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#41
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substractive vs. additive equalizing.
But if you're trying to fix an error in a mic or a room,
the EQ should make things better. That's what we've been led to believe. I'm not fully convinced. With respect to room EQ... Severe room resonances often remain audible, even though the analyzer reports a net flat response. I've heard this. The problem is almost certainly caused by the inability of, shall we say, two-dimensional equalization to correct a three-dimensional error. |
#42
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substractive vs. additive equalizing.
Roy W. Rising wrote:
Ethan Winer wrote: On Jul 17, 1:43 pm, Roy W. Rising wrote: Really steep slopes cause objectionable phase distortions. I've heard them! How do you know that what you heard was phase shift? Using the best descriptive words you can muster, describe what you believe to be the objectionable part of a roll-off with a slope steeper than 6 dB per octave. --Ethan OK. I do not know that the cause was phase shift. That said, the mode was not roll-off. The device was a '60's McCurdy EQ with 3- or 6KHz haystack boost. When used with more than about 6dB of 6KHz boost, sources had what I can describe only as a "pinched" characteristic. I think it was masked in the JBL monitors being used, it became evident when EV Sentry 500s became available. The system was very well engineered, I seriously doubt anything down stream was being overloaded. In another circumstance, a UREI 1/3 octave graphic EQ was being used to correct for known minor discrepancies in a 'standard' mic. A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. Whatever was bothering them was gone. I can't remember which publication long ago raised this topic and offered the "6dB/octave" suggestions. The arguments were persuasive. Probably not the article you're thinking of but an interesting read: Michael Gerzon's "Why do equalisers sound different?" http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf Later... Ron Capik -- |
#43
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substractive vs. additive equalizing.
Michael Gerzon's
"Why do equalisers sound different?" http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf Michael Gerzon was truly a genius, and I would normally defer to his opinion about anything. But I don't think he's correctly interpreting Harwood's discovery... In the late 1950s, H D Harwood at the BBC made a discovery whose importance is still not fully recognised. In investigating the performance of loudspeakers, he discovered that low-level delayed resonances severely coloured the reproduced sound even if these resonances were 40dB below the main speaker response. At first sight there's nothing very world-shaking about that. But consider what the effect of such a delayed resonance is on the amplitude and phase response; 40dB down means a signal whose amplitude is only 1% of the main signal. This means that the amplitude response must vary only between 99% and 101% of flat, ie within ±0.1 dB. The effect on phase response must similarly be within 1/100 rad, ie within ±0.6o. In other words even in the late 1950s Harwood showed that variations in phase response of around only 1o and in amplitude response of ±0.1dB produced audible colouration. I'm not going to get into any discussion about this. Draw your own conclusions. |
#44
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substractive vs. additive equalizing.
Roy W. Rising wrote:
OK. I do not know that the cause was phase shift. That said, the mode was not roll-off. The device was a '60's McCurdy EQ with 3- or 6KHz haystack boost. When used with more than about 6dB of 6KHz boost, sources had what I can describe only as a "pinched" characteristic. At an early morning guess it was ringing. In another circumstance, a UREI 1/3 octave graphic EQ was being used to correct for known minor discrepancies in a 'standard' mic. A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. Whatever was bothering them was gone. Sharp peaks are always annoyingly audible. So if you eq to fix something and misses because of not using a parametric it gets easy to produce a sharp peak next to the dip you're trying to fix. FFT analysis can be great at telling you where and how large that pesky peak that needs fixing is, my CK1's had an 8 dB peak at 16 kHz when I bougth them some 10+ years ago, now it is only a 6 dB peak ... correcting it makes violins and multiple singing wimmen sound right, otherwise "there is something wrong". Kind regards Peter Larsen |
#45
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Delayed resonance, was: " substractive vs. additive equalizing."
William Sommerwerck wrote:
In the late 1950s, H D Harwood at the BBC made a discovery whose importance is still not fully recognised. In investigating the performance of loudspeakers, he discovered that low-level delayed resonances severely coloured the reproduced sound even if these resonances were 40dB below the main speaker response. He wrote about it in Wireless World, I think, in the 1970's. Also about the importance of directivity index. At first sight there's nothing very world-shaking about that. But consider what the effect of such a delayed resonance is on the amplitude and phase response; 40dB down means a signal whose amplitude is only 1% of the main signal. This means that the amplitude response must vary only between 99% and 101% of flat, ie within ±0.1 dB. The effect on phase response must similarly be within 1/100 rad, ie within ±0.6o. In other words even in the late 1950s Harwood showed that variations in phase response of around only 1o and in amplitude response of ±0.1dB produced audible colouration. No he didn't, what he explained was that the delayed resonance was griveously audible BECAUSE the output from the resonance continued after the exiting signal had stopped. And that is why you can't fix a poor transducer with EQ, and why metal center domes are so darn obnoxious. I'm not going to get into any discussion about this. Draw your own conclusions. Kind regards Peter Larsen |
#46
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substractive vs. additive equalizing.
"Roy W. Rising" wrote in
message Really steep slopes cause objectionable phase distortions. I've heard them! This sentence misses being 100% correct for the lack of a little word: can. It takes a ton of phase shift equally applied to both channels to be audible. 1,000 degrees in the midrange, more at higher frequencies. It takes almost no phase shift applied to just one channel that is mixed with other similar channels to be audible. There are steep slope digital filters that have no phase distortion to speak of. They are called "linear phase filters". They are commonly used on better DAC chips. They can still have characteristic audible colorations, or not. It all depends. |
#47
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substractive vs. additive equalizing.
"Ron Capik" wrote in message
Probably not the article you're thinking of but an interesting read: Michael Gerzon's "Why do equalisers sound different?" http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf The short answer to the question: "Why do equalisers sound different?" is: because that is what they are supposed to do. ;-) When someone says that they tried to equalize something and it sounded worse, then they either had inadequate tools, or were personally not up to the task, or were on mission impossible. Sometimes it is impossible to know which is the problem. |
#48
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substractive vs. additive equalizing.
"Ty Ford" wrote in message
al.NET When speaking in real-time though one of Bob's boxes with the phase rotators engaged and headphones on, your own voice sounds "stuffy", sort of like you have a head cold. That effect is no big challenge with a straight up parametric eq. |
#49
Posted to rec.audio.pro
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substractive vs. additive equalizing.
Michael Gerzon's
"Why do equalisers sound different?" http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf The short answer to the question: "Why do equalisers sound different?" is: because that is what they are supposed to do. That isn't at all what he meant. He meant... "Why do equalizers with nominally 'identical' settings sound different?" |
#50
Posted to rec.audio.pro
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substractive vs. additive equalizing.
"William Sommerwerck" wrote in
message Michael Gerzon's "Why do equalisers sound different?" http://www.audiosignal.co.uk/Resourc...fferent_A4.pdf The short answer to the question: "Why do equalisers sound different?" is: because that is what they are supposed to do. That isn't at all what he meant. He meant... "Why do equalizers with nominally 'identical' settings sound different?" All answers to that question involve linear distortion, nonlinear distortion, gain staging, and/or noise. There are no effective standards for the correlation between the controls and the response (amplitude or phase) curves on equalizers. Not all equalizers have inaudible distortion at extreme settings, even if they have inaudible distortion at nominal settings. Equipment varies in terms of noise and clipping points, both input and output. The variances are not all well documented. |
#51
Posted to rec.audio.pro
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substractive vs. additive equalizing.
"Peter Larsen" wrote:
Roy W. Rising wrote: OK. I do not know that the cause was phase shift. That said, the mode was not roll-off. The device was a '60's McCurdy EQ with 3- or 6KHz haystack boost. When used with more than about 6dB of 6KHz boost, sources had what I can describe only as a "pinched" characteristic. At an early morning guess it was ringing. Could be. The cause??? In another circumstance, a UREI 1/3 octave graphic EQ was being used to correct for known minor discrepancies in a 'standard' mic. A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. Whatever was bothering them was gone. Sharp peaks are always annoyingly audible. So if you eq to fix something and misses because of not using a parametric it gets easy to produce a sharp peak next to the dip you're trying to fix. FFT analysis can be great at telling you where and how large that pesky peak that needs fixing is, my CK1's had an 8 dB peak at 16 kHz when I bougth them some 10+ years ago, now it is only a 6 dB peak ... correcting it makes violins and multiple singing wimmen sound right, otherwise "there is something wrong". Kind regards Peter Larsen In the second example, my use of "minor discrepancies" was to connote that the adjustments were less than plus/minus 2dB. Adjacent sliders were less than 1dB apart. No "sharp peaks" were present. -- ~ Roy "If you notice the sound, it's wrong!" |
#52
Posted to rec.audio.pro
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substractive vs. additive equalizing.
On Jul 19, 8:03 pm, Roy W. Rising
wrote: OK. I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. You already got good replies from Peter and Arny. I'll add only that people seem to blame "phase shift" for various ills they can't describe. This is a great example. I know a guy who swears he can hear the phase shift in a 10-foot guitar cable, which of course is ridiculous. But he absolutely believes that's what he hears! If you watch my Audio Myths video I linked earlier there's a demo of phase shift. I urge you to download and play around with the free Sanford Phaser VST plug-in used in that demo: http://www.ethanwiner.com/aes/ --Ethan |
#53
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substractive vs. additive equalizing.
OK. *I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". *None could be any clearer. *When the EQ was flattened, all agreed to leave it that way. steep filters create ringing due to Gibbs phenomenon no matter how the phase or group delay has been equalized If you take a perfect square wave and simply truncate the harmonics, without any change in phase or anything to the the remaining harmonics, you will still have Gibbs ringing steep min phase filters cause the most familiar post ringing steep filters with linear phase and flat group delay have equal pre and post ringing you can argue that pre ringing is more objectional compared to post ringing because we are not accustomed to hearing that that leads to the conclusion that stepe filters with min phase may sound better compared to steep filter with linear phase... but the best sounding filters are filters that are not steep. it is an unfortunate fact of nature that you simply can't create a steep change in the frequency response without messing up the time domain response. this is not a limitation of analog or digital, it is just the way nature works.. http://en.wikipedia.org/wiki/Gibbs_phenomenon Mark |
#54
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substractive vs. additive equalizing.
"Mark" wrote in message
OK. I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. steep filters create ringing due to Gibbs phenomenon no matter how the phase or group delay has been equalized If you take a perfect square wave and simply truncate the harmonics without any change in phase or anything to the the remaining harmonics, you will still have Gibbs ringing This should be a major concern to people who spend a lot of time listening to ideal square waves and ideal impulses. When you're dealing with musical sounds, everything is ringing quite a bit anyhow. That's what musical instruments do - they resonate and ring. steep min phase filters cause the most familiar post ringing So do virtually all musical instruments. steep filters with linear phase and flat group delay have equal pre and post ringing The good news is that linear phase filters most popular use is as brickwall filters in digital, and the sample rate is usually high enough that the ear's sensitivity to *anything* is very low. you can argue that pre ringing is more objectional compared to post ringing because we are not accustomed to hearing that Pre-ringing from linear phase filters is probably most audible when people downsample to sample rates with nyquist frequencies in the normal audible range. that leads to the conclusion that stepe filters with min phase may sound better compared to steep filter with linear phase... Or not. A conclusion with the word "may" in it is arguably not a conclusion but a speculation. Regardless, its not exactly exact. but the best sounding filters are filters that are not steep. Unless you use the steep filter to deep-six some objectionable sound at a frequency they put out-of band, which is mostly why people use them. it is an unfortunate fact of nature that you simply can't create a steep change in the frequency response without messing up the time domain response. AFAIK nobody heavily filters music with filters that run inside the audio band, that they already like the sound of. this is not a limitation of analog or digital, it is just the way nature works.. Right. Most of the things that people complain about when digital does it, they would complain about it if analog was commonly used (or even able!) to do it. |
#55
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substractive vs. additive equalizing.
If you take a perfect square wave and simply truncate the
harmonics without any change in phase or anything to the remaining harmonics, you will still have Gibbs ringing. That's not Gibbs ringing. It's simply a waveform. |
#56
Posted to rec.audio.pro
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substractive vs. additive equalizing.
On Jul 20, 6:17*pm, "William Sommerwerck"
wrote: If you take a perfect square wave and simply truncate the harmonics without any change in phase or anything to the remaining *harmonics, you will still have Gibbs ringing. That's not Gibbs ringing. It's simply a waveform. There are numerous references about Gibbs phenomenon and ringing. There ARE other forms of ringing that could be called "simply a waveform" but the ringing on a step (square wave) caused by a steep filter (truncation of Fourier components) is called Gibbs. I don't care to debate this with you.. Mark |
#57
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substractive vs. additive equalizing.
Arny Krueger wrote:
This should be a major concern to people who spend a lot of time listening to ideal square waves and ideal impulses. Isn't that what synthesizers (and their equivalent virtual instrument plug-ins) mostly produce? AFAIK nobody heavily filters music with filters that run inside the audio band, that they already like the sound of. Your definition of "music" is traditional. Today's music is frequently mangled with heavy filtering that's modulated so that a rather boring sound becomes dynamic. It starts with a wah-wah pedal and goes up (in degrees of phase shift) from there. Those of us who record what we hear are less concerned with those effects because we don't intentionally apply them. -- "Today's production equipment is IT based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson |
#58
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substractive vs. additive equalizing.
Roy W. Rising wrote:
"Peter Larsen" wrote: Roy W. Rising wrote: OK. I do not know that the cause was phase shift. That said, the mode was not roll-off. The device was a '60's McCurdy EQ with 3- or 6KHz haystack boost. When used with more than about 6dB of 6KHz boost, sources had what I can describe only as a "pinched" characteristic. At an early morning guess it was ringing. Could be. The cause??? A filter is always also a resonator, the use of the wording Q says it, and a 1960's McCurdy EQ - whatever that is since it pre-dates my audio experience - is likely to use a combination of induction and capacitance, ie. have real induction coils in each band; real components do not always exhibit theory-compliant performance. Iron cored inductors may have been what was in it, and those are "un-simple" components that are not good at behaving like simple theoretical models suggest that they should. Also - at a guess - they may have been dimensioned so that they were saturated when a large boost was used with a strong signal level present. That surely would sound "pinched" referencing my experience with intentionally overloading Bang and Olufsen tape reocrders used for tape echo by having a treble boost in the loop. In another circumstance, a UREI 1/3 octave graphic EQ was being used to correct for known minor discrepancies in a 'standard' mic. We need to talk about what you can and can not fix when it comes to transducer oddities. Frequency response per se is easy to address, such as the example issue with the CK1's and what appears to be a resonance in front of the membrane. But frequency response aberrations in transducers do NOT have the neat shape in the specsheet, it is smoothed so that a minor bump may hide s narrow sharp peak, as with that CK1 violin sound issue. A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. Whatever was bothering them was gone. Some of the time it is like that when EQ'ing transducers, it could be fun to speculate along lines such as whether the mind knows a "probable" combination of colorations from an "improbable" one in case one doesn't get the response fix exactly right. Some transducers have aberrations that are musical - Shure 565 is to my sonic preference such a transducer and some are obnoxious, to my sonic preference the SM58 is just that, I reckon the grid is ""acoustically worse"". [une snippage] In the second example, my use of "minor discrepancies" was to connote that the adjustments were less than plus/minus 2dB. Adjacent sliders were less than 1dB apart. No "sharp peaks" were present. Roy, here you are plain wrong because of not considering the whole system and because of being mmislead by smoothed measuring data. You were equalizing a transducer, such always have sharp peaks - good transducers just don't have quite as large ones as bad transducers - and a narrow sharp 8 dB peak that is lowered by 2 dB and has the area on both sides of it lowered with it is quite likely to sound worse rather than better, this because the peak is still there relative to its surround response and a wider dip has been added, worst case changing somethng that is audibly obnoxious to something that is obnoxious combined with something that is audibly boring. Some of the time the ""thingies"", be it monitor loudspeakers or recording mics, are better left sounding as the do ex works; ref. also the general "can I eq my monitors + room discussion" that comes up every now and then, frequency response fixes to my sonic taste have to match the actual frequency response aberration. Kind regards Peter Larsen |
#59
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substractive vs. additive equalizing.
On Jul 20, 4:46 pm, "Arny Krueger" wrote:
When you're dealing with musical sounds, everything is ringing quite a bit anyhow. That's what musical instruments do - they resonate and ring. I nominate this for Quote of the Week. --Ethan |
#60
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For live sound, it really depends on where you're at in the signal chain. For mains EQ I rarely boost any eq because of how it sonically makes the the overall sound strange to my ears.
So I'd be a "minus only" guy for mains EQ and use an FFT scope to figure out where to cut - it's most often somewhere around 250 because of room excitement and design. Yes there are some FFT areas where it would seem you'd want to boost but I don't. For channel eq there's no limit - it's just what sounds right + or -
__________________
www.livesoundaudio.com Sacramento Audio Rentals |
#61
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substractive vs. additive equalizing.
Ethan Winer writes:
On Jul 20, 4:46 pm, "Arny Krueger" wrote: When you're dealing with musical sounds, everything is ringing quite a bit anyhow. That's what musical instruments do - they resonate and ring. I nominate this for Quote of the Week. And as Ethan will appreciate, rooms where we might hear those instruments will also resonate and ring -- but hopefully in a good way (such as a good, repertoire-appropriate concert hall), rather than a small bedroom studio with +/- 20 dB below 200 Hz, with RT30 componenents equally as crazy! Frank Mobile Audio -- |
#62
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substractive vs. additive equalizing.
Ethan Winer wrote:
You already got good replies from Peter and Arny. I'll add only that people seem to blame "phase shift" for various ills they can't describe. This is a great example. I know a guy who swears he can hear the phase shift in a 10-foot guitar cable, which of course is ridiculous. But he absolutely believes that's what he hears! I'll bet he can hear the _amplitude_ alterations made by the high frequency rolloff in a 10-foot guitar cable, though. Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway? --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#63
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substractive vs. additive equalizing.
Ethan Winer wrote:
On Jul 19, 8:03 pm, Roy W. wrote: OK. I do not know that the cause was phase shift ... sources had what I can describe only as a "pinched" characteristic ... A group of 'golden ears' listened to reference material and characterized what they heard as "something wrong". None could be any clearer. When the EQ was flattened, all agreed to leave it that way. You already got good replies from Peter and Arny. I'll add only that people seem to blame "phase shift" for various ills they can't describe. This is a great example. I know a guy who swears he can hear the phase shift in a 10-foot guitar cable, which of course is ridiculous. But he absolutely believes that's what he hears! But he may be right about that. If the input is a 1M input, a wee amount of capacitance can make a (subtle) difference. Might not be phase shift, though. Guitar players are really serious about the guitar-amp interconnect. But they won't (usually) buffer... If you watch my Audio Myths video I linked earlier there's a demo of phase shift. I urge you to download and play around with the free Sanford Phaser VST plug-in used in that demo: http://www.ethanwiner.com/aes/ --Ethan -- Les Cargill |
#64
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substractive vs. additive equalizing.
On Wed 2038-Jul-21 12:35,
Live Sound Audio writes: For live sound, it really depends on where you're at in the signal chain. For mains EQ I rarely boost any eq because of how it sonically makes the the overall sound strange to my ears. WOuld agree with that. WHen I've had a knowledgeable person to read appropriate displays etc. I've used pink noise and analysis tools, otherwise, old blind sound dog uses those two things on the side of his head. FOr channel strip eq I try to avoid it if at all possible. If I can't get a channel sounding good with a modest cut (preferred) or boost then I try for a different capture technique, i.e. move microphone, or a different mic altogether. Regards, Richard .... Remote audio in the southland: See www.gatasound.com -- | Remove .my.foot for email | via Waldo's Place USA Fidonet-Internet Gateway Site | Standard disclaimer: The views of this user are strictly his own. |
#65
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substractive vs. additive equalizing.
On Jul 16, 4:34*pm, (Scott Dorsey) wrote:
Nono wrote: After being confused for a while after reading and hearing all this propagating of substracting EQ-ing as a better alternative for additive EQ-ing, I come to the conclusion that this claim is inacurate to say the least. It is a good rule of thumb, but it is not a hard and fast rule in any way and you don't need to overintellectualize it. It is a good idea to think about tone shaping in terms of cutting things out, and many equalizers sound better that way. It is like the three-to-one rule for microphone placement; it's usually a good idea but it's no more than that. *However, it can be a good rule to follow until you learn the exceptions. --scott -- "C'est un Nagra. *C'est suisse, et tres, tres precis." ______________________ Not *exactly* on topic - but involves use of an equalizer: I analyzed several "Equal-Loudness Contour"s(essentially modified F-M curves), and applied the settings to my Windows Media player EQ. Assumptions: We do not know the Q of each band's center point on the 10-band EQ that WMP comes with. Classic Fletcher-Munson: http://www.customanalogue.com/elsino...unson_700W.gif Updated Equal-Loudness Contour: http://upload.wikimedia.org/wikipedi...indos1.svg.png (I find that correlating to the 60 or 80 phon curve yields best results for moderate to low-level listening.) That said, I do find that when I approximate the curves in the above links, I find my self listening to music for longer periods at a time, at lower volumes, through both headphones and through loudspeakers! It's quite phenomenal. I subconsciously find myself hearing MORE of the recording at LOWER avg. volume levels! I encourage anyone to try this and report back their results. The Settings I derived: Windows MP EQ: 31Hz +12dB 62 +6dB 125 +1dB 250 0 adjustment(leave flat) 500 -1dB 1kHz 0 adjustment(flat) 2kHz -1dB 4 -3 8 +4 16 leave flat Again these are for MODERATE to LOW volume listening ( 50dB and under) - not for rocking your house party or jamming through your Sennheisers(!) Listening too loud may not harm your ears - but it could clip your amp or damage your speaker drivers with all that bottom. -ChrisCoaster |
#66
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substractive vs. additive equalizing.
Richard Webb wrote:
[...] FOr channel strip eq I try to avoid it if at all possible. If I can't get a channel sounding good with a modest cut (preferred) or boost then I try for a different capture technique, i.e. move microphone, or a different mic altogether. If the sound is wrong, shouldn't you try moving the mic first? -- ~ Adrian Tuddenham ~ (Remove the ".invalid"s and add ".co.uk" to reply) www.poppyrecords.co.uk |
#67
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substractive vs. additive equalizing.
On Wed, 21 Jul 2010 15:08:42 -0400, Scott Dorsey wrote:
Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Pedantically, it's the high source impedance of the pickup (designed to require 1M input at the amplifier) that screws things up with capacitance. Whoever decided that was a good idea anyway? I've always assumed it was a legacy from tube amplifier days, when 1M input impedance was normal anyway, and a high-z pickup coil gave you a nice big signal. For the same reason mics used to come with High-z, at least as an option. -- Anahata ==//== 01638 720444 http://www.treewind.co.uk |
#68
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substractive vs. additive equalizing.
"Live Sound Audio"
wrote in message For live sound, it really depends on where you're at in the signal chain. For mains EQ I rarely boost any eq because of how it sonically makes the the overall sound strange to my ears. So I'd be a "minus only" guy for mains EQ and use an FFT scope to figure out where to cut - it's most often somewhere around 250 because of room excitement and design. If you're trying to equalize out dips due to acoustic cancellation by peaking the electrical response, you're on mission impossible. You can waste amazing amounts of power in these dips, and have very little to show for it but hot equipment. |
#70
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substractive vs. additive equalizing.
On Jul 21, 3:08 pm, (Scott Dorsey) wrote:
Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway? Les Paul? Leo Fender? :-) --Ethan |
#71
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substractive vs. additive equalizing.
Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote: Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway? Les Paul? Leo Fender? :-) --Ethan Les Paul used a lo-Z Les Paul Recording for some span of time. -- Les Cargill |
#72
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substractive vs. additive equalizing.
ChrisCoaster wrote:
On Jul 16, 4:34 pm, (Scott Dorsey) wrote: Nono wrote: After being confused for a while after reading and hearing all this propagating of substracting EQ-ing as a better alternative for additive EQ-ing, I come to the conclusion that this claim is inacurate to say the least. It is a good rule of thumb, but it is not a hard and fast rule in any way and you don't need to overintellectualize it. It is a good idea to think about tone shaping in terms of cutting things out, and many equalizers sound better that way. It is like the three-to-one rule for microphone placement; it's usually a good idea but it's no more than that. However, it can be a good rule to follow until you learn the exceptions. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." ______________________ Not *exactly* on topic - but involves use of an equalizer: I analyzed several "Equal-Loudness Contour"s(essentially modified F-M curves), and applied the settings to my Windows Media player EQ. Assumptions: We do not know the Q of each band's center point on the 10-band EQ that WMP comes with. ...snip... -ChrisCoaster However, "we" can surly measure the 10-band EQ, can "we" not? Later... Ron Capik -- |
#73
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substractive vs. additive equalizing.
Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote: Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway? Les Paul? Leo Fender? Les Paul decided it was a bad idea and started making guitars with a 600 ohm balanced "recording" pickup for a while.... no more hum, no more buzz, no more cable issues... --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#74
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substractive vs. additive equalizing.
Ethan Winer wrote:
On Jul 21, 3:08 pm, (Scott Dorsey) wrote: Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway? Les Paul? Leo Fender? There was a Les Paul Recording model with an XLR connector and low impedance pickups, and 'ol Chet also used low impedance pickups at one time. I suppose it wasn't embraced by the manufactueres for a few reasons: - Cost. They'd need more gain and a more expensive connector - Compatibility. They'd have to make amplifiers with both low-Z and high-Z inputs for the next 50 years - Documentation and education. They'd have to teach all the guitar players, and even harder, all the dealers, what it's all about. -- "Today's production equipment is IT based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson |
#75
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substractive vs. additive equalizing.
On Jul 22, 1:49*pm, Ron Capik wrote:
However, "we" can surly measure the 10-band EQ, can "we" not? Later... Ron Capik --- Hide quoted text - - Show quoted text - ________________ Expand on that thought please, Ron? -CC |
#76
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substractive vs. additive equalizing.
ChrisCoaster wrote:
On Jul 22, 1:49 pm, Ron Capik wrote: However, "we" can surly measure the 10-band EQ, can "we" not? Later... Ron Capik --- Hide quoted text - - Show quoted text - ________________ Expand on that thought please, Ron? -CC I'm guessing there must be dozens of programs out there that can be used to measure the Q of an equalizer. One way would be run some pink noise through the Media Player then set the EQ to some value and run the same noise through the Media Player. Take FFTs of both and subtract the two FFTs. In a quick experiment using your settings I got something like: 31 +11.2 62 +7.5 125 +3.3 250 [set as 0.0 reference] 500 -1.9 1L -2.0 2k -2.0 4k -3.6 8k +2.0 16k ~-1.0 Later... Ron Capik -- |
#77
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substractive vs. additive equalizing.
On Jul 22, 2:13 pm, (Scott Dorsey) wrote:
Les Paul decided it was a bad idea and started making guitars with a 600 ohm balanced "recording" pickup for a while.... no more hum, no more buzz, no more cable issues... Thanks Scott (and Les and Mike). Yes, I actually remember that now. :- ) --Ethan |
#78
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substractive vs. additive equalizing.
Mike Rivers wrote:
Ethan Winer wrote: On Jul 21, 3:08 pm, (Scott Dorsey) wrote: Doesn't take much shunt capacitance to totally screw things up when you have a 1M input impedance? Whoever decided that was a good idea anyway? Les Paul? Leo Fender? There was a Les Paul Recording model with an XLR connector and low impedance pickups, and 'ol Chet also used low impedance pickups at one time. I suppose it wasn't embraced by the manufactueres for a few reasons: There were two versions of this guitar, the first had the XLR connector and an impedance converter as part of the cable. That was the Les Paul Professional. It was followed after no more than two years by the Les Paul recording that had a normal output connector and had a switch on the face plate to switch between high and low impedance. |
#79
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substractive vs. additive equalizing.
On Jul 22, 6:17*pm, Ron Capik wrote:
ChrisCoasterwrote: On Jul 22, 1:49 pm, Ron Capik wrote: However, "we" can surly measure the 10-band EQ, can "we" not? Later... Ron Capik --- Hide quoted text - - Show quoted text - ________________ Expand on that thought please, Ron? -CC I'm guessing there must be dozens of programs out there that can be used to measure the Q of an equalizer. One way would be run some pink noise through the Media Player then set the EQ to some value and run the same noise through the Media Player. Take FFTs of both and subtract the two FFTs. In a quick experiment using your settings I got something like: 31 *+11.2 62 *+7.5 125 *+3.3 250 *[set as 0.0 reference] 500 *-1.9 1L *-2.0 2k *-2.0 4k *-3.6 8k *+2.0 16k *~-1.0 Later... Ron Capik -- OK, that gives you settings actually quite close to mine - and to what the curve suggests. But where are the Q-measurments, oh intelligent master? -CC |
#80
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substractive vs. additive equalizing.
"ChrisCoaster" wrote in message
On Jul 22, 6:17 pm, Ron Capik wrote: ChrisCoasterwrote: On Jul 22, 1:49 pm, Ron Capik wrote: However, "we" can surly measure the 10-band EQ, can "we" not? Later... Ron Capik --- Hide quoted text - - Show quoted text - ________________ Expand on that thought please, Ron? -CC I'm guessing there must be dozens of programs out there that can be used to measure the Q of an equalizer. One way would be run some pink noise through the Media Player then set the EQ to some value and run the same noise through the Media Player. Take FFTs of both and subtract the two FFTs. In a quick experiment using your settings I got something like: 31 +11.2 62 +7.5 125 +3.3 250 [set as 0.0 reference] 500 -1.9 1L -2.0 2k -2.0 4k -3.6 8k +2.0 16k ~-1.0 Later... Ron Capik -- OK, that gives you settings actually quite close to mine - and to what the curve suggests. But where are the Q-measurments, oh intelligent master? The Q of a peaking-type equalizer can approximated by the difference between the 2 -3 dB points on either side of the peak, divided by the center frequency. |
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