the single track. I am not a recording engineer but spent way to many hours in the studio. I started working in PA many decades ago. Sat in the studio watching Stanley Clark and many other jazz musicians. I've known Allen Sides of Ocean Way Studios since I was in high school and that was in 1973.
Kind of grew up in a recording studio so I just postulated that you would only clean up the single track. I am not a recording engineer but spent way to many hours in the studio. I started working in PA many decades ago. Sat in the studio watching Stanley Clark and many other jazz musicians. I've known Allen Sides of Ocean Way Studios since I was in high school and that was in 1973.
The first Lounge audio product was a Pultec type eq that had two twists to address these kinds of problems. For your ringing tambourine I took the high frequency boost section and gave it its own extra amplifier for more gain. This allowed one to use the high freq. shelf cut to basically wipe the highs out. Then bring in the high boost to pick what part of the high frequencies sound the best and somewhat reject the rest. It worked well with the two filters fighting each other and caused some soft compression and tilted gelling of the top end. I also added a midrange cut/boost pot that had a phase control that allowed boost with that portion of the frequency 180 deq. out of phase so something like a synth lead be differentiated from a guitar lead. I was thinking of the Jeff Beck/Jan Hammer thing where sometimes it is hard to tell if you are hearing guitar or synth.
All this was done with 20v/us opamps and three terminal regulators because I didn't really know better back then. This product never got off the ground for a number of reasons. It was Lounge's false start.
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My dad was a session drummer here in Los Angeles. He had a Teac 80-8 and Model 5 mixer. I started cutting my teeth on recording on that setup in 1979. Learned analog modular synth at Pasadena City Collage 1980. Went to ITT Tech '86-"88. Worked at Alesis all through the 90's. Did a short stint at Hoveland, then GrooveTubes. Then 8 years at the great Harman/JBL factory in Northridge (the best job and people to work with I ever had). Then 2010 Harman layoff and started Lounge.
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morinix,
If you know percussion and drums then you may know my friend Airto Morrera who was the number one percussionist for about a decade with Down Beat magazine. I live about five minutes away from him. Flora Purim his wife who is now living in Brazil sang with Chick Correa and Return For Forever. Her daughter is my good friend and I've known her since she was in high school. I manufactured parts for the JBL XPL speaker system. I had something to do with the original EON speaker.
If you know percussion and drums then you may know my friend Airto Morrera who was the number one percussionist for about a decade with Down Beat magazine. I live about five minutes away from him. Flora Purim his wife who is now living in Brazil sang with Chick Correa and Return For Forever. Her daughter is my good friend and I've known her since she was in high school. I manufactured parts for the JBL XPL speaker system. I had something to do with the original EON speaker.
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