Again, you need a signal-dependent EQ that operates with knowlege of the exact overall system gain. You've chosen a hard problem.
Quote from: Woodinville on 15 January, 2012, 07:05:16 PMAgain, you need a signal-dependent EQ that operates with knowlege of the exact overall system gain. You've chosen a hard problem.JJ,How is low-volume bass boost different than other "maintaining intensity" problems?
Quote from: hellokeith on 15 January, 2012, 11:20:47 PM ... How is low-volume bass boost different than other "maintaining intensity" problems?It's not, and they all have the same problem.
... How is low-volume bass boost different than other "maintaining intensity" problems?
First, the Fletcher-Munson data is not accurate. See the work of S. S. Stevens, much better. Ever notice how a Fletcher-Munson loudness compensation control never sounds right? That's because they got it wrong to begin with.
Second, Woodinville is right, it IS a hard problem, it's not a fixed curve, and it is highly dependent on the specific acoustic play level, so it has to be dynamic.
Next...why number them...BIG assumption that everybody mixes to a standard level in a standardized monitoring environment. Not in the music industry! Film, yes, but not music. And that -20dbFS would be nice, but doesn't happen after mastering, especially pop stuff. Not even close. Pretty much have to ignore dbFS in this case, it's not relevant. System acoustic play level is though. But in the context of correcting for differing hearing response at differing levels. You're in no way matching the mix environment, there's just no way to know what it was, and it's not important anyway.
No, you can't do it based on a volume control setting. Been tried by many people for many years, but it doesn't work. The reason is simple: the correction required is dependent on SPL, which a volume control may influence but doesn't predict and is not the only thing that affects it. Hotter signal into it, and you turn it down, but that would change the compensation inappropriately. There were even attempts to calibrate the compensation by adding another control, but it doesn't work because program dynamics are not fixed. No, the correction must be tied to specific SPL, not a control setting. That's actually where many people trying this messed up.
... And finally, it's been done, and done quite well. It's called Audyssey Dynamic Volume and Dynamic EQ. Rather than base their idea on existing loudness research, their algorithm is based on what was essentially reverse-engineering human loudness perception. They took LOTs of data on lots of subjects, with lots of different program material and the result is pretty darn good. The big advantage is, once an Audyssey system has been calibrated it knows the exact SPL at every moment regardless of volume control setting or variations in program material, so it can apply the right correction dynamically. Pretty darn smart, those guys.
... And Splice, please realize that the thing you want to build must be signal dependent, and must be tied to absolute presentation level as a function of frequency. Signal dependency is not an option, it's a requirement.
I'm missing some crucial piece of understanding. Please bear with this "bear of very small brain" for a bit..."Signal dependant" - do you mean in time or frequency? To me, one implies a dynamic EQ that adjusts itself according to the current level or spectral content of the signal (e.g Audyssey processor), the other a static EQ, the curve of which is adjusted according to the auditory system behaviour described by the "equal loudness" curves.
Perhaps my statement as to Fletcher-Munson getting it wrong was a bit to generalized. Their data was accurate for the conditions in which it was taken, and the test equipment available in that day. But, since those conditions included pure tones as stimuli presented as a frontal field in an anechoic space, the resulting curves don't represent the actual correction needed for real listening environments. The really unfortunate part of Fletcher-Munson is that the curves became widely adopted, but almost entirely misunderstood. They were applied as complete loudness correction curves, when in fact, they represent human hearing response (in those specific test conditions). Loudness compensation doesn't need to correct for human hearing response, it just needs to correct for the variance in response at differing levels.
Interestingly, F-M and Stevens disagree on loudness growth at low frequencies, and having built systems using both models for loudness growth, I've been much, much more successful with a variation on F-M than I have with Stevens (annoyingly that work belongs to long-former employer, not even a recently former employer, and it hasn't been put to any use at all).As to the flatness concern, once you realize that the bandwidth of the critical bands emerges as a factor, F-M makes a great deal of sense, actually.But, as far as loudness ratio, I've had much more success with loudness ratios using a model I can't talk about (snarl, hiss, grumble) very much that are derived from F-M. Fletcher and Munson show more loudness growth at threshold than Stevens, and that's also been my experience.
... the resulting curves don't represent the actual correction needed for real listening environments. ... Loudness compensation doesn't need to correct for human hearing response, it just needs to correct for the variance in response at differing levels.
To complicate things, as most equal loudness curves go, even Fletcher-Munson, the high frequency portion of the curves above 1KHz are parallel, and so no adjustment is required in that range. But designers applying the F/M equal loudness data to a loudness comp circuit often used the entire curve! So we had boost at the top and bottom, and of course, the wrong amount at the bottom in any case.
Stevens work included a wide variety of stimulus methods, from diffuse, free-field, earphones, etc., and included several subjective quantities as well (annoyance, etc.). One of his test systems extended down to 1Hz! And while that's not useful for loudness compensation, it's notable since other research stops at 20Hz.
As to the supposed loudness compensation built into music by composers (Ravel, et al), their hypothetical compensation is valid for only one listening position: the conductor's podium. Ever other seat in the house will hear something else. However, no seat will have a basic level change anywhere near 20dB. Yet that's the kind of level shifts we see in recorded music played in private listening conditions. With that kind of offset, and looking at any equal loudness contour curve family, anyone can see for this to work it must be dynamic and must operate with the knowledge of actual playback SPL. No fixed modifier would be correct at anything but one specific SPL.
The dual-control loudness compensation idea has been tried (Yamaha, late 1970s, early 1980s, Apt-Holman, 1978), but has not survived even though the Apt-Holman implementation actually applied correction based on the Stevens data. The reason is simple: people can't be depended upon to make continual subjective evaluation and apply correction. Two knobs might get you close, but only at one SPL (at least some music still has dynamic range), and one volume setting. The knob would require constant adjustment, something no listener will do.
Bob Katz has made some excellent inroads in studios, but there's decades of music already recorded and released without any of that, and still today volumes of music released without standardization.
I don't now how else to make the point that compensation must be dynamic, but if all of the above doesn't do it, perhaps ask yourself: if it's so simple as to be a fixed, static correction, why at this point in history have we moved completely away from fixed-curve and dual control systems? Why do the pre-eminaet voices in this field all say it has to be dynamic? Must be something they know. ...
What would be your comment on why F-M didn't work historically? And why the Stevens-based systems worked markedly better, if much more rare? I'd have some ideas, but I'd rather hear it from someone who made a F-M system actually work.
An alternative scheme which may be more user friendly is to again have two knobs - one labeled "volume" and one labeled "bass", which actually sets the operating point of the loudness compensation. Adjust the volume control to your desired level, regardless of the original intended playback level, then adjust the bass control to your taste - "not too heavy, not too light". But behind the scenes, the two controls are actually linked, so any subsequent adjustment of the volume control automatically applies the correct level of loudness compensation.
I remain unconvinced. I'm not proposing a "fixed, static" correction. My proposal is also different than any "dual control" system I have seen, and I have been looking hard. And if by "dynamic" you mean that the EQ adjusts itself based on the (varying) level of the source, then I disagree strongly. That would be equivalent to twiddling the bass tone control to match the loud and quiet parts of the music, and we just don't do that. (Well, I don't, anyway.)
One more time... My system has two knobs. As I originally envisaged it, one knob is more or less "set and forget" for a given genre and input source, especially if the source has Soundcheck or Replaygain. The other knob is the main "volume" control. Adjusting this control also applies the correct amount of "loudness compensation" for that volume. In concept, the bass tone control is ganged to the volume control. Where this differs from other schemes is that the ratio of bass to overall level is fixed, and matches the ratio inherent in the "equal loudness" curves.
You need frequency domain equalization (i.e. a filter curve) that varies with the signal (and of course frequency and presentation level), and where the actual gain of the system post-filter is known to a dB or so.
Quote from: splice on 31 March, 2012, 02:56:00 AM ... if by "dynamic" you mean that the EQ adjusts itself based on the (varying) level of the source, then I disagree strongly. That would be equivalent to twiddling the bass tone control to match the loud and quiet parts of the music, and we just don't do that. (Well, I don't, anyway.) Ok, but that's precisely what is required.
... if by "dynamic" you mean that the EQ adjusts itself based on the (varying) level of the source, then I disagree strongly. That would be equivalent to twiddling the bass tone control to match the loud and quiet parts of the music, and we just don't do that. (Well, I don't, anyway.)
... Yes, I understand what you are saying, but please understand that this has been done, and did not succeed because because of an error in concept outlined in your previous sentence, "the ratio of bass to overall level is fixed, and matches the ratio inherent in the "equal loudness" curves." The curve families show the ratio of bass to overall level is not fixed, it's a non-linear relationship. Because it's non-linear, every time you change the overall level, you operate at a point where the rate-of-change in bass sensitivity is different, and the lower the overall level the faster the rate of change in bass sensitivity. It's the rate-of-change problem that dictates the fact that compensation cannot be fixed. It must track the rate of change of bass sensitivity of the ear. The ear/brain system has what is essentially a volume expander that is both frequency and level dependent. The expansion ratio is dependent on the specific SPL as well as the specific frequency of stimulus. That's why it takes a family of equal loudness curves to show what's actually going on, and also takes something fairly complex and dynamic to perform the compensation. We're kicking around the details of what curve-set to follow in other posts, but they all have this non-linear ratio characteristic.