Forum Replies Created
-
AuthorPosts
-
2019/07/19 at 10:48 pm #85355
Addenda:
The “AudioWiki” https://www.sengpielaudio.com/calculator-db-volt.htm
Attachments:
You must be logged in to view attached files.2019/07/19 at 10:38 pm #85354V…
***content removed – personal comment that does not contribute to the thread*** Seemingly simple statements such as “clipping starts at 0dBFS” is so incorrect and misleading. For one thing, the “0dBFS” point is not a “starting point”. A digital signal by definition cannot exceed 0dBFS, thus in your terminology it would be an ending point. It s the proverbial “brick wall”. Such seemingly small discrepancies are anathema to those seeking a fuller understanding.
Clipping can occur with a live sound system at many points and may or may not be subject to detection by metering. Most such metering is RMS if for no other reason than the longer measurement sample is more “real world” than transient peak measurement…which is very difficult to implement.
The meters can’t tell you how it sounds. You still have to use your ears.
Next time you cite some document or white paper on any subject, be so kind as to provide complete citation and a link. As is, all we have is your interpretation of the un-named source, comparing it to “internet opinion”…***content removed – personal comment that does not contribute to the thread***
2019/07/19 at 7:27 pm #85350There is nothing , per the technical paper, not some internet opinion, that will maximise intelligibility except UPwards compression. That is based on measuring our room, noise, yada yada, and attempting to match the optimum S/N and SPL level according to the tech paper.
This interpretation/assertion is not correct. You are considering only volume levels, not sonic content, speaker system deployment/tuning and other salient factors. If the HVAC is noisy, see about putting some proper baffling in the ductwork. If this is not practical, implement some noise cancelling.
It would seem you are treating the symptoms, not the cause.
2019/07/19 at 7:16 pm #85348Which red light? What documentation? Why dbfs?
The Qu metering/lights are dbVu, but whichever way you prefer to measure, the user guide simply states that the red LED associated with the metering system lights 3dB before the onset of clipping for either dBfs or dBvu. See attached screen shot:
Attachments:
You must be logged in to view attached files.2019/07/18 at 11:07 pm #85340V…
If your quest is for vocal intellgibility in live sound, dynamic processing such as compression or expansion of the problem source/channel is not the path to take. Instead of managing dynamic range and mixing by volume, you need to deal with it in the sonic realm. This is more easily done on a properly deployed and tuned PA system.
To fairly address the problem it needs to be defined. As is, the question seems to relate to the technique you assume will solve the problem without ever having really defined the problem, mentioning only dynamic range, background noise and such. As stated above…if your quest is vocal intelligibility, dynamc processing is the wrong path.
Mixing live audio is more than controlling envelopes and balancing volume levels. So…
What is your desired result: to achieve vocal intelligibility or have your idea of using dynamic processing validated?
2019/07/16 at 5:19 pm #85267A good explanation for those interested in the four basic forms of dynamics: https://www.izotope.com/en/blog/mixing/expanding-on-compression-3-overlooked-techniques-for-improving-dynamic-range.html
My 2p: I don’t think this problem can be solved purely in the desk using dynamics. Some form of noise rejection is needed. Perhaps even look at a differential pair? Good old fashioned physics might be the solution here.
DC
David…
All well and good IN THE BOX. The link clearly speaks of this as a studio technique rather than live audio. The curious may attempt to export such studio methods to a live sound scenario, but whether or not it succeeds with active sound reinforcement is kind of a crap shoot. I work in both worlds as well as live broadcast and a good few basic processes serve across the board. But in the case under “discussion” here, such transposition results in diminished system headroom. If you have plenty to begin with, then you are in the lucky minority. If you lack headroom you’ll soon run into some pretty ugly consequences.
2019/07/16 at 4:48 pm #85266Let’s take a look at the process and procedure here:
A major problem with ping-pong posting in quest of answers and solutions is the tendency of the initial statement (OP) to be couched in terms of the Socratic method rather than the Scientific method ( https://en.m.wikipedia.org/wiki/Scientific_method ). So rather than seeking expansion of or addition to ones knowledge, the objective tends toward seeking affirmation of ones preconceived opinion on an issue of which little knowledge or actual experience exists.
There are many “tells” in such postings. In this case the inquiry regarding the possible efficacy of using AMM and the statement that dynamic processing does not affect system headroom (GBF) indicate: 1. an unfamiliarity with Dugans algorithms and 2. a common mistake carried over by studio folk used to working “in the box” rather than a live audio scenario involving a PA system.
***content removed – personal comment that does not contribute to the thread***
2019/07/16 at 12:41 pm #85260someone come in who knows what their doing and go through the system front to back to make sure it’s operating as the best it can be.
Exactly. It would seem that this entire thread is based on several questionable assumptions, little or no information on system components/deployment, program material and other salient facts.
What is the problem? One time it’s dynamic range, another time it’s lack of mic technique, then it’s voice intelligibility. These are three separate albeit interactive points of focus. There’s no “one button fixes all” solution. If anything, patiently working with the mic users over time to instill good technique is the best overall approach.
But as quoted above, any proposed remedy based on the various possible techniques assumes a competently deployed, tuned and maintained PA to be effective. Without comprehensive and specific information in these regards, everything is pretty much in WAG territory…except the bit about mic technique.
2019/07/16 at 12:53 am #85240I am retired, EE degree, ABD math, PhD Computer Science, and worked as a systems engineer. And taught at 3 unis. And I can read and grok technical info quite well.
I am more than qualified to understand what is going on and what our options to improve things are.I have worked with recording since the 60s when I was a member of AES but did not pursue audio full time as a career.
I helped with a small Behringer analog board at my last church. We now just got a new digital AH board to use at our new church.
While I did not do media mastering, the personal work I have done with a DAW to edit mix and premaster is quite applicable.IOW, you have almost zero experience in live sound, the field in which I began by building my own system components from kits back in the early 1960’s and continue to the present. None of your CV is applicable to the real world of live audio engineering.
2019/07/13 at 9:47 pm #85179gain before feedback will not change at all.
no feedback problem when they talk louder so no way for upwards compression to cause feedback at all.This is just wrong.
***content removed – personal comment that does not contribute to the thread***
2019/07/13 at 5:51 pm #85169As intimated in the two previous posts, when nothing you try is working there is a better than even chance that the problem lies with you.
2019/07/10 at 11:25 pm #85100This is a quest for a technical solution to a human problem. In such situations I have gotten all parties involved to sit down together and come to an understanding of the requirements and responsibilities involved in effctive presntation and participation.
It has been successful more often than not.2019/07/10 at 7:48 pm #85090Friend…
This forum is for discussion of Qu series mixers, not for getting educated in the basics of sound reinforcement. I suggest you join and post your requests for general audio tasks on the Pro Sound Web. There is only one requirement: you must use your real, full name to participate.
I’ll leave you with one hint:
You need limiting, not compression. Yes, you can set up a compressor as a limiter, hence it being called a compressor/limiter in common audio parlance.
Good luck.
2019/06/25 at 6:17 pm #84828It’s not rocket surgery:
Analog splitters. Much cheaper than a stage box and many uses…to split, NEVER to combine.
2019/06/18 at 10:44 pm #84690You have something inserted on that Mix. Could be any FX or processing.
-
AuthorPosts
