tag:blogger.com,1999:blog-83944022096026570032024-03-11T21:50:39.429-07:00Audio InvestigationsAudio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.comBlogger754125tag:blogger.com,1999:blog-8394402209602657003.post-53238162944751546592024-02-13T21:17:00.000-08:002024-02-13T21:27:26.577-08:00Make Playlist updated<p><a href="https://sourceforge.net/projects/makeplaylist/">https://sourceforge.net/projects/makeplaylist/</a></p><p>You can download the latest version of Make Playlist (which I abbreviate to "mplay") from Source Forge at the above link.</p><p>This is my 3rd update this year which addresses key issues that were bugging me.</p><p>For one thing, I've added in automatic reset for music files the way they play by default. Folders for music albums do not reset automatically (that was a very bad idea, only released in 0.79 then quickly abandoned). Instead, you must exhaust all the music files in folders you have listed, then mplay automatically resets all files just after the last one is read.</p><p>So there is no need for a separate "-reset" operation (unless you want to force one, like the old days, with the new option "-no_reset").</p><p>This required a suprisingly large rewrite. I'd wanted this feature from the beginning, from 2021, but every time I tried to implement it I got bogged down. A large code re-organization was needed (I added a new file, mplay_libs.tcl, which must be included where the other mplay programs are kept). But I also kept the changes to as little as possible for now, which is how I was able to get it done.</p><p>For another, I found the mixing or blending algorithms in both splay and shufflelinks were not very good. I had to rethink the entire blending process after discovering that the relative size of the files you are blending, which is not necessarily known in advance when running a script, is a key factor. So I cam up with 3 parameters which allow a new blending algorithm to adapt to all possible relative size differences.</p><p>As always, even if you're only interested in music files, it's necessary to get ideas from all 3 scripts, MakeMusic, MakeVideos, and MakePix to see different techniques for creating playlists in action. I make a fairly basic script for music playlists, because that's what I need and it's where most people would start, and a very complex one using tricks only a professional music programmer might need (such as mixing playlists) in MakeVideos. mplay is, by design, capable of creating playlists for many kinds of media programs: any that either accept an .m3u playlist file, or that accept a simple folder of files (which is faked with a set of links to the actual files), and the tricks shown might be used in many music playlist creation scenarios.</p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-51861814888412167852024-02-10T08:47:00.000-08:002024-02-10T09:21:22.097-08:00Amplifier on/off<p>In less than one month I got very tired of using the Aragon 8008 BB amplifier without home automation control of the power amplifier as I had with the previously used Hafler 9300. It was wonderful to turn on background music from the moment I get up, and ultimately turn off everything upon going to bed, each with the press of a convenient remote button.</p><p>The inconvenience of having to manually turn the amplifier on and off grew even greater in the last few weeks as I've had my new projector screen up most of the time, and that means I have to crawl underneath the screen to turn the amplifier on and off. (BTW, having a video projection system is one of the best things in the world, and it's especially good to have it backed up with high quality audio like my living room stereo. IMO it's not necessary to have 5.1 channels and that sort of thing. I use mixed down stereo from my Oppo BDP-205. It's better to have wonderful 2 channels than mediocre 5 channels. Wonderful 2 channels is often mind blowing compared to what I've heard before on TV speakers, etc. But a big projection screen makes the most difference of all.)</p><p>To switch the Hafler amplifier I had been using a standard Insteon On/Off module (all my Insteon modules are controlled by my highly programmable <a href="https://www.universal-devices.com/">Universal Devices</a> ISY994i--without which I would never have standardized on Insteon) and the On/Off module had worked fine for several years switching the Hafler several times a day until I burned out the Hafler itself doing some speaker testing on an arcing Acoustat. I've switched to using the Aragon amplifier which I'm thinking now I might like better anyway. (And the arcing speaker is now in my repair "queue"). </p><p>With my ISY994i home controller, I control everything within my home with my own programming (which is endlessly being expanded and improved by me) and I do not rely on any external "cloud" like the "Insteon Cloud" which was abandoned by its <a href="https://arstechnica.com/gadgets/2022/04/insteon-finally-comes-clean-about-its-sudden-smart-home-shutdown/">sadly collapsing manufacturer </a>a few years ago but might be back now under user managed support. I never trusted "clouds" anyway and personally I avoid using "clouds" whenever I can. When I upgrade my Universal Devices controller to a newer version I will be able to handle other home control protocols such as Zwave. But for now, Insteon is all my home control system can support, and Insteon modules are still widely available on eBay and are sometimes being sold off in huge lots for cheap, so it's hardly worth migrating to anything else yet. The biggest advantage Insteon had was the wide array of different kinds of modules that work together, and low prices (compared to Creston and the like). Controlled with a Universal Devices home controller, you can make a collection of Insteon devices do just about anything. I now have about 40 Insteon devices automating my home, and they either work perfectly or I change the programming until they do (adding more repeats or waits if needed). One advantage of this kind of <i style="font-weight: bold;">barely working</i> system is that it's fairly unhackable. Nobody outside my home could reliably gain control of anything <i style="font-weight: bold;">except</i> my Universal Devices controller, which itself is fairly secure and also fairly obscure. However I'm willing to give up that security advantage when I migrate to newer devices which may be wifi controlled, that's also secure enough IMO.</p><p>I knew from both the ratings and previous experience that it was not good to use Insteon modules with the Aragon amp. Last time I tried using an Insteon module with the Aragon, the Insteon module started acting very funky within a few days, turning it self on and off randomly. I had a similar experience with an Insteon module switching my 1000 watt amateur radio power supply.</p><p>And the ratings of the Insteon (15A resistive, 3/4 HP) are not encouraging because the Aragon is anything but a resistive load, especially when it's being turned on and off. With it's two 1100 VA transformers, the startup load of the Aragon quite possibly exceeds 15A for a brief instant upon turn on. My calculation suggests around 18 amps (2200VA / 120V).</p><p>There aren't many automatic AC switches that advertise handling more than 15A. Furman makes a 20A trigger switched dual output (CN-20MP) but it's extremely pricey ($394) and it uses an actual 20A plug with the horizontal blade that doesn't fit into standard 15A outlets. I'd need to get a new outlet, which raises three additional issues: (1) cost for installation by electricians, (2) I detest 20A outlets because in my experience the greater number of "angles" causes more arcing when you plug into them, and (3) the amplifier circuit currently uses my favorite Pass & Seymour 15A industrial grade wall outlets which grip tighter than anything else--except, and often unfortunately, Insteon On/Off module outlets, you have to pull on plugs so hard to get them out it can break the module--Insteon should have made stronger cases for their modules--I've broken about 5 of them--and I had never physically broken a single X10 module prior to migrating to Insteon, the problem was that X10 just no longer worked in a home filled with switching power supplies.</p><p>Furman also makes a 15A model which gets around the 20A plug issues (CN-15MP), but would it be robust enough? (My guess is probably, but it also is very expensive, just a bit less than the 20A model, it lists for $350 though sometimes is sold for around $250). And there is another company (Lowell) which makes 15A and 20A switchers which are much like the Furman units except 40% lower in price, which is still a lot, and the 20A version has the same plug issue.</p><p>Then I realized I already had in my junk pile an Xantech AC-1, nominally rated at 15A. Checking out it's detailed ratings, I see now it can handle up to 30A peak inrush current. Well that's exactly what I need.</p><p>(The Xantech is considerably more expensive than an Insteon On/Off Module, and is much simpler, having only a DC trigger input to switch the output. But it's a very robust looking metal box. Sadly these are no longer made, but can still be found. Before they were discontinued, they were priced about the same as the 15A Lowell units and less than the Furman ones. There are also AC switchers made by Niles which look about the same as the Xantech.)</p><p>So I hooked it up. To generate the 12V DC trigger voltage, I used a Sony 12V AC adapter (rated at 300mA, an old unit originally intended for charging portable telephones) which itself is switched on and off with an Insteon On/Off module. That gave a very nice 1 second delay from turn-off to amplifier shutdown, indicating some degree of regulation in the AC adapter and thus an ability to ride out tiny power glitches. (Unregulated AC adapters can sometimes keep the rated voltage for minutes, which is inconvenient if you are trying to use them to generate DC trigger voltages).</p><p>It's working great, and I expect it will last a long time since it seems to be used within ratings.</p><p>*****</p><p>Without some kind of automatic switching it's tedious to try to keep background music running most of the time, and I am more and more believing that is a good idea.</p><p>I think it is very good to have background music running nearly all of the time (except when you are watching movies or doing "serious listening"). As long as the music doesn't include words it rarely interferes with concentration either.</p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-84860211353842329392024-01-16T09:47:00.000-08:002024-01-19T07:52:06.894-08:00makePlaylist (mplay) version 0.79 released<p>I've talked about it, posted a video of it, and now it's available for download.</p><p><a href="https://sourceforge.net/projects/makeplaylist">https://sourceforge.net/projects/makeplaylist</a></p><p>This allows you to automatically generate random playlists. Now you can preface or mix in newer files in various ways, combine folders with weighting factors, and mix in favorites. It makes for very infinite and very interesting scripting possibilities, with examples given. One program included figures out how much of the last playlist you actually played and allows you to reset the history for those files that weren't actually played. </p><p>Works immediately on older Macs* or Linux, on other systems Tcl must be installed first as all the programs are written in Tcl. The example scripts are written in (linux/mac) bash.</p><p>(*Since I use an old Mac running 10.13.6 myself, I'm not sure about the status of Tcl in newer Mac releases. I think it may not have been included by default with M1 macs, but not sure. It's still available for nearly everything because Tcl is a very lightweight programming language, lightweight but powerful. But you may need to download and install it first, and I have no recent experience with that myself because it was always included on Macs I have used.)</p><p>UPDATE: since I first made this post a few days ago, mplay has already been updated to version 0.79 which fixes a lot of things that had been bugging me</p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com1tag:blogger.com,1999:blog-8394402209602657003.post-72673972831820692102024-01-14T08:50:00.000-08:002024-01-14T09:40:28.836-08:00New EQ, Day 3<p> After messing with the "Background" subwoofer EQ (again, last time was a few months ago) in the right channel, which I had been re-EQ-ing because of the new speaker, but the subwoofer hadn't actually change...</p><p>I decided I first needed to test and make similar changes on the left. Measurements at the doorway and the table showed overly bulging response at 31.5 Hz. It looked better cut back to zero as with the right channel.</p><p>But listening to Grouse, We Want To Be Loved, I decided this all made the deep bass line, which reaches down to 31.5 Hz, a bit less easy to follow. So I restored boosting at 31.5 Hz in both channels in background to about +2dB. (In the Serious Listening EQ, it's boosted 5dB because of suckout at listening position.)</p><p>Also, I could not confirm with pink noise any serious issue at 20 Hz, so I decided to reduce the cut there from -4dB to -2dB.</p><p>Current settings:</p><p>Left (last one adjusted)</p><p>20 Hz -2dB</p><p>25 Hz +2dB</p><p>31.5 Hz +1.5 db</p><p>Right</p><p>20 Hz -2dB</p><p>25 Hz 0dB</p><p>31.5 Hz +1.5dB</p><p>I think I might make both channels more like the Left settings, mainly boosting 25 Hz at +2. Otherwise, the response IS falling at 25 Hz relative to 31.5. Even with that +2 boost it is falling at 25 and 28 Hz relative to 31.5, but it is falling only ever so slightly at 28, which is a good improvement.</p><p>But it was too late to go any further. Actually, once again, for best results I need to sweep this with an oscillator, or use a LF tuned keyboard (as I have done before).</p><p>Just like a down sloping high end, required because of the higher side reflectivity of small rooms, a slightly tilted up bass is generally required, it's just a question of how much and exactly where. Finer adjustment of the hinge points may mean less boosting is needed to retain musical coherency. There is probably some perceptual compensation for room gain, and the total room system output should invert that compensation.</p><p>Now, morning after, taking a look at the graphs confirms my memory and shows some other interesting things. First, what it looked like with the old boosting (which I think was +5db, maybe only +3dB), in the kitchen doorway:</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjS2nXozeeCbAJnxApEZ43lO7-s__IPr-LLcuxWbv3l8tUPQJj0GU4l_3vTICkXEtZQ3_lIH3o3WKzbs-TSB9XmOEIqdNFvZo1F0sHVmcK2jhac71emjRrThDVcZq1pcjlKMoZrqedRXHMsIL2VnwXS3vYwm8X1eqvCJ8OCPSNSE4QC7rjcegqBcKXwZsQ/s2208/IMG_1640.PNG" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjS2nXozeeCbAJnxApEZ43lO7-s__IPr-LLcuxWbv3l8tUPQJj0GU4l_3vTICkXEtZQ3_lIH3o3WKzbs-TSB9XmOEIqdNFvZo1F0sHVmcK2jhac71emjRrThDVcZq1pcjlKMoZrqedRXHMsIL2VnwXS3vYwm8X1eqvCJ8OCPSNSE4QC7rjcegqBcKXwZsQ/w400-h225/IMG_1640.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Left 2+2 with +5dB deep boost at kitchen doorway</td></tr></tbody></table><p>Reducing the 31.5 Hz boost to 0dB but the 25 Hz boost at +2dB, it looked like a much flatter high plateau in the deepest bass (though I later re-raised 3.15 part way to +1.5 for musical reasons, not measured).</p><div class="separator" style="clear: both; text-align: center;"><br /></div><br /><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg03ls7CVFCgOlnAH9oIQUBi-fIkQ2-Pb-ZAxVPMAf-J_QLKZhwXN2aC32vbO4OH1koTx5xJI7f-eOqKYVVZ6bMooIS0RBEGHXsrPCrVZ2GQhiZw98btuh3jLaRucYnn79jfKFKQgUtKiPGP8EpO-M7zTWoR1wKOSLm_3riGKhCJSb49gYtafaqTnoQYrQ/s2208/IMG_1641.PNG" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg03ls7CVFCgOlnAH9oIQUBi-fIkQ2-Pb-ZAxVPMAf-J_QLKZhwXN2aC32vbO4OH1koTx5xJI7f-eOqKYVVZ6bMooIS0RBEGHXsrPCrVZ2GQhiZw98btuh3jLaRucYnn79jfKFKQgUtKiPGP8EpO-M7zTWoR1wKOSLm_3riGKhCJSb49gYtafaqTnoQYrQ/w400-h225/IMG_1641.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">+2dB at 25 Hz, 0dB at 31.5</td></tr></tbody></table><br /><div>One thing overlooked here last night was the incredible broad depression from about 80 Hz to 300 Hz.</div><div><br /></div><div>It's almost as if (and I have tried that at times) attempting to compensate for the lack of 125 Hz in the right channel by giving it some extra oomph in the other, which only "partly" works.</div><div><br /></div><div>That looks like the next thing which needs to be investigated. What the hell is going on at 125 Hz.?</div><div><br /></div><div>With the background EQ, in the doorway, it's nice looking now in the right channel but part of a long depression in the left.</div><div><br /></div>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-39721392146660577212024-01-13T10:08:00.000-08:002024-01-13T18:06:53.124-08:00EQ'ing new speaker, part 2<p> In the last post, I purported to show how removing most of the boost I had given the Acoustats a few years ago, +3dB and 1/3 octave at 1013 and 853 Hz, fixed the balance problem between new and old speakers. It seems like the newer speaker doesn't need that boost, perhaps because of tighter diaphragm. My initial approach was to cut those two boosts by 1-2dB to +1.5dB at 1013 and +2 at 853. But there was still a troubling difference around 500Hz, the newer speaker having a suck out at that point. At first I thought it was because the 853 Hz boost wasn't wide enough.</p><p>Fixing that was very problematical. Boosting 500 Hz, with +3dB in a 1/6 octave GEQ boost, produces a peak around 600Hz that's even more alarming than the depression it replaced.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhjf89ozxN-abnQejpZbRhxSIWZ0SmbWPvPyrszYu1d4MtWzKWF1_HIRcVU32t2w2kXPZfRcTydXe-Sw1fFc_7j83uNhgPnKWye7zgKA1aVbGM7iYURb1HZyW1XLiPd1koeZFQfmfd8AVa9SBujIKe6qed_0TsbLK_yaaQY9DX0LMnBCBzJwi91fi4Ftmg/s2208/IMG_1623.PNG" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhjf89ozxN-abnQejpZbRhxSIWZ0SmbWPvPyrszYu1d4MtWzKWF1_HIRcVU32t2w2kXPZfRcTydXe-Sw1fFc_7j83uNhgPnKWye7zgKA1aVbGM7iYURb1HZyW1XLiPd1koeZFQfmfd8AVa9SBujIKe6qed_0TsbLK_yaaQY9DX0LMnBCBzJwi91fi4Ftmg/w400-h225/IMG_1623.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;"></td></tr></tbody></table><br /><p>So the issue is we need to push 500 Hz up, but 600 Hz down. It looked to me like the down part was actually the most important, and I found a 1/6 octave cut at 563 Hz worked pretty well. After doing that, the peak seemed to move up to 800 Hz so it was clear I needed to remove any boosting from 853 Hz at all, so that as of January 13 at 4am the newer speaker EQ now looks like this:</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiz7N0KvVWK9Wftd0BKA9dYKrdxXeVYGn16LtgrjPAB-xP8zDge8mixuP5jswn4Z5uH8ZuKDDiBM6MqdlPzIz0YArch13GNpNB9NcS-qC3pTfWJF5q7JXogZ1Kqov9hKe4QdBqHqOzx71YvbTm7yDme7CyUnStTVA4wW-lXfMPG-KS4d8URNfKD5FaaleA/s4032/IMG_1627.jpg" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="3024" data-original-width="4032" height="300" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiz7N0KvVWK9Wftd0BKA9dYKrdxXeVYGn16LtgrjPAB-xP8zDge8mixuP5jswn4Z5uH8ZuKDDiBM6MqdlPzIz0YArch13GNpNB9NcS-qC3pTfWJF5q7JXogZ1Kqov9hKe4QdBqHqOzx71YvbTm7yDme7CyUnStTVA4wW-lXfMPG-KS4d8URNfKD5FaaleA/w400-h300/IMG_1627.jpg" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;"></td></tr></tbody></table><br /><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjkdaTjrl2naGWWpZoVRO-zvRyzi-oWpLimuZXNwjK7edWejkyC_m-xbpOvnlcNxbkcCUPY9daNWhYpUYtE5Is_phES9tEN0PIYgTcJ4Fr_aq9rEFa72-5nczy2PP6mc_qs0v9_gm15_cexEpd4JCLx6JrEBRUE-v1yDsfq_G4ZP3U9HV-6kE-Bkhi9iJs/s4032/IMG_1628.jpg" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="3024" data-original-width="4032" height="300" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjkdaTjrl2naGWWpZoVRO-zvRyzi-oWpLimuZXNwjK7edWejkyC_m-xbpOvnlcNxbkcCUPY9daNWhYpUYtE5Is_phES9tEN0PIYgTcJ4Fr_aq9rEFa72-5nczy2PP6mc_qs0v9_gm15_cexEpd4JCLx6JrEBRUE-v1yDsfq_G4ZP3U9HV-6kE-Bkhi9iJs/w400-h300/IMG_1628.jpg" width="400" /></a></div><br /><p>Which measures (with the iPhone 8s held about 1 foot from the back of the chair, something I'm now trying to be more consistent with)</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEj9P6CknoSx6mvVWPShpFd2LFBxw-kIlQqWp8a8ygxEa6L9ICXuF3aWsTUQeSBtb4VUnGdme-yX6sYx06XSNPOdg3bZdp0txaUnsHNLedzCDamM2d3hyphenhyphenGKdbTpL-7rSaMBdcdDjQY0JvwcTQl9gJj0-3V_K4Qf4Td5zq31GVvSTtcdfzN7KVyJFjrnTLdI/s2208/IMG_1626.PNG" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEj9P6CknoSx6mvVWPShpFd2LFBxw-kIlQqWp8a8ygxEa6L9ICXuF3aWsTUQeSBtb4VUnGdme-yX6sYx06XSNPOdg3bZdp0txaUnsHNLedzCDamM2d3hyphenhyphenGKdbTpL-7rSaMBdcdDjQY0JvwcTQl9gJj0-3V_K4Qf4Td5zq31GVvSTtcdfzN7KVyJFjrnTLdI/w400-h225/IMG_1626.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">New Left 2+2 with EQ updates above</td></tr></tbody></table><br /><div>Even the Analyzer app thinks I've nailed the EQ at 1kHz, not that I trust such things. But it's reasonably uniform from about 200 Hz to 2kHz where it starts an intended downturn pretty much as intended.<br /><p>I'd still be interested in pushing down the region from about 600 to 800 Hz slightly, I don't believe that should be louder than 1kHz. And possibly broadening the EQ right around 1kHz which is slightly depressed. But I've already gone beyond the level of fine tuning which should be done with RTA. I need to switch over to doing oscillator sweeps so I nail the critical frequencies where change is actually needed. RTA is simply too approximate, you can't really see what is going on, though it's useful for getting the big picture.</p><p>But I think this is already pretty darn good and far better matching the right speaker than when I started.</p><p>I also took a brief look at the funky dips at 40 and 70 Hz. Those are because I have been using the "Background listening" Bass (subwoofer) EQ for these measurements which has a suck-out at the listening position but sounds better (less boomy) everywhere else in the house. </p><p>Measurements roughly confirmed this. I decided for now upon two different places to measure the background EQ, the doorway to the kitchen (which is about equivalent to the couch) and the kitchen table.</p><p>At the doorway, the response rises with decreasing frequency below 180 Hz, peaking around 38 Hz. I've needed to roll off 20 Hz in the background response to prevent the whole house mode that occurs around 20 Hz, which I can feel, for example, at the sliding glass door (which perhaps I should also measure...).</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgZaflMaUIag9cCxO14fPbajlFvbsvbCigjUKnfVNPaGuiQ4seMZTgDYcxcMGzgZG-A_Wef8N7aCbMnkvl9_BXpiQyGKoUYhI6gSde849KCNLtjTp249PahXjqyV3Jsl5GoBLq69HmCGw100lHsP-AmkiKPZazZUCTgyr2XFZHBbkNXjKxx1KcAcwZN43Y/s2208/IMG_1629.PNG" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgZaflMaUIag9cCxO14fPbajlFvbsvbCigjUKnfVNPaGuiQ4seMZTgDYcxcMGzgZG-A_Wef8N7aCbMnkvl9_BXpiQyGKoUYhI6gSde849KCNLtjTp249PahXjqyV3Jsl5GoBLq69HmCGw100lHsP-AmkiKPZazZUCTgyr2XFZHBbkNXjKxx1KcAcwZN43Y/w400-h225/IMG_1629.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Kitchen Doorway with pre-existing 25-31.5 Hz boost</td></tr></tbody></table><br /><p>This is pleasant actually, gives a kind of loudness compensation for background listening, and requires little change from the 'flat' response at the listing position with the Listening Position EQ setting.</p><p>But I felt the rise including 31.5 Hz was just too much. </p><p>It was pointless to boost 31.5 and 25 Hz at all, since that boosting was actually causing them to stand above everything else so much. So I dialed those boost back to zero in the right channel of the Background EQ. Even with 0dB boost at 31.5kHz, there is still a bulge there, dropping lower (by design) at 25 and 20 Hz.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgYRSKDZ8ptriFqXu0VLAFV8Yye5qEe5iGuj13g7gFdWtM-rUnVgnAj3OQfEuHp1mORZZHRSj4WPNb0FIPUwThBurPAo8f_xGsBZ9QHjfdXrKTyfGVQ2UUq10zWwKjRTUajggQbVgFwwXAFLP0wpvlMwJKGo044JLZYYiK9TLxAgWKUcvHdCpy2AMF2zFE/s2208/IMG_1632.PNG" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgYRSKDZ8ptriFqXu0VLAFV8Yye5qEe5iGuj13g7gFdWtM-rUnVgnAj3OQfEuHp1mORZZHRSj4WPNb0FIPUwThBurPAo8f_xGsBZ9QHjfdXrKTyfGVQ2UUq10zWwKjRTUajggQbVgFwwXAFLP0wpvlMwJKGo044JLZYYiK9TLxAgWKUcvHdCpy2AMF2zFE/w400-h225/IMG_1632.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Kitchen Doorway with no boost 21-31.5 Hz</td></tr></tbody></table><br /><p>That was fine at the dining room table position too, showing the slightest peaking around 31.5 Hz even with no boosting. But the dining room table shows a huge notch at 56 Hz, a peak at 125, and another smaller notch around 180 Hz which need further exploration.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjQwvW2Fapeth4rgDaNaPgt3naUhd9C47XV7JAdDsPEeGTOR3CaOPEPQRpPoLn3L29aYzlgw6zQJQGF7yPIUL9xkTugosU9JUfkiHC0lxWXXFsib2yVM7_tRm_so0dWTy5WOtuPQLp99prf7lDJLUjSGIpOV6382eTXoeFTyVCxAZ_SkKkskmOB8PjzwAA/s2208/IMG_1635.PNG" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjQwvW2Fapeth4rgDaNaPgt3naUhd9C47XV7JAdDsPEeGTOR3CaOPEPQRpPoLn3L29aYzlgw6zQJQGF7yPIUL9xkTugosU9JUfkiHC0lxWXXFsib2yVM7_tRm_so0dWTy5WOtuPQLp99prf7lDJLUjSGIpOV6382eTXoeFTyVCxAZ_SkKkskmOB8PjzwAA/w400-h225/IMG_1635.PNG" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Dining Room Table</td></tr></tbody></table><br /><p>I do most background listening from here, so it should be better than this. And certainly the recent removal of boosting 31.5 Hz is only to the good as well. 20 Hz is doing fine suggesting the cut there is of no harm here. Save for 56, 125, 180, and 720 or so other, Background EQ'ing looks all to the good.</p></div>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-15613295366710586462024-01-11T11:49:00.000-08:002024-01-11T16:08:18.809-08:00The White and Black 2+2's<p>My blog photo now represents the new living room system reality.</p><p>I'm using one white Acoustat 2+2 from my original pair of 2+2's that I bought in 2019.</p><p>It's mate had an issue I was long aware of, but mostly thought I'd solved. I don't believe I had.</p><p>On some recordings, there was a terrible resonance with some combination of bass frequencies. I first heard it on the Grouse album 'We Want to be Loved' with the track 'Suicide kills me'.</p><p>I pretty much eliminated the issue with my 125 Hz 8th order crossover, which also resolved what seemed to be other issues resulting from rear-reflection and Acoustat resonances.</p><p>When testing the new black 2+2 I just bought in December 2023 before purchase, I noticed a similar issue with 'You Know Too Much About Flying Saucers' on Hello Waveforms by William Orbit, on one of the new speakers.</p><p>Well that same new speaker had a continuous but almost inaudible arcing problem I could only hear with my ear right up to the speaker. I could not smell anything and believed the ozone emanation was trivial, not unlike with a small motor perhaps. So I figured it was fine in the well ventilated but rarely used living room, but perhaps not so fine in the bedroom.</p><p>But rapid ABX switching of the speakers, and/or leaving the speakers connected to the running amp while plugging it in, or some other experimental mistake, I blew out my Hafler 9300 in one channel.</p><p>I don't really know that the arcing speaker was at fault. Perhaps the ABX switching itself is unsafe (but it never hurt the Hafler or Aragon before, only the Krell). But the coincidence of having an amp fail within a month after connecting to this arcing speaker makes me wonder.</p><p>(Also, the guy I bought it from, I well knew was having endless weird problems related to the 2+2's if not other things.)</p><p>Now perhaps the problem only occurs when the speaker was being charged up (though once again, that was never a problem before). I'm resolved now to plug and unplug acoustats from AC power ONLY WITH THE SPEAKER LEADS DISCONNECTED. You want the HV to stabilize before connecting an amp to it, especially if there is an occasional shorting issue.</p><p>But that kind of thing can't be prevented, for example, if there is a temporary power failure, when the power is restored once again the interfaces with be charging, mostly likely with the speaker leads connected.</p><p>So, I consider the second black 2+2 with arcing to be unsuitable for use except with "expendable" amplifiers. Perhaps things like one of my two HCA-1000A's. Not that special. AND MOST CERTAINLY NOT WITH MY ARAGON 8008 BB !!! WHICH HAS UNOBTANIUM TRANSISTORS!</p><p>And then I decided also that on complex music, the arcing 2+2, even if the arcing wasn't audible, still made for it sounding fatiguing after awhile, which I determined might well be traced to the arcing causing unstable membrane voltage.</p><p>Now the faulty black 2+2 has been moved off to my bedroom so it can be easily accessed for repairs. I've decided it's the first 2+2 I want to fix, though it might be harder to fix. I want to end up with two of the black ones, because I think they are better somehow, tighter membranes perhaps. They were much later production, most likely made in Arizona instead of Florida, though not as far as I can tell using the later improved coating. Perhaps other construction details had been improved. Or perhaps they were just less used.</p><p>I can now mostly rule out the issue being a difference in interfaces, because I have been using one of my old Red Medallion interfaces with the new 2+2's, and my white 2+2's had a C mod done on them by a well regarded Acoustat refurbisher.</p><p>But it is true, though I often forget and doubt it means much, that the white 2+2 I am still using still has the non-Medallion transformers. I plan to swap it with the other large-interface I have which has the a recently made Medallion transformer I installed in my first attempt to repair it, before I noticed that the issue was that the 'crossover' board had been connected incorrectly. I then didn't bother to take out the Medallion transformer, I figured it was better anyway, even though there was now a mismatch with Medallion HF on one side and not the other, which I could never tell that any difference resulted from that.</p><p>THEN, I will have Medallion HF on both sides. There will still be a lack of Medallion LF on one side, but that is said to be even less important for the sound. It seems one of the primary purposes of the Medallion upgrade was to prevent transformer failure with improved insulation. It may have also resulted in tighter coupling, and the improved sound was hyped, but never taken by anyone as a kind of 'night and day' difference, the old non-Medallion transformers sounded great too, and just doing the "C" mod alone is supposed to make a far bigger sonic difference than the Medallion transformers, though they're great if you got 'em.</p><p>Anyway, even right now there's still a difference between my good white and black 2+2's, despite both having "C" mod, and probably not just because one has Medallion's and the other not.</p><p>Now in these response I'm going to show, ignore the bass below 125 Hz, that's may look bad because of the new 'background' EQ that was operating in the bass, which produces a suckout around 45 Hz at the listening position.</p><p>(Update, I've discovered an error in the sub 1kHz eq, the left and right don't match in the GEQ section, it looks as if I've simply failed to dial in the Right channel, which is possible, back after my chairside EQ died. It's unlikely the two channels previously needed lots of eq in the left channel and none in the right, in fact the right white speaker, now removed, was very problematic wrt EQ. That explains some of the weirdness below 800 Hz in the right channel.)</p><p>I've always turned the HF level on the white acoustats to about 12 o clock, when I now understand 'flat' is supposed to be 3 oclock. So what would 12 o'clock be? I'd guess around -2. I previously thought 12 o'clock sounded best and was supposed to be flattest. (I'm applying a vast amount of HF downwards EQ above 2kHz too, without that and the HF adjustment it would be about flat to 17kHz.) I have a smallish room which probably requires downward HF EQ. It's not a particularly lively room, just about balanced perfectly for conversation.</p><p>To match it I've dialed the interface on the black units down to the lowest HF level the compact interface permits. It's a very limited range of the 16 ohm resistor. I'm guessing that level is around 2dB also. And lo with these settings the frequencies 10kHz and above are fairly well matched.)</p><p>Also note that EQ from before IS being applied, as well as the 8th-order phase corrected crossover at 17kHz for my supertweeters when I no longer have supertweeters, but I haven't gotten around to taking it out and think it might improve the sound anyway because of the phase correction.</p><p>That EQ was dialed in by ear with the original white 2+2's, which seemed to need a 3dB (!) boost at 1kHz to sound good, also at 850 Hz at 3dB. Otherwise, the midrange sounded weak, perhaps not just with regards to it's immediate surroundings. Prior to EQ 2kHz measured higher than 1kHz, even with 2kHz dialed down.</p><p>Well, it seems the 1kHz boost is simply not needed with the good black 2+2.</p><p>Taking that out, or even just cutting it back to 1dB was sufficient to fix the 'balance' problem I had fixed by ear the day before by cutting the left channel 2dB in my Tact. (But if I dialed it back to 0dB, I'd then need to cut the right side to get it to balance, so I'm now using 1dB boost at 1kHz for the black 2+2, pending further investigations.)</p><p>Now, it would be hard to believe that the Acoustats were originally voiced with a 3dB suckout at 1kHz. I think that correction may have been needed because of membrane aging on the white 2+2's.</p><p>The result is that images are nicely behind the speakers as they should be, much clearer images than I've ever had before thanks to the speaker swap, removing the supertweeter towers, and possibly other recent changes.</p><p>Anyway, prior to adjusting the 1kHz EQ down in the right channel, here is how the white and black speakers now on the left and right sides respectively measured:</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi0gkyscfmKzVDwFaucNMwMYhh8QIyEkXOwZmVe1QVApBTA16C9ivh_iXUtp0NEY-TLnU1c_wMjJon-XyUdB6L7ZOFCiUnMFcJdN67KZJ69ABDdVkYXquSHmbgbdh-y4MSDsASujwXZ7Q_CMAf_RxYWky2MgZ_CJE0xy_vaMw47VY_svcoXnF5McygWFA0/s2208/IMG_1611.jpg" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi0gkyscfmKzVDwFaucNMwMYhh8QIyEkXOwZmVe1QVApBTA16C9ivh_iXUtp0NEY-TLnU1c_wMjJon-XyUdB6L7ZOFCiUnMFcJdN67KZJ69ABDdVkYXquSHmbgbdh-y4MSDsASujwXZ7Q_CMAf_RxYWky2MgZ_CJE0xy_vaMw47VY_svcoXnF5McygWFA0/w400-h225/IMG_1611.jpg" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Left Channel with White 2+2</td></tr></tbody></table><br /><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhzL8s2_WMOo2M0ZgMHi3GcPBcicsKVd-0yz76yT7m7ir9DgGbZt-AIPd-Ec0dwY0HWpKMhiz4LECfF52KTihKKKy1ttegtPdefosbRDl8a19LMUTEgyaNzxFbYZ4UGHIamF74IrNghVil9jn1TJoM06WHczL72m-96J1sjpWEfH-1tJH8M08haPK96zi4/s2208/IMG_1612.jpg" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhzL8s2_WMOo2M0ZgMHi3GcPBcicsKVd-0yz76yT7m7ir9DgGbZt-AIPd-Ec0dwY0HWpKMhiz4LECfF52KTihKKKy1ttegtPdefosbRDl8a19LMUTEgyaNzxFbYZ4UGHIamF74IrNghVil9jn1TJoM06WHczL72m-96J1sjpWEfH-1tJH8M08haPK96zi4/w400-h225/IMG_1612.jpg" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Right Channel with Black 2+2</td></tr></tbody></table><br />You might notice as I just did that the actual level of the 1kHz bulge in the right channel doesn't actually reach quite as high as the narrower bulge in the right channel. True, but the above measurements are WITH a -2dB reduction in the right channel level, as I needed to achieve left to right balance.<div><br /></div><div>Don't pay much attention to irregularities above 2kHz as the measurement device is only being held in approximately the same position each time.</div><div><br /></div><div>After taming the midrange bulge by tamping down the +3dB EQ at 1kHz to +0dB, it's looking a little different:</div><div><br /></div><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgDukZ-5KW0sfcEt5v5UgHLEVW-7aOnktID6yhYBDzdby9Q5IO5NEfQqZdvRIW2EXlXRz9zbi_Sbk4W2mShwIReqInAEcquK03SqG97umgGDnNozRwjaI5OF8m4P2s0j69AbTrz6lDQGC9Q9BPVd6uRX8NO5jpZ9b2yYtFfmEQDbDmuKRdf1UuOgPw2pj0/s2208/IMG_1613.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgDukZ-5KW0sfcEt5v5UgHLEVW-7aOnktID6yhYBDzdby9Q5IO5NEfQqZdvRIW2EXlXRz9zbi_Sbk4W2mShwIReqInAEcquK03SqG97umgGDnNozRwjaI5OF8m4P2s0j69AbTrz6lDQGC9Q9BPVd6uRX8NO5jpZ9b2yYtFfmEQDbDmuKRdf1UuOgPw2pj0/w400-h225/IMG_1613.jpg" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Black 2+2 with 0dB boost at 1kHz</td></tr></tbody></table><br /><div>But now 1kHz is lower than 800kHz, so I decided to up 1kHz to +1dB and cut the +3dB boost at 800Hz to 1.5dB, resulting in this:</div><div><br /></div><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjd9m5L7E4wWDKz4oJsv2abtLtRDn5rdRu2YbL4YjzUnmgfjST85IjQz8P7CK79Zec3ZGnSnQhZmweTtPms2sALWGXjxEhTGqUx_GelH4j5H835ZGfkyT5hT55iWy3Gtp2aA1KfA2RE_HslEQSe0ddhWYF_WtrkvTg0cjoKynmGTnYHvIG57Q710Lw4h4U/s2208/IMG_1614.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1242" data-original-width="2208" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjd9m5L7E4wWDKz4oJsv2abtLtRDn5rdRu2YbL4YjzUnmgfjST85IjQz8P7CK79Zec3ZGnSnQhZmweTtPms2sALWGXjxEhTGqUx_GelH4j5H835ZGfkyT5hT55iWy3Gtp2aA1KfA2RE_HslEQSe0ddhWYF_WtrkvTg0cjoKynmGTnYHvIG57Q710Lw4h4U/w400-h225/IMG_1614.jpg" width="400" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Black 2+2 with less EQ at both 800 and 1000 Hz</td></tr></tbody></table><br /><div>That's the best yet in terms of flatness through the 700-1500 Hz range, and with these changes it no longer needs a -2dB overall level reduction in this channel. Apparently the image shift was caused just by the boosted midrange.</div><div><br /></div><div>However, noting that there's still a measured boost in the left channel, I'm wondering whether THAT is correct too. But exploring that will require subjective testing, the way I arrived at the need for a boost there in the first place.</div><div><br /></div><div>Sound level measurements with RTA are never definitive, only a guide to what's probably right.</div><div><br /></div><div>But it occurs to me now that perhaps I boosted the midrange mostly because of issues with the right channel white 2+2 that has now been replaced. It might have had quite a suckout in the midrange which I was compensating for by boosting both channels.</div><div><br /></div><div>Meanwhile, I should also address the sub 1000 Hz issues that are apparent in these graphs.</div><div><br /></div><div><div><br /><p><br /></p><p><br /></p><p><br /></p></div></div>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-90209233463756213272023-12-31T10:28:00.000-08:002023-12-31T10:40:36.934-08:00Electrostatic speakers and Bass EQ<p>This is two separate topics I was thinking about yesterday.</p><p>It seems like the good channel of my newest Acoustat 2+2's sounds louder than my older 2+2's, and this is true regardless of interface used. Substituting a plain vanilla C mod interface makes it sound more like my older C modded 2+2, but still not identical.</p><p>(Note: I have not tested speakers in identical or equivalent positions yet. The speakers are very hard to move and I may not even get around to such rigor until I've removed the two 'bad' speakers from the room, one from the older pair and one from the newer pair. Then I will have a pair of speakers, one old and one new, in equivalent positions ("left" and "right") though not actually identical, and I will be forced to level match and EQ match them for good stereo, and only THEN will I be able to state with better authority how the old and new speakers differ.)</p><p>I'm thinking electrostatics can age in various ways:</p><p>1) The HV can droop. I wouldn't have thought this to be a problem because all my interfaces (except my original 1+1's which may be the best sounding of all) have been refurbished by knowledgeable people.</p><p>2) The membrane can stretch. I was previously thinking this would only be a problem if there were actual 'slapping,' otherwise the panel would just move to where it was supposed to regardless of stretching. BUT that was wrong, in fact the resistance of the panel to movement is a critical part of its operation. Stretched membranes can cause bad resonances and intermodulation.</p><p>3) The insulation on the stators (if applicable, as with Acoustats) can weaken and break. In serious cases you get periodic arcing (as with the bad new speaker) and then the speaker develops level dependent distortion. But there could be distortion from mere 'weak' spots that weren't arcing as such.</p><p>Every panel is also going to vary from every other panel in minor construction details that result in slight resonances at different slightly different points. (Though, by and large, I'd expect most brand new speakers to be almost the same, but diverging with aging.)</p><p>ALL speaker units differ, even if made by the same manufacturer on the same day in sequential serial numbers. However, you'd expect that the closer you get to being on the same day in sequential numbers, the closer they are likely to be. So units made years apart in different factories are likely to be different than their mates made at the same time.</p><p>I'm thinking because of how large electrostatic speakers effectively have been hand-made they probably differ more in their tiny resonances and such more than factory made dynamic drivers. OTOH the electrostatics generally have (less so nowadays) less distortion and resonances to begin with.</p><p>Funny I've never read anything from Linkwitz about using electrostatic speakers. He made dipolar speakers using dynamic drivers. So I don't know what he thought about electrostats. But his methodology for picking out the better drivers was ultimately quite simple. You have to listen to them. Only then can you tell if the particular set of defects every speaker has is problematic or not.</p><p>For rigorous testing, playing one speaker by itself on a mono signal is the most revealing. (Linkwitz also may have recommended female voice, which I hardly ever listen to.)</p><p>And so I've been doing, with the good (non-arcing) unit for a few days now. Playing just that one speaker. And this convinces me that although it may sound different from my older pair, it's still very good if not significantly better. It was the arcing speaker that made the new pair fatiguing.</p><p>The new good speaker just seems to have remarkable clarity and musicality. I'm afraid I might have to 'dumb it down' with EQ to make it match the other one, but it will probably still sound better, more headroom, etc.</p><p>I'm thinking this may be because it has a tighter membrane. Or maybe other construction details varied, as this newer speaker likely was made in a different factory. However it doesn't have the '5 wire' connection that would indicate the new kind improved wire insulation, if I'm understanding that correctly.</p><p>*****</p><p>I've just made another tweak to my 'background' EQ. (I lost my original 'background' EQ when my chairside EQ unit died, so perhaps I'm just 'recovering' what I had previously done, which I had to guess at first.)</p><p>This proved necessary when I was listening to some bass records including 'Bass Erotica' which was almost unlistenable before making them, and afterwards could (for the first time) be cranked way up and enjoyed much more, with the bass lines becoming easy to follow.</p><p>I'm now cutting 50 Hz by 5dB, and the flanking frequencies of 40 and 63 by 4.5.</p><p>I believe is needed because of the room modes around 45 and 52 Hz. In the center listening position, these modes don't get amplified (in fact, they get attenuated) so I don't cut them in the primary EQ (as when I last did it, I was running the oscillator while sitting in the listening chair..previously I had often done tuning while listening near the DEQ, which is more like the room boundary). So they have to be cut in the background EQ now.</p><p>Which brings to mind how many EQ systems deliberately have multiple presets. <b> I am now thinking it is a requirement that if you use EQ at all, and if you listen to background music on your main system at all, you must have separate EQ's for serious listening at the listening position and casual listening everywhere else.</b></p><p>I need this in particular because my listening position is anti-modal. Near the center of the room, the larger modes cause cancellations instead of augmentations (they augment at the peripheries).</p><p>I choose to listen in the center of the room anyway so I can be as close to the speakers as required for the widest possible stereo separation. That opens everything up in an incredible way. A friend clued me into this. I was previously a back-of-the-room listener, with speakers in front, and that felt 'natural.'</p><p>I could in principle achieve similar separation with the speakers in the middle of the room, and listening in the back. From the standpoint of getting good bass (and especially getting the best possible bass out of electrostatics) that is far better. Then I would not be listening from an anti-modal position, and I might even get away with no EQ and no subwoofer too. (Though any time a subwoofer is being used, you are almost certain to need EQ anyway simply because of that, subwoofers excite room modes in ways that panel speakers avoid. But if the panel speakers were in the center of the room, you might get away with only one EQ for both serious and background listening.)</p><p>But that just doesn't fit my home and my lifestyle. I do not have a 'dedicated' listening room. I have a living room I walk through about 100 times a day, and have movie parties in 1-3 times a month. Speakers in the middle of the room would mess everything up, and Acoustat 2+2's are not very easy to move around.</p><p>Then people tell me I could simply "enlarge" my living room. Geez. I have enough trouble keeping up with home issues as it is. Last year I spent $21,000 on foundation repair and related issues. It was because of the fear of issues like that I built an entirely separate climate controlled storage building with its own very heavy duty foundation ten years ago. I didn't want to tack more rooms onto my already struggling house. (I had bigger plans for the storage building, but in the end it became just a storage building, because I needed that.) As serious of an audiophile as I am, the changes needed to enlarge my living room are basically unthinkable. And I have other major home improvements already long in the queue, such as a new patio cover and carport.</p><p>Ultimately I think it's not bad at all for me to figure out how to fix these sorts of problems either.</p><p>Anyway, with these changes, it seems I can play anything at any level and enjoy it much more in the background now.</p><p> </p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-39474197962383923862023-12-25T11:36:00.000-08:002023-12-25T23:33:52.938-08:00The Grinch Lost<p> Since October, the Grinch has visited me many times.</p><p>Upon returning from vacation on October 31, I discovered the DEQ for the supertweeters had become dysfunctional. Attempting to bypass, I then discovered the entire right channel of the supertweeter had died--they were shorted. Some kind of catastrophic failure I hadn't noticed. It appears the amplifier is still OK.</p><p>I managed to get a replacement for the nearly unobtainable D21AF right away. Suprisingly, it has been much harder to replace the Vifa tweeter in back.</p><p>But then, I discovered that merely moving the supertweeter towers out of the center made the imaging far better. It turns out there's a "central" rule: don't put stuff in between dipolar speakers.</p><p>So, the grinch lost that one. While it may take a year (I've determined not to hurry it) to get a new supertweeter system in place again, it will be far different. It will not be on a tower in between the speakers. My current plan is to have the supertweeters behind the Acoustats, just above "interface" level. More recent testing has revealed the Acoustats are nearly acoustically transparent, so little harm in putting the supertweeter behind and then compensating for the time difference digitally, as I always do.</p><p>The grinch has really determined to mess up my audio life in the past couple days.</p><p>In the previous week I figured out to hook up my modified QSC ABX switch box to switch speakers instead of amplifiers. That seems to work fine (but I need to inspect if the amplifier is ever "shorted"). Only then was I able to nail down the difference in sound between my newest 2+2's and my original pair, with instantanous switching. Then I learned a bunch of things:</p><p>1) Instantaneous switching is near essential for making audio comparisons</p><p>2) The new speakers sounded way louder, but measured softer. They could sound more dynamic without any level compensation (and which way???) but had a glare that became objectionable quickly. Possibly lowering HF level and changing to C mod would help.</p><p>I then started working on lowering the level. It took sanding the variable resistance bar in the interface in the left channel. I had disconnected the speaker wires during this time. But the first time I plugged the speaker wires in it didn't work, and it took several more attempts of sanding to get it to work. I also plugged the speaker in while it was connected to the amplifier (which I now think could be a no-no).</p><p>I had the channels reversed and messed with fixing that. Somehow in the process of modifying the speaker and reversing the (complicated extended and automatically switched) connections I managed to kill my Hafler 9300. One channel first became increasingly noisy (which at first I thought was the modified speaker, or the connections) and ultimately it went out completely. It took some testing before I realized it was the 9300 that had failed, it was not a matter of the (now very complicated) speaker connections.</p><p>I'm now worried that it was actually the new 2+2 on the right side, which makes a very soft arcing sound about once per second, that had killed the amplifier. Perhaps especially when I plugged in the speaker.</p><p>I can't blame the new speaker for certain, but I now think it's a risk not worth taking. I'm not going to play that speaker with a non-expendable amplifier until the arcing has been fixed.</p><p>So the grinch blew up my favorite amplifier, killed my new pair of speakers, and killed my new speaker testing plans and even my imagined post-testing scenario: how I first planned to use the new Acoustats in bedroom if I couldn't get them to sound as good as the older ones (which of course had been the original hope).</p><p>Pretty bad.</p><p>But it gave a renewed purpose in life to my Aragon 8008 BB amplifier that's been waiting there in the living room all this time, as my backup and "alternative" amplifier to test to see how audible amplifier differences are (try as I might, in blind A/B testing, I've never been able to consistently identify which is which...which right now comes in handy...because I know I'm not 'missing' anything...I don't have to pine over the now dysfunctional Hafler).</p><p>In fact, I had spent the last month or more pondering the sale of the Aragon amplifier. Though it's clearly the 'highest end' working amplifier I own, I just didn't need it. If anything, I thought the Hafler sounded slightly better overall (though not being able to prove I could hear this difference). My sighted listening impression is that the Hafler sounds 'sweeter' (less distortion) whereas the Aragon sounds 'punchier.' </p><p>I didn't need a backup amplifier, I was thinking, because the Hafler (unlike the Krell FPB 300 before it) seemed totally reliable. I'd never had one of my two Hafler 9300's fail. I thought they were virtually indistructable.</p><p>And now that I'm leaving the Aragon on all day, I'm thinking I might like it better. It's got at least twice as much power and current handling capability (it has two 1100VA toroidal transformers, that's a total of 2200VA--too much for one Insteon switch to handle btw). I'm feeling the extra bass punch and solidity, exactly what I'd been hoping to get from the newer speakers (and sort of did, but also with listening fatigue).</p><p>Both amplifiers have impressively wide bandwidth, I know it's 300kHz for the Hafler, I thought the Aragon was much less than that but in fact <a href="https://hometheaterhifi.com/volume_4_4/aragon8008bb.html">Secrets of Hi Fi and Home Theater reported 550kHz</a> for the Aragon. <a href="https://www.stereophile.com/content/aragon-8008-power-amplifier-measurements">There is no indication of any early rolloff in the frequency response shown by Stereophile for the 8008 (regular model)</a>, though there is a slow rise in the distortion floor above 1kHz, nevertheless the amp exceeds it's 0.07% THD spec. The Hafler does have lower distortion, but it hardly matters at these levels. The Aragon gets high praise in all the High End Audiophile magazines, even The Absolute Sound, meanwhile the Hafler got high praise in Audio magazine (it SHOULD have been noticed more IMO, it was a sleeper). The Aragon has the basic "good" design standard by the 1990's for bipolar amplifiers: complementary symmetry DC coupled with servo, not unlike the GAS Ampzilla of 1973, but with much better parts down to the teflon coated wire and high end circuit boards. And massive heatsinks so it doesn't need a fan. And the high bandwidth plastic transistors were a special limited edition from Toshiba before with extra fat die for elevated ruggedness on top of high linearity. They ran out of replacements in the 90's, all you can get now are the more conventional variety. Mine was in the earliest series that was contract built by a manufacturer in Connecticut that really knew their stuff, the circuit boards and everything look instrumentation grade. Later Aragon had their own factory in California. Then they were absorbed by Klipsch, then spun off a separate company again today. Always in the niche right below the insanely high end. Back in the 80's they were distributed by the same people as Krell, perhaps that's how the rumor got stated that Dan D'Agostino himself designed the basic 8008 series of amplifiers. If there's any truth to that, it would have to be the cocktail napkin kind of design, where D'Agostino sketched out the basic design, and an engineer named Robbii did the detailed design work, his name is on the circuit boards. I think it's more or less textbook as far as the basic amplification circuit, except the bias is probably simpler than the textbook kind which I think would be more regulated. I don't think it's as finely tuned as something by D'Agostino, Curl, Pass, or another of the amplifier designer legends might have created. But if you're not cutting corners, by the 90's it was fairly easy to do a top notch Class AB+ amplifier that would blow everything from the 70's away. So this is the "value" high end amplifier with over-the-top high end parts and build but without the fancy designer label (though nowadays Aragon is plenty fancy). And that's good enough, it doesn't take a one in 10 million genius to design a good audio amplifier, pretty much all you have to do is copy the one of the transistor manufacturer design examples, with a bit of fine tuning. And if you're not pushing things to the limit, it's probably going to last longer too.</p><p>I met the Mondial (Acurus and Aragon) co-founder named Tony. He showed me the Aragon in a back room of an audio meeting (after the Meitner amps had failed spectacularly in an audio meeting playing Apogees...they weren't supposed to be doing that and the store owner was pissed). Tony bragged about how could stack stuff on the amp without issues and he even "warmed it up" the sound by putting a blanket over it for 15 minutes. He was absolutely fearless about it, but I was scared to leave it covered that long so I took the blanket off a bit earlier, but indeed the sound was already warmer. I've discovered the bias supply is very "flexible." As the amp warms up, the bias goes even higher. But if you put the amp on tall feet, as I did, you then give it "too much" airflow and the amp cools down and the bias goes down too. So with tall feet I had to re-adjust the bias higher. I've set it so that with the tall feet and nothing on top it runs at a perfect 125F after one hour where it measures and sounds great. Balancing the heat of the two channels is essential too, otherwise they "cycle" pushing each other up and down. There is incredible thermal mass and it takes a long time to heat up or change. And of course if it gets too hot, it shuts down, or at least is supposed to, well before anything breaks.</p><p>I guess you could say this is "old school biasing" before Nelson Pass patented 'optical bias' went mainstream (btw, it's what made most later era Krell amps with plateau biasing possible, and I'm pretty sure it's used in Levinson and most other high end class AB amps). Old school biasing has it's "advantages" in being kind of self-limiting, but it really needs to be adjusted for your exact situation to perform perfectly. But then it does really well. It's fairly easy to access the Aragon controls but not easy to get them set just right, to avoid cycling, etc, and depending on the height of the feet under the unit.</p><p>The Mondial founders were not themselves engineers, they were audio marketing guys who felt there was a missing niche just below the very high end they could sell along with other brands (which originally included Krell, which was sold by the same distribution chain) and not compete with them, by having more basic designs.</p><p>I've owned the Aragon 8008 BB longer than any other amplifier without any failures, and the way it's conservatively designed and built you'd expect that (but often don't get as much durability from the more prestige brands, which are often more like race cars needing pit stops after every lap). So on top of everything else, the Aragon is a survivor. At this point in my life, my system is all survivors, and I honor survivors. (Some things survive because it's easy for me to get them fixed, others like the Aragon just keep on working, and that's good because fixing it could be somewhere between very expensive and impossible.)</p><p>I know really don't have to continue the speaker test in earnest, which in a way is a kind of relief. The plans I had for moving everything in the house around to make way for another major pair of speakers are for the time unimportant. Now I merely need to plan for the somewhat simpler storage of all the new speakers, and keeping the arcing pair out in an accessible way so I can work on it.</p><p>So here are my wins:</p><p>1) My most high end working amplifier has renewed its purpose, and I'm liking it better than ever.</p><p>2) I don't have to sweat all the speaker moves I was planning.</p><p>3) I get to tear down an electrostat speaker with no worries since it's already broken (unlike the clocks I took apart as a kid and into my teens).</p><p>4) I've learned to love my original pair of Acoustat 2+2's better than ever before. They've stood high even against another pair of 2+2's. And it's clear my slightly attenuated HF helps. The default position sounds way too bright and becomes fatiguing.</p><p>5) Failure after failure, and my living room stereo keeps working and sounding better all the time despite them.</p><p>6) I'm finding that indeed the higher power of the Aragon 8008 BB makes it possible to play louder than I felt like doing before, and then it's magic. To really appreciate this at first it may help to be drunk, as I was on the evening of December 25. I had to move the Mapleshade damper back on to the Hafler to stop the rattling. And the right channel of even my older 2+2's has a rattle at louder levels, even with my 8th order linear phase crossover. Eventually I'm going to have to rebuild that too... (I was hoping to THIS time around, but the new pair aren't a suitable replacement yet.)</p><p>I plan to continue to test the one new 2+2 that is not arcing, to see how similar I can get it to sound to my originals, etc, so I can use at least that one as a backup unit for now.</p><p>I not only fear that an arcing speaker could destroy the amplifier driving it (by sending spikes of HV back to the amplifier) it probably cannot sound as good, because the uniform charge of the panels is disrupted, causing distortion. So not really worth further sonic testing. Once it's arcing, it's over. I should have known that.</p><p><br /></p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-55265639134925987752023-12-22T11:24:00.000-08:002023-12-22T11:24:13.600-08:00My B mod and C mod Acoustats<p>I am still struggling to define the difference between my two sets of Acoustat 2+2's.</p><p>It's certainly not just a matter of B mod vs C mod. It's also that the level control was adjusted downward on my first 2+2's. I adjusted it that way thinking it was flat, but it was actually 2-3dB down perhaps. And I also liked it that way, in fact EQ'ing down a bit more. Though it might be better flattening out above 12kHz rather than continuing on a downward slope.</p><p>But I haven't adjusted the newer-to-me B mod speakers to that same level, because it's a bit harder to do, you have to open the interface case and fool with a strap (which requires deoxit cleaning). I'm now very certain I like the old knob level control better, but there should be terminals you could measure for exact settings. (Actually, in the older interface style, like my first speakers, with the original A mod in them, you could measure the DC resistance across the input terminals to set the HF level repeatably. But since mine have C mod, I can't do that, and have to break the case apart--difficult in the older interface style--to make measurements, which I'd been planning to do since forever.)</p><p>Also, I wasn't even sure whether I liked the greater HF output or not. It looks like it might even be objectively an improvement. And sometimes I've thought it was. But I'm leaning the other way right now.</p><p>But there could be still other factors.</p><p>What's really mind boggling is that the newer-to-me speakers (B mod, higher HF level) sound a lot louder. So I adjusted the level downward, finding the matching point to be about 2.5dB. After making that change, a lot of the added "punch" of the newer-to-me speakers went away. But they seemed more similar if still different.</p><p>Anyway, objective level measurements don't back this up at all. In fact the C-weighted pink noise of the newer-to-me speakers doesn't measure louder...it measures 2-3dB softer! Changing the weighting to pink noise brought the two speakers much closer, but with the newer-to-me speakers still about 0.5dB softer (about the margin of error).</p><p>Note that this is still with the newer-to-me speakers in front of the others, which seems to make surprisingly little difference.</p><p>The 'louder sound' of the newer to me speakers was especially apparent to me with pink noise at a high level. At levels I used to think OK the newer speakers sound overwhelming, whereas the older ones sound slighly distant and unfrightening. (And this is with the C weighted noise of the more distant sounding speakers actually being 2-3dB louder!)</p><p>I'm still finding the 'harder' sound of the newer speakers to be sometimes bothersome.</p><p>So I don't know what the issue is. Is it that the B mod has more distortion, making it 'sound louder' ? Or is it that some of the old-to-me speaker panels have weakened? Or is it just the HF level difference?</p><p>Should I swap my old modified C mod interfaces in, and with or without adjusting HF level to -2dB?</p><p>Should I just reset the levels in the newer interfaces to -2dB (which isolates the level change)?</p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-41101131649388673672023-12-10T09:32:00.000-08:002023-12-28T10:52:47.300-08:00Acoustat "C" mod<p> The Acoustat "C" mod (as found in the MK 121-C interfaces) is always strongly recommended by Acoustat gurus such as Andy Szabo (who has answered questions about Acoustat for over 10 years at DIYAudio) and Roy Esposito (who rebuilds transformer interfaces). They are both former Acoustat employees (the Acoustat company does not exist anymore). "C" mod can easily be applied to any previous Acoustat transformer interface. Though the factory "C" version included the Medallion transformers also included with "B" version, the Medallion transformers are not necessary to perform the mod.</p><p><br /></p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhsGcmOY0TNAmMvSM4fghzczv_vuOk4xZp_Gb_fh2TDUSG9Y8xQKPGp9YwYE-z_qVV_KhOkXkJFOoar08HapC_qTLU82C3xeDJ8-Yx71_z3jD171nJ56zU05YzN5h0ceZjGsv-H0WD3i2g97_d-tuQ4Ct38y5PXA6lwtVJuP7uSVLQEmzPweKot-bcf2g8/s678/Acoustat_Transformer_Mixer_Coupling.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="651" data-original-width="678" height="307" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhsGcmOY0TNAmMvSM4fghzczv_vuOk4xZp_Gb_fh2TDUSG9Y8xQKPGp9YwYE-z_qVV_KhOkXkJFOoar08HapC_qTLU82C3xeDJ8-Yx71_z3jD171nJ56zU05YzN5h0ceZjGsv-H0WD3i2g97_d-tuQ4Ct38y5PXA6lwtVJuP7uSVLQEmzPweKot-bcf2g8/s320/Acoustat_Transformer_Mixer_Coupling.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Acoustat A and B vs C mod</td></tr></tbody></table><br /><div>(Ignore the green arrows, which are intended to show the backward secondary coupling that is alleged by one DIYAudio poster to cause saturation in the HF transformer of the original design. Others dispute that. I wonder why saturation would not also be caused directly by the direct coupled primary circuit of the original design.)<br /><p>"C" mod was the last iteration of interface designs before the significantly different Spectra series of Acoustat speakers were introduced. So "C" mod was the last factory design for Acoustat models such as 1+1 and 2+2.</p><p>One of the reasons it is recommended is that it attenuates the low frequencies being applied to the HF transformer, reducing distortion. </p><p>At first glance, it seems very strange that in the original MK-121 interface there is no actual "crossover" on the primary side of the two Acoustat transformers. With both the "A" and "B" mod interface designs, the HF transformer is DC coupled to the input signal, with only a capacitor bypassed series resister in line (whose purpose is tuning the high frequency peak around 10kHz, though it seems weird such a large capacitor and small resistor would have that effect, when their own RC constant suggests they work in the midbass, but their effect is just enough to shift the ultimate HF pole).</p><p>The actual "crossover" for the HF transformer is only on the secondary side, a 0.01uF capacitor coupling the HF transformer output to the panel drive signal. The LF transformer has no "crossover" at all except it's own inductance rolling off the high frequency response, which was the reason why the Acoustat dual transformer interface was invented in the first place.</p><p>Is there a difference between crossing over the transformer on the primary or secondary side? I'm not sure, I would have never thought about crossing over a coupling transformer on the secondary side, that it even works OK wouldn't have been obvious to me. It would have seemed to me that even with the secondary wide open, the transformer primary would operate like a choke, and would still be affected by potential bass saturation. (However, chokes do less and less at low frequencies anyway.)</p><p>Comparing that capacitor with the 50k resistor coupling the LF transformer, a first approximation of the highpass cutoff of the HF circuit is (2pi/RC) 318 Hz.</p><p>I can see a one pole at 318 Hz might be a bit inadequate for fully isolating the HF transformer from low frequencies. If the C mod HF control were turned all the way up, it would introduce a second cutoff caused by the 57uF capacitor being loaded by 16 ohms, around 174 Hz. Additional loading from the transformer itself and turning the control to midpoint, push this cutoff somewhat lower than that.</p><p>Further protecting the HF transformer from low frequencies seems like a good idea (though it's not clear how much of a problem it was).</p><p>But at what cost?</p><p>People smarter than me, or at least more familiar with SPICE modeling, have analyzed the B and C mod circuits. And the incredibly curious thing is despite how they look entirely different, and wouldn't work the same at all, in fact they work very much the same. There is relatively small difference between the interface outputs with B or C mod in place. Which is as it should be, presuming B mod was reasonably well designed in the first place.</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjKLrSuABMg5yHVP7N1ec8tS_8i0fCUQ9xEfTBK-Bq4Fmf4i9AcAzuUk2zxB8VQIG0kyjLZL-CY6Nxz2DxbMYdQcrFdGJwlco2vR-esSJmjhnBEo4qfzthIvNhuTFgIFk_Rq55XaUpuvqPzLe8d3PXJMrwz5hiiG649MiqIq5f1RwyF95n32wuJ83Q1SDs/s768/MK-121_interface_results_04b.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="659" data-original-width="768" height="275" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjKLrSuABMg5yHVP7N1ec8tS_8i0fCUQ9xEfTBK-Bq4Fmf4i9AcAzuUk2zxB8VQIG0kyjLZL-CY6Nxz2DxbMYdQcrFdGJwlco2vR-esSJmjhnBEo4qfzthIvNhuTFgIFk_Rq55XaUpuvqPzLe8d3PXJMrwz5hiiG649MiqIq5f1RwyF95n32wuJ83Q1SDs/s320/MK-121_interface_results_04b.png" width="320" /></a></div><br /><p>But there is some difference, the two very different circuits obviously do not work exactly the same. I would have expected more rolloff in the deepest bass, but in fact the C mod might have slightly higher output, like 0.1dB or so, at 20-40 Hz. But it's in the midbass through the midrange, 100-1000 Hz, that the C mod has up to 2dB less output. The absolute advertised gain for the C mod in the extreme highs around 10kHz and above is very very slight, it's only relative to the 2dB attenuated output at 1kHz that it's "relatively" significant.</p><p>If you played these two versions in a simple test without adjusting the levels to compensate for the greater loudness of the B mod, I'd bet that almost all of the time audiophiles would prefer the louder seeming "B" mod. It might still win in a properly level matched (best to use C-weighted pink noise) comparison because 100-1000 Hz is basically the heart of the music. The relative and tiny absolute gain in the very extreme highs above 10kHz is debateable, <span style="font-size: xx-small;">I've had an ear tuned notches tamping 2200, 5200 and 9100 for C mod actually</span>. It certainly has more 'punch' if you don't adjust the level downwards (when switching from C mod to B mod) to compensate (I've found a 2.5dB downward adjustment in gain for B mod (vs previous C mod) for the Acoustats to work best by ear, eliminating the 'listening fatigue' I get when it's set higher, relative to the subs, but after you compensate for the difference in output with a 2.5dB downward adjustment...I don't yet if the 'punch' is different, I think it is slightly but not as much as I'd hoped when I first heard it, the increase in punch was shocking, but I hadn't realized I was playing 2.5dB louder 100-1000 Hz either. (<span style="font-size: xx-small;">Plus, I'd always set the HF level down before by accident, not realizing "flat" was 3 o'clock instead of 12 o'clock. So it's louder in the highs not, despite not being a difference <i style="font-weight: bold;">because of</i> the change to B mod in the my most recent 2+2's but instead because so far I've left the newer speaker at it's factory "0dB" setting, which is tricker to change then with my first 2+2's where I could just turn a knob, but now I'm thinking I like more highs anyway, so I'm not planning to adjust the new but rather to measure how the HF level control on the older one's works, and how much and whether I still need the EQ's with the newer speaker for it to sound best. So, added highs is an additional factor possibly adding punch.)</span></p><p><a href="https://www.diyaudio.com/community/threads/all-acoustat-panels-can-give.282031/page-2#post-4650682">In fact, a number of Acoustat users and even some self-appointed Acoustat gurus do prefer the "B" mod, a fact I only discovered yesterda</a>y (note: the figures in this post come from that thread).</p><p>I'd generally found the C mod midbass through midrange to be somewhat weak, but I've never broadly EQ'd that upwards, I've only notched out a few peculiar resonances to make it more smooth, and curiously added two tiny 1/3 octave 3dB boosts at 850 and 1kHz, where there was a curious suckout, I determined when I had the chairside EQ (in the repair queue for about a year now). But I had been disturbed by the general 'weakness' of that region. Well, going back to the B mod fixes that general 'weakness' from 100-1000 Hz, or at least the new speakers do.</p><p>The significant loudness increase with my newest 2+2's (which have B mod) might be entirely due to B mod! Which I had previously assumed was a step backwards. But there might be other reasons too.</p><p>With more midrange, midbass, and deep bass, the B mod might have "punchier" sound as well.</p><p>The only noticeable lack in the B mod from the measurements is a very slight extension in ultimate high frequency response, which looks pretty small and fairly unimportant to me.</p><p>I'm not going to assume C mod is necessarily better any more. I now have the ability to swap B and C mod interfaces into my newest 2+2's and I will measure them and listen to them and see.</p><p>Note that while the B mod will play as much as 1.5dB louder than C mod with the same input signal, the ultimate loudness limit (provided sufficient amplifier power is available, < 200 watts into 4 ohms) will likely be about the same. The transformer and panel limitations will remain the same.</p><p>DIYAudio poster Bolserst posted some more illuminating simulations and measurements to the aforementioned thread.</p><p>First, he shows B mod mixer drives the HF transformer with the full bass signal, but the C mod mixer rolls off the bass to the HF transformer (this doesn't seem to show the effect of the capacitor in the secondary circuit however).</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhLMTI53n0GaK3rPFVaUJAfhr9dmk5JMNsn5IhkSvlk06AhpLMT44-VVL6WU0m7an0hPst7dB-5z8tiwRusPGWQiWefbmxd4JZ23_eBymuo_Z5OZ23n0Qc-sU3K7hyphenhyphend0xUibubH1QHUnD2SnJ5KjivWWUZTuIFabHyz5Ce62V9rxvUba-7dD42f7wGMbdQ/s768/MK-121_interface_results_01.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="552" data-original-width="768" height="230" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhLMTI53n0GaK3rPFVaUJAfhr9dmk5JMNsn5IhkSvlk06AhpLMT44-VVL6WU0m7an0hPst7dB-5z8tiwRusPGWQiWefbmxd4JZ23_eBymuo_Z5OZ23n0Qc-sU3K7hyphenhyphend0xUibubH1QHUnD2SnJ5KjivWWUZTuIFabHyz5Ce62V9rxvUba-7dD42f7wGMbdQ/s320/MK-121_interface_results_01.png" width="320" /></a></div><p>In later graphs, he shows these spice models do fit the measurements almost exactly. Including the curious lump in the LF transformer response, which goes away in the C mod. It's actually that "lump" which causes the boost in B mod vs C mod output in the midrange. The HF midrange response is identical!</p><p>There is no change to the LF circuit at all, so what explains this lump and how it goes away with C mod? There are no changes in the LF circuit at all! That brings us to Bolserst's theory that the C mod reduces the low frequency backward coupling of the LF transformer into the HF circuit, which somehow pulls the LF response higher.</p><p>Greater output in the midrange might make it sound better, on the other hand, if there's more distortion added, that could make it sound worse. In my previous measurements the Acoustat (with C mod, all I had ever used before) is very low distortion, but perhaps distortion is slightly higher in B mod.</p><p>Since I've readjusted the level to -2.5dB from before, I haven't had any more episodes of listening fatigue, but I wouldn't be surprised if C mod was slightly cleaner. So perhaps it will win in the end? I'm going with my newer speakers anyway because I can switch from the current B mod to my original 1+1 speakers hot rodded (external Solen cap) interfaces, because my latest 2+2's have the compact interfaces which are quickly swappable.</p><p>That feature alone is important, but for now I'm thinking they sound at least as good with B mod as my originals do with C mod, or possibly better.</p><p> *****</p><p>Update December 23</p><p>No question anymore, the 'hardness' of the new-to-me 2+2 with a B mod interface is intolerable. I don't know yet what the problem is, perhaps I just need to turn down the HF level comparable to my other 2+2's. Perhaps the C mod does reduce audible distortion, and once you've gotten used to it nothing else will do. I can make all these changes with equipment on hand, I would love to swap out the interfacees in the new-to-me 2+2's with my hot rodded 1+1 interfaces from back in the day, with huge external polypropylene cap and everything. But to be fair, I'm going to take the first step and changed the new-to-me 2+2's to their lowest HF level and see what that does.</p><p>It occurs to me that the way the HF level control "works" is that it actually creates HF ringing. The "flat" position is with a little ringing, and the +2 is with more ringing. -2 is probably what you need for no ringing.</p><p>And the ringing probably is more audible with B mod having in effect higher Q.</p><p>Still, I imagine many would prefer the 'louder' sounding sound.</p><p>Update December 28</p><p>I should not have conflated the differences between my two sets of Acoustats with the difference between B and C mod.</p><p>It now appears the fatiguing quality may have been mostly that the arcing in the right speaker, though impossible to hear except with ear at speaker, was making it more distorted. The electric charge was non-uniform.</p><p>Also the difference in HF level. But now, very indirectly so far, it does seem like B mod may sound somewhat louder irrespective of HF differences. (And also that in my room and to my taste, the HF level needs to be turned down at least 2dB.)</p><p>I will shortly be able to swap in my C mod interface, which will be the closest I've come to doing a fair test. I still have the B mod speakers in front of the C mod, giving the B mod an artificial advantage one would think.</p><p>I've disconnected the arcing speaker, because it might have damaged my amplifier, and I can't risk my next amplifier, the amplifier I'm using now, which I'm liking more than anything ever right now, the Aragon 8008 BB, with Mapleshade carpet feet and my superlative bias adjustment.</p><p>Until I get that arcing fixed, I'm only going to play it with "expendable" amplifiers. For now that speaker going back to the repair pool.</p><p>I might be first repairing my original right channel with C mod. It also seems to have some distortion playing You Think Too Much About Flying Saucers, but, playing the same right channel signal on the new B mod 2+2, there is no distortion.</p><p>So it's looking very much like when and if I get them matched, I'm going to be mixing my new and old 2+2 pairs until I get at least one of the defective ones fixed.</p><p>And to really do that correctly, I'm going to pull my old modified C mod interface out, so both will have C mod, at least for starters.</p><p><br /></p><p><br /></p></div>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-8773462023141420602023-12-07T13:22:00.000-08:002023-12-09T20:48:11.865-08:00Acoustat 2+2 (updated Dec 8)I've complicated my life by buying (at very low price) two more stereo sets of Acoustat speakers, 2+2's and 1+1's. This means I now have 2 sets of each variety.<div><br /></div><div>I don't actually intend to use these speakers in any current room, except that I might place 1 set of 2+2's (whichever I decide is least good) into the Laboratory, replacing the 1+1's already there. After all, the 2+2's are better, etc. Ultimately, the 2+2's will be the best speaker for the new Gym, after I expand my existing Gym to the full width of the garage after I build a carport. (I'm not sure I will live to see all these imagined improvements anymore, however, because I haven't done anything in more than a decade of thinking about them, and I have less money now.)</div><div><br /></div><div>My main hope was that the new 2+2's would be better than my pre-existing set, and replace them, leaving the older pair as backup or experimentation.</div><div><br /></div><div>I'm not at all sure that is going to be the case yet. In fact, one of the speakers has a notable rattle played full range. But the same is true of one of my pre-existing ones. The possibility exists that I will end up using one of my old ones and one of my new ones. Then it will look pretty wierd, with a white 2+2 on one side and a black 2+2 on the other.</div><div><br /></div><div>Anyway, my first month is chalked out to be a full examination of both sets of Acoustat 2+2's. That is possibly the main reason I bought them, for testing and experimentation. Having only 1 set, I am unable to conceive of minor repair and/or modification. My system would be done for the duration. Now I have a second set I can experiment with, including taking the grill cloth off. (The grill cloths on the newer set is after-market, looser, and possibly easier to remove than on the originals.)</div><div><br /></div><div>The newer set also has the "compact" interface unit which is far easier to remove...in fact you can normally leave it unscrewed and simply pop it off whenever. Having the compact interface also means that I could use my already Solen-cap equipped original 1+1 set, to which I added a large external capacitor (not an easy modification!) to replace the original 47uF electrolytic.</div><div><br /></div><div>My original set is the from the original production run, with the impressive looking "large" interface unit strapped to the speaker. It was refurbished and modified to "C" mod by a famous modifier. The new ones are both Blue Medallion. They might also have been modified to C mod but I don't know yet.</div><div><br /></div><div>Things I want to know are:</div><div><br /></div><div>Which units are more plagued by low frequency rattles, resonance, and similar issues? How hard do they have to be driven to exhibit these flaws? (I intend to do only very safe non-destructive testing in these regards, but I already know several songs that are pretty likely to cause problems)</div><div><br /></div><div>Are the new sets "C" mod?</div><div><br /></div><div>How have I tuned the HF control on my original set and what difference does changing it make? What position is actually best? Is it better to achieve the HF balance I like with EQ or with treble adjustment? (I am currently using EQ, I have the control set in the middle which I think means flat though it is not marked.)</div><div><br /></div><div>How do the frequency responses of the original compare with the new? Can they be adjusted to be the same, or whatever is best? Does the C mod vs B mod make a difference in frequency response or sound (if one unit is B mod, which I don't know).</div><div><br /></div><div>And that's just the start. My thinking was I belabor many other mostly useless stuff and don't focus on my main midrange speaker, which is old and possibly needs repair or rethinking.</div><div><br /></div><div><b>Well, with just one set it's almost untouchable.</b> I don't even dare adjust the treble control for fear of losing my current assumed best judgement. I need to measure the control first. And with large interface and C mod that's quite difficult to do since scads of bolts need</div><div>to be removed including those on the brace.</div><div><br /></div><div>The downside is...I've got so many other things to do these days (holidays!) AND my system was not only sounding totally wonderful and <b>I was just getting used to the vastly improved imaging and center stability after moving the supertweeter towers out.</b> Now I've got two sets of speakers in the room I can only put in front of each other (hardly any room for anything else) until I do hoped-for experiments as I've described above. But it's surprising how acoustically transparent Acoustats are. Having a second pair powered (!) but no signal right in front of the powered ones has surprisingly very little effect on the out-of-room sound...what I hear the most. I think the supertweeter towers s had more deleterious effect on the out-of-room sound regardless of whether the supertweeters were unpowered. (They may have contributed more to the out-of-room sound in the first place too, when powered vs not.)</div><div><br /></div><div>One of the problems is not just that they take a lot of space to store...they are fiendishly hard to move. Since they're taller than all doorways they need to be tilted just so. It's pretty much a two-person job.</div><div><br /></div><div>That means any moving around is going to be, by necessity, a two person operation. But I think I can assemble them mostly as one person, working on the sides... (So far, I've always assembled and disassembled 2+2's with help.)</div><div><br /></div><div><a href="https://www.stereophile.com/content/acoustat-22-loudspeaker">Here's a fairly positive 2+2 review by J Gordon Holt in 1984.</a> (Harry Pearson was far more negative, coining the term "credit card coloration.")</div><div><br /></div><div>I dispute Holt's suggestion that the Acoustats have a less punchy sound because of an alleged high frequency suckout.</div><div><br /></div><div>In fact, I find that with the controls set to neutral they require slightly more cut in the 2-6 kHz (and 12 kHz) regions to sound good.</div><div><br /></div><div>This yields a fairly evenly but slowly declining response curve from 2kHz upwards. That's what seems to sound the best, possibly because of the large degree of high frequency reflected sound created with a dipole speaker.</div><div><br /></div><div>The lack of punchiness, I believe, comes from other issues.</div><div><br /></div><div>1. Dynamic speakers sound artificially punchy because of cone resonances, and people are used to that.</div><div><br /></div><div>2. The Acoustats have deliberately designed broad positive resonance below 100 Hz by design to compensate for bass loss due to dipole design.</div><div><br /></div><div>This resonance is not described in any Acoustat literature I've seen. It is rarely discussed at all. Most probably don't know about it.</div><div><br /></div><div>Yes, the Acoustats have to have this resonance to even sound remotely like it has bass. That is the magic, right there. A tower of RTR</div><div>electrostatic tweeters would have very little bass.</div><div><br /></div><div>While electrostatics apply force over area (fairly) evenly, that does not necessarily mean the motion directly follows that force. The motion also depends on the mass of the diaphram, it's elasticity, it's tension, and similar factors. So yes, there can be resonances.</div><div><br /></div><div>Linkwitz added electronic bass boost to his dipole speakers. He published how he calculated how much was necessary, and everything else about building the (analog IC based) circuits required.</div><div><br /></div><div>Acoustats have that bass boost in the panels themselves, a function of the relatively high mass of the membrane compared with other electrostats. (The Acoustat membrane is 2-3x thicker than most electrostats.)</div><div><br /></div><div>So, I think the problem is that the needed resonance in the Acoustat bass adds time dispersion to the the bass, reducing the "punch."</div><div><br /></div><div>I've always thought that the Acoustats were fundamentally flawed from the vision of being a "full range" speaker. They are not "full range" to the fully demanding. They need a subwoofer.</div><div><br /></div><div>And therein is a big rub, because the subs available in the 1980's were not as good or cheap or plentiful as today.</div><div><br /></div><div>Today we can have incredibly good subs powered by their own internal digital amps, with DSP processing to optimize them.</div><div><br /></div><div>Now we can easily have subs that mate with electrostatic panels.</div><div><br /></div><div>And all the better to use FIR based linear phase high order filters for the crossover.</div><div><br /></div><div>That's how I can get a sonically wonderful 8th order phase corrected Linkwitz-Riley crossover for the subs and panels at 125 Hz, far above the resonance and wall reflection issues.</div><div><br /></div><div>So the resonance, even rattles, doesn't matter much to me anymore. But that also means I could get by with a large 2+2 with thinner membrane and no bass resonance at all. That would be my preference, but nobody makes such a beast.</div><div><br /></div><div>Speaker makers want to make full range 'speaker systems' not parts for audio enthusiasts to use in constructing their dream systems.</div><div><br /></div><div>And I can simply not imagine building a electrostat speaker to my own requirments.</div><div><br /></div><div>So I am left to working with 2+2's, and now I have two sets so I can be more fearless about experimentation.</div><div> </div><div>**** Update December 8</div><div><br /></div><div>I determined yesterday that all four new Acoustat interfaces are "B" mod Medallion. My first 2+2's had been modified to "C" mod. So I have the perfect opportunity to compare the two approaches. I can compare my old vs new 2+2's, and also swap in my earlier 1+1 interfaces (with jumper change) which are C mod plus Solen 47uF polypropylene cap.</div><div><br /></div><div>However, the "tunings" of the interfaces may be different too.</div><div><br /></div><div>I powered the interfaces with AC from nearly the moment they arrived, because I think that discourages cats. I have NEVER had a problem with cats clawing Acoustats (as J Gordon Holt reported). The front panel fabric is too "floppy" like a cat-proof screen door, cats recoil against things with that feeling. They could potentially claw the sides, which are hard. But I have never seen it. I think it also helps if the cats hear the new Acoustats playing music as soon as possible. Cats respect music. Things that make music are "alive" to cats, and they don't like to mess with living things bigger than they are that are. I wonder if J Gordon Holt had his cats in the room while he was playing music. Also, finally, I think the bias transformer system creates a very subtle noise we can't hear but cats can, also lending the sense that the Acoustats are alive. And I suspect might be able to "feel" the electrostatic field in front and behind the panels. Their whiskers are very very sensitive to the slightest force. I think they'd be more inclined to go after planar magnetics.</div><div><br /></div><div>Oh, wait, maybe once or twice I saw a cat clawing the side It was just like one scratch to get my attention. I haven't noticed any mark. Curiously it was when I hadn't been playing music for more than a day, so it seemed like that cat was telling me to turn the music (I normally have background music of some kind playing) back on.</div><div><br /></div><div>Speaking of background music, I personally think it's a good idea and vastly preferable to thinks like "news" and "talk radio," which are bound to get you enraged, depressed, or something other than just bubbling along. News is more swiftly digested and understood in print.</div><div><br /></div><div>I've always been very inconsistent about it. sometimes having background music on, other times not, it has been not most of the days of my life perhaps. That's why I automated my system in 2021 and programmed an automatic playlist generator, so I can keep the music playing.</div><div><br /></div><div>I also worked on the sound of the FM radio so I could keep that playing the classical music radio station all the time. Previously the sound would bother me in one way or another after an hour or two and I'd have to shut it off. So now I have:</div><div><br /></div><div>1) take the signal from the fixed output of my Pioneer F-26 (one of the greats)</div><div>2) Run it through a Musical Fidelity X-10 V3, which buffer the impedances and removes ultrasonic crap (I wonder if the V3 isn't similar to the "Noise Filter Buffer" made by audio genius Mitch Cotter. I would not be surprised if the NFB also used nuvistors as they are wonderful for this task.)</div><div>3) Sample with a dedicated ADC, I found the Black Lion Audio Sparrow works extra nice at 24/48.</div><div>4) monitor and EQ the signal a bit with Behringer DEQ 2496. I found a small bass notch below 32 Hz helps, otherwise there can be a very objectionable 20 Hz rumble.</div><div>The digital feeds into my system on coax. When I record FM, I do so with a dedicated Marantz digital recorder, but only with the preceding</div><div>digital chain, as I found the analog to digital conversion in the Marantz to be somewhat "fuzzy," and it works best with either 48kHz or 96khz digital inputs which it resamples in any case (ASRC based inputs) to 48kHz.</div><div><br /></div><div>With those changes, FM became tolerable to listen to all day long, though I still turn it off when I'm thinking or whatever. If I selected my system wide "mute" control, it comes back on in one hour and 1 minute. If I changed the home control selector to DVD and nothing is playing on or streaming from my Oppo BDP-205, it can be sllent all day or until I remember to put something on (as I'm about to do right now).</div><div><br /></div><div><br /></div><div>I hooked up the speakers today. At first there was a problem with the right channel, hardly any sound, and the interface was even labeled "bad." But as I was removing it to swap with another interface, I noticed one of the wires had come loose. It doesn't hold to the screw well, since apparently the repairer/modified replaced the original wing nuts with integrated locking washers, and if you get the loop connector in between the wrong things (above the lock washer I presume) it doesn't hold.</div><div><br /></div><div>Now they're working fine. I first noticed they were roughly the same as my originals. But then I noticed a slight increase in harshness from the FM radio signal I'm listening to. I suspect their brightness controls are turned up more than my originals, and/or it's a C mod (my originals) vs B mod difference, with the B mod sounding brighter. Since the circuits are different, they can't be compared simply by measuring the resistance setting of the HF adjustment. I'll just have to experiment with different HF adjustments to see if the new Acoustats can be tuned to match my originals more closely.</div><div><br /></div><div>*****</div><div><br /></div><div>When I tested the 2+2's on December 7 before buying, I played William Orbit's "You Know Too Much About Flying Saucers." Even at a fairly moderate level, this invoked rattles on one 2+2 unit but not the other.</div><div><br /></div><div>Rattles were among the reasons I devised my 125 Hz 8th order phase corrected crossover for my original 2+2's, though not using this song, using another one by Grouse. 4th order didn't get rid of them, so I had to go to 8th order, which helped in other ways too.</div><div><br /></div><div>I didn't really know the Orbit tune would induce rattles, I just knew it was bass heavy and I liked it.</div><div><br /></div><div>Now I've tested the new 2+2's playing Orbit through my crossover. No rattles, even at uncomfortably high levels. (The bass from my subs sounds very clean.) I reversed the Acoustat channels and it was still free of rattles.</div><div><br /></div><div>So perhaps my motto ought to be, "Bring me your old, tired, rattling Acoustats, and they will sound perfect in my system."</div><div><br /></div><div>The new ones in fact sound great as I used them. They seem to have more "slam," maybe it is the boosted highs after all, or it could be the medallion bass transformers my older system doesn't have.</div><div><br /></div><div>Still seems to support my belief that these are really best used above 125 Hz, as you might expect with electrostatics. But possibly they both Acoustats handled the bass better when they were new, and their mechanical structures more solid, and perhaps membranes tighter as well.</div><div><br /></div><div>So maybe the correct understanding is that you want these crossed over at 125 Hz for the long run. Initially you may enjoy the full (even if somewhat fake...boosted with membrane resonance) electrostatic bass, but eventually looseness and rattles will set it. It's above 125 Hz that it's really indestructible, above that pesky resonance.</div><div><br /></div><div>It's so ironic that an old friend was an Acoustat Monitor (with tube amps) lover who used the Acoustat monitors on the bass. His system in 1981 had Hill Plasmatronics on the highs, sometimes Magnepans in the mids, and Acoustat 4's for the bass. "There's nothing like Acoustat bass," he said.</div><div><br /></div><div>Well, dipolar electrostatic bass is fine in principle, but you'd need still larger panels to get real bass out of a dipole without any boost or resonance in the membrane structure itself. And in fact electronic boost might not be bad in such a system, best all digital with soft limiting and phase correction.</div><div><br /></div><div>Or perhaps dipolar bass is not so fine in principle. It doesn't activate room modes nearly as much, but perhaps it also has less "impact." Another friend was always complaining about electrostatic bass, he regards it as fundamentally wrong, but some people get "imprinted" (his word) on it.</div><div><br /></div><div>In a way, I think he's right. You don't get adequate bass from Quads or Acoustats or many other "full range" electrostatics. I haven't heard the highest end Sound Labs. I only think he's wrong in dismissing electrostatics (not to mention crossovers) altogether. Electrostatics are great used in the middle, between subs and supertweeters. In that range there is no single driver that works as well as a point or line source electrostatic. They just don't do the very extremes very well, most audibly in the bass. And I think it is wrong to expect them too. You just need an adequate digital crossover system like mine to get them to mesh well with subs.</div><div><br /></div><div>For background music standing up and penetrating the house, nothing works as well as a tall line source electrostatic. Every other room seems adjacent to the concert hall, which is all the more pleasant in background as "not in your face."</div><div><br /></div><div>***** Update December 9</div><div><br /></div><div>There is little doubt in my mind now (ie, there is still some doubt) that these new-to-me 2+2's have more dynamic punch. As much as 300% more punch. But, along with this, there is also little doubt (though perhaps a bit more doubt) that over time, they sound more fatiguing. Perhaps I just got tired of examining the 'punchy' sound with relevant recordings at high level. Or perhaps the two observations are part of the same coin. As I previously reported, Stereophile complained of lack of punch and specifically blamed a softness in the highs around 2-4kHz (which is exactly what is required for good sound with most speakers--the Linkwitz/Gundry dip). I wasn't buying their analysis, but perhaps they were right.</div><div><br /></div><div>There's little question at all that the highs are set to a higher level, though it might be the factory selected level in both cases. The new B mod 2+2's have the big power resistor with a strap, which has a clear marking for the "0dB" position. The older 2+2's have a knob which has no marking, but turned right in the middle (straight up) would seem to be the suggested position (though, since it was modified from A mod to C mod, who knows).</div><div><br /></div><div>Possibly Acoustat decided to crank up the default level for the tweeters after the Stereophile review (they reviewed an A mod version) so the B mod has a higher default tweeter setting.</div><div><br /></div><div>Or possibly part of this is due to the B mod vs C mod. The C mod might be less fatiguing as it reduces primary current in the HF transformer. OTOH, the C mod might be less punchy, as it also introduces a second highpass cutoff in the HF circuit response. With B mod the HF transformer primary is direct coupled, with C mod it is not.</div><div><br /></div><div>If I am lucky, the punchiness will not go away when and if I turn down the highs enough to cure the fatiguing problem.</div><div><br /></div><div>So here are some possibilities:</div><div><br /></div><div>1. Punchiness and fatiguing are opposite sides of the same coin. Once the highs are adjusted the same, both speakers will be identical in both parameters.</div><div><br /></div><div>2. Punchiness is caused by one factor, and fatiguing is caused by another. Some guesses are:</div><div><br /></div><div>a) punchiness is caused by B mod vs C mod, fatiguing is caused by level of highs</div><div>b) punchiness is caused by medallion transformers (B mod unit) , fatiguing by level of highs</div><div>c) punchiness is caused by medallions with C mod, etc</div><div><br /></div><div>I need to assess the level of highs in new vs old Acoustats.</div><div><br /></div><div>**** Saturday Afternoon</div><div><br /></div><div>I now have the measurements. As expected, the new 2+2's have significantly extended highs, which only begin to turn downwards from a 4kHz shelf around 14kHz, whereas the old ones begin turning down around 8kHz, mind you this is with the same eq notches I applied for better sound for the first pair, notches at 3kHz and 12kHz. If I continue to use new pair, I might just readjust the notches rather than the speaker HF level. But I do plan to see how HF level works, for the first time, that at least might be what makes this whole exercise worthwhile.</div><div><br /></div><div>Also, the new 2+2's have significantly larger level, at least 2dB higher measured right in front of the speaker in the same relative place.</div><div><br /></div><div>It looks to me like the new ones have either a rebuilt or a less deteriorated HV level. Possibly it was modified/repaired with slightly higher voltage level than stock. But it could all be a matter of relative deterioration.</div><div><br /></div><div>My older pair was worked on by a famous Acoustat modifier around 1998 or so, I think. It seems he would have repalced the HV diodes or other parts if they were deteriorated then. So this makes the difference look more like the new ones having higher-voltage-than-stock.</div><div><br /></div><div>But even that 2dB level can hardly describe the difference in impact the speaker have, listening to pink noise. The new 2+2's sound 10dB louder. At the same level where the old 2+2 sound as smooth as riding way above Cloud 9, the new ones sound like you're riding just below the peaks of Cloud 9, riding through one mountain of cloud after another. The dynamic contrasts are frightening. Just listening to pink noise. Yet it measures about equally flat, just needs a small bit of EQ readjustment perhaps, and/or lower HF level.</div><div><br /></div><div>So I think it's the effect of a higher bias voltage, getting more of that "real electrostatic" sound. Maybe it was more like they were new too, but I think also it's a kind of nitro upgrade done by the same guy (also an Acoustat modifier, if less renowned) who put on new socks and may have done other upgrades, including Cardas jacks and new possibly more acoustically transparent socks. I think the upgrade in</div><div>bias voltage is minor compared with new, but large compared to elderly bias supply. Possibly the diodes were 5-10% higher voltage than new, is my guess, but 30% higher than old.</div><div><br /></div><div>I have, never used, a suitable HV 15kV probe. I burned out one of my meters years ago using a 5kV probe in my first attempt to measure Acoustat bias voltage.</div><div><br /></div><div>Along with it, there's a slight tick about every 20 seconds in one of the new speakers. It takes ear nearly to speaker to hear it. It's quite tolerable like that, though since I bet it results from a tiny bit of arcing it generates ozone and it might get worse, or even self-heal. I smell no ozone and I doubt it's a serious generator.</div><div><br /></div><div>While the highs do need re-EQ if not HF adjustment, I'm liking the more dynamic sound. My old 2+2's were just too laid back. That was their problem.</div><div><br /></div><div>Perhaps that difference will survive HF level, EQ, and speaker level adjustments. Or perhaps not.</div><div><br /></div><div>*****</div><div><br /></div><div>Just turning down the panel level by 2dB relative to the bass (unchanged) seems to have eliminated the "listening fatigue" problem. Somehow, the new panels play louder. I've also noticed that the upper panels on my older 2+2's seem to have diminished level compared to the bottom panels. A more systematic examination should be done.</div><div><br /></div><div>I've also noticed that the manual for the A version of the 2+2's says the "flat" position is 3 o'clock, not the 12 o'clock I had been presuming. So it looks like I've had the HF level too low. Which seemed unintuitive to me because I had to add some EQ notches in the treble anyway. But it seems the new speaker has more <i style="font-weight: bold;">extended</i> response, and perhaps that's what the HF control being higher does. The combination of my existing HF notches and the more extended response sounds about right, more transparent and punchy than before.</div><div><br /></div><div>This may just be about the adjustment, and not the many other differences between my two pairs of 2+2's.</div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div><br /></div>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-73213379557711491622023-11-20T19:50:00.000-08:002023-11-20T19:50:34.040-08:00Supertweeter Changes<p>My next generation supertweeter, to be unveiled on October 31, 2024, will have multiple lines of protection:</p><p>1) Two series fuses, fast and slow blow. I'm going to start with 1a fast and 0.5a slow.</p><p>2) I'm going to stick with the 0.47uF series cap, but upgrade to a polystyrene or polypropylene instead of polycarbonate, and 400v rating or better.</p><p>3) I may put in a shorting lamp prior to the fuses so it doesn't cause nuisance blowing.</p><p>4) I will stick with current 150W amplifier, which is robust. As long as I have series capacitor like 0.47, I need about 100W or more.</p><p>5) I will program the miniDSP used by tweeter with only one option, the 17kHz high pass, so it can't accidentally be turned to flat.</p><p>6) The amplifier will be Insteon switched, and only turned on for "Feature Music." It will be on a power strip that also powers a purple or ultraviolet indicator of some kind, so you can see if tweeter amp is turned on or not.</p><p>7) There will be blinking level lights set to some high level like 1 watt, or perhaps multiple LED's.</p><p>8) No Behringer will be required. The miniDSP will be moved over to where the DAC is, to keep center of soundstage clear.</p><p>9) The tweeter in back will be soft mounted to the back of the Acoustat interface, firing rear, then timed to match front arrival or not (NE19VTS tweeters.) This is all I expect to do at first.</p><p>10) If tweeter in front, maybe later, Dynaudio in protection cages soft mounted to front of base. Dynaudio might also be mounted in back facing forward, depending on how acoustically transparent the Acoustats seem to be at 20-30kHz.</p><p><br /></p><p> </p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-32596362989104079842023-11-20T19:36:00.000-08:002023-11-20T19:36:44.016-08:00Acoustat version info<p>Acoustat 2+2 panels came in several versions:</p><p>3-wire original</p><p>5 wire with red bias wire</p><p>5 wire with yellow wire with spiral red stripe: improved coating, made under Hafler ownership</p><p><a href="https://www.diyaudio.com/community/threads/acoustat-answer-man-is-here.183168/post-7410305">https://www.diyaudio.com/community/threads/acoustat-answer-man-is-here.183168/post-7410305</a><br /></p><p> </p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-26977812359854644002023-11-08T11:00:00.012-08:002023-11-10T15:48:35.230-08:00Supertweeters Removed<p>A week after my return from my vacation in the San Diego area (notably, where much of my audio insanity derives from, where I worked at a high end store and met many high end audiophiles, though I only visited only one such audiophile on this trip, the other notably and disappointingly having given up on the habit) I noticed the super tweeter DEQ unit, which mostly operated as an RTA visual display indicator for the above 17kHz super tweeter signal, as well as an extra digital level control, had a greyed out display.</p><p>I measured the output of the supertweeters using the RTA app on my phone, and sure enough there was no output at all. When the display goes out on a DEQ, the I/O functions <i style="font-weight: bold;">could</i> still be working, but probably not. Probably there's a power supply failure that leads the entire DEQ device to be dysfunctional, not just the display. That is how it has seemed to work in every failure I've paid careful attention to so far. BTW this unit was connected with digital input and digital output, merely serving to pass (and slightly modify) digital signals from the super tweeter miniDSP which does the ultimate high pass crossover, and the Emotiva Stealth DC-1 which does the conversion to analog and final level adjustment, which I've tended to set around +7dB since the super tweeter signal is so low and also highly attenuated by the tiny capacitors I connect the physical drivers through.</p><p>Damned! (And ironically, I'd just been talking about my super tweeters in San Diego, as my still practicing audiophile friend shared his stories with Dynaudio drivers, though he used D28's and not the D21AF's I have been using as front facing supert weeters.)</p><p>I did a quick bypass of the DEQ unit. Then it was still dead in the right channel, but working in the left.</p><p>The way these are hooked in leaves me little opportunities for easy testing. I ultimately had to move the massive sand filled 40 inch high Target stands with LS3/5A boxes on top, which is not getting any easier for my 67 year old body despite the (very limited) strength training I do. Which I didn't bother to do until exhausting all other opportunities. And then it was clear on the proven good signal of the right channel which plays fine on the right super tweeters that the left super tweeters were not working at all.</p><p>This was true even bypassing the capacitors. Measurement suggests both super tweeters on the right channel, including the now unobtanium D21AF in front, now have open windings.</p><p>Double Damned!</p><p>But now perhaps is also the best opportunity to figure out what it sounds like <b><i>with the super tweeters and their stands removed.</i></b></p><p>You would probably not be favorably impressed by the fact that I just hadn't done that experiment before. I'd only compared the situation with the super tweeters powered and not, and powered always seemed somehow <i style="font-weight: bold;">magically</i> better. Bass seemed clearer, and highs seemed more balanced, never harsh or closed in or honky.</p><p>Always, the difference was very slight, and possibly only sighted bias. I doubted I'd be able to pass a blind test.</p><p>But I wanted to make the supertweeters <i style="font-weight: bold;">as good as they could be</i> before trying the Acoustats 2+2's with them completely removed, which I knew was going to have a lot going for it.</p><p>When I had previously (and for over 11 years) using Acoustat 1+1, they were narrow enough that I could place the super tweeters and their stands on the outside of the Acoustats. In that position, there was very little negative effect...perhaps even a slight positive, from having the extra stuff on the outside. It seemed to make the image even wider from left to right. But because of the added width of the 2+2's, and my need for a pathway through the room from the front door to the kitchen, meant that I had to put the supertweeters on the inside, if I was to use that assemblage at all. I knew from the start it was acoustically problematic. But super tweeters are so me. I didn't want to give them up.</p><p><b>But now that I've done so, there's no going back. It's clear to me from 4 years of listening to the previous arrangement that taking the supertweeter "towers" out of the way opens up the center of the image. The effect sonically is much the same as it is visually. Moving the super tweeters out of the way opens up the complete image. Previously it was parcelled and often cartoonish. Now it is expansive and every little bit from left to right and front to back can be mapped out. It has more slam too.</b></p><p>This <i style="font-weight: bold;">opening up </i>is even audible in other rooms where the stereo image, as such, isn't.</p><p>The problem as I suspected all along was that the heavy 40 inch sand filled target stands, and LS 3/5A boxes bearing super tweeters, was just getting in the way of the natural dipole response of the Acoustats, complicating the natural 3d line source approximation with lots of spurious reflections.</p><p>So back to the drawing board on super tweeters, in more than one way. I managed to snag 3 more of the high technology Tymphany NE19VTS-04 tweeters I was using in the backs, anyways, on ebay, they were unobtainable in stock anywhere else. I bought them as a Vifa product (at Madisound, IIRC, who doesn't even list them now). Later they appear to have been also sold as Peerless. It's Tymphany either way. The Dynaudio D21AF's in front are probably too heavy to deal with in future, and about as unobtanium as it gets (but miraculously, just while writing this paragraph, I managed to snag a replacement unit on eBay, so hopefully my loss is now corrected, I will still have a pair of working D21AF's for comparison if nothing else. And, btw, in comparisons the Tymphany's seem about as good as the Dynaudio's, at about 1/20 the weight, a true engineering miracle that it seems nobody else but me has recognized (they have essentially flat on axis response to 40kHz...you want at least that or there's no hope at all. I always suspected the D21AF's would handle any power available from my 150W amp, but in this situation both D21AF and NE19VTS died in the same catastrophic situation...I'm still not sure if the amp failed or what, perhaps just a full signal digital signal combined with all that gain I was applying, was just too much, I need to be more careful about that, also thinking of using smaller amp, 50W would be sufficient).</p><p>I plan a new supertweeter perhaps mounted only in the back of the Acoustats, possibly firing backwards only (and timed to arrive with the front) or both ways, on a strap of metal bent to fit under the Acoustat base board to the middle of the transformers.</p><p>I honestly believe the supertweeter effect is greatest in the reverberant field anyway, and not in the audiophile obsession--leading edge transients. Otherwise there would be almost universal discontent with super tweeters. Many of the leading edge tweeters (Scanspeak comes to mind) have pronounced peaks above 20kHz, which presumably extends the ultrasonic response much as I was doing. Now if this were all about the leading edge, it would be almost impossible to get it exactly right. I tried very hard (using ARTA program measurments) to get it as exact as possible, and most supertweeter users have never done anything like that. And move your head 1mm and it's different, the supposed leading edge may start out inverted, etc. Anyway, I think it's much more about the reverberant field reflected sound that leaks into your ears from the side. It's not that critical, down to the 0.1mm and so on, to get the supertweeters exactly coincident, and impossible in most cases anyway.</p><p>In actual soundfields, leading edge transients are rare. All is buried in the jiggling ambient noise.</p><p>What adding a super tweeter does is make this ambient noise <i style="font-weight: bold;">sharper</i>. That actually makes it sound softer because it's simpler.</p><p>Anyway, what counts is getting the supertweeter added sharpness within a few ms of the actual signal. Probably even 7ms (about 7 feet in distance) or so from back reflections is fine, but I could also try compensating it for simultaneous first arrival at the listening position. I could also try a separate front firing unit, with a different delay...</p><p>Anyway, for now I'm enjoying a more wholesome sound, full size players, wholistic, musical, etc., from not having any super tweeter stuff in the critical area in between the speakers.</p><p>You might say, I should have tried this before.</p><p>Update on Friday.</p><p>I haven't yet moved the LS 3/5A's based supertweeter assemblages completely out of the living room, which I presume, following ancient Linn advice (yes, no kidding, I first heard this in the day from the Linn Rep where I worked in 1977, though I honor it roughly in the breach) will make it just that much better still. Also, I need to get it out of the way for appearances sake. And it will take some doing because every place where equipment of this value could be put is already filled up. Especially inside the house, I need to carve out junk of lesser value and do something with it. Another thing I observed on my trip is that it's critically important to get on with the transfer or disposal of junk equipment or you'll still be stuck with it when you're too old to move it very easily. So I've got years of work cut out for me here, it looks like. And perhaps it's a good excuse, just about as good an excuse as owning cats, to be getting up and moving around now and then. So I've moved all the new odds and ends to be investigated onto the Coffee Table in the living room, and now I can move the LS 3/5 A's (which are so heavily modified as to be not really that anymore, they combine D21 with B110 and external driver connections and internal magnet shielding...because I was previously using in proximity to a CRT TV in Kitchen. But they originated as now extremely valuable LS 3/5As, and I believe I still have enough parts to make them all whole again and like original except a capacitor or two which might have been lost from the crossovers. Which I probably still have, if I could only find it within dozens of potential boxes, often hard to get to. So that's how things go. This time around, I thought they made a dandy acoustically solid base for the supertweeters, and they did, except too large and massive and complicated when positioned in between the speakers combined with the similarly large Target 40 inch stands. I'm now convinced the best supertweeter arrangement is on axis in back of the Acoustats, and possibly back firing alone is sufficient combined with compensating anti-delay (delay for midrange and bass only) for the first bounce (to and from the back wall). That, combined with a new lower powered amp and/or protection/monitoring system, etc, will be the NEXT GENERATION supertweeter arrangement. AND, I should probably wait about another year or so before deploying, so as to have the current sound firmly set in my mind. AND I've needed a pause in supertweeter deployment anyway (and bear in mind my supertweeters are truly super...I hear nothing coming out of them as such...I only believe they have a beneficial effect on how other things sound) so I can optimize the actual audible part of the HF spectrum. Back in 2021 I deployed a chairside EQ and began experimenting with midrange EQ's (which are now dialed into my midrange unit) that went up to a cut around 12kHz. The measured effect in pink noise was a 3dB/octave rolloff which became noticeable around 2kHz, but instead of just being a limited Linkwitz/Gundry "dip" it was a slice, basically continuing to 17kHz (at which point the Acoustats swiftly drop off).</p><p>This is dialed in as a set of 3 PEQ's actually, each one of which was tweaked by ear. But such tunings are never certain. I've long wanted to re-analyze the possibilities. I may be rolling off too much of the audible highs, long before it even gets to for-me inaudible highs above 16kHz. So you would think I should optimize what I can hear before bothering with what I can't, but my practice of audio, which presumes the possibilities of apparent subliminal effects (there has been controversial and unreplicated research showing brain wave changes when the bandwidth is increased from 20 to 40 kHz). But then perhaps too much I waste my time on things which might actually be of little value compared to the things that may well (and objectivists would believe) have important value.</p><p>So much as I keep wanting like a junkie to get the supertweeters back up again somehow, with some combination of bubble gum or something, I should resist that temptation, and even set make a rule that I won't again try super-tweetering for about a year, both to get an all new better and safer approach up and running, and to familiarize myself with the sound without it, and to get the audible parts of the Acoustat EQ understood if not dialed in better.</p><p>So I think I'll choose Halloween 2024 as the first date I will permit myself to run the supertweeters again, in some new setup, hopefully a much perfected one, with no extra stuff in between the speakers, and just a little bump above the back transformer, and a working power limiting system to prevent future burnouts, and probably a different amplifier (in my own collection I might try the 60 watt Marantz 15b which has 'just enough' bandwidth and presumably Sid Smith got the internal slewing and limiting right, this amp is legendary for good sound. I've always wanted an excuse to use them. But I also keep lusting after the few $200-300 Chifi Class A amps still available on ebay.</p><p>Whatever amp will probably be right out front and center where the Aragon 8008 BB is now. I've done enough A/B'ing to say there's no audible difference between the Aragon, as currently biased (on the high Class AB side: about 28mV instead of Klipsch specified 12mV across the collector resistors) and the Hafler 9300's. And under no situation is the Hafler underpowered. The speakers begin to sound dynamically limited at exactly the same point with either amp. So I don't need the greater (measured) 600W (spec 400W) into 4 ohms that the Aragon provides, the 350W (spec 250W) that the Hafler provides is sufficient. And the Hafler 9300 is better in every measured way, 0.002% THD instead of 0.02% for example, and even better HF damping factor (it was even better than the Krell FPB 300). The Trans Nova circuit in the 9300 is about as simple and fast as it gets. However, Parasound HCA-1500A is almost identical, and slightly superior in near clipping distortion levels. With the relays and stuff it's just less of a 'perfectionist' amplifier IMO than the 9300. But winning amplifier of earlier epochs basically need not apply. Before 1982 or so it was impossible to make amplifiers this good, then better parts made it easy.</p><p>The problem is, I'm not sure I want to sell the 8008 BB, and I've got no other place to put it, certainly not without removing the 2" spiked Mapleshade carpet feet.</p><p>I'll keep the A/B setup, but only temporarily put other amps to the side during A/B testing, rather than in front, which will be dedicated to the supertweeter amp in future setups starting next Halloween. (I also have parts to build a Pass Class A...an effort which now seems unimaginable. The whole purpose of the Pass amp was for the super tweeters in the first place. I have years worth of repair work to catch up on first. And at the current rate, they will be many years more before much is done.)</p><p>I've got plenty to do until then to give the supertweeters very much priority. I've now got 3 essential but dead DEQ's, all needed a power supply refurb. I should get the first one going to get back to having the chairside EQ which was so helpful to EQ experimentation.</p><p>I need to get things moving through "the laboratory" again and back working again. There's an incredible pile of to-be-fixed stuff in there now.</p><p><a href="https://www.schiit.com/products/aegir?gad=1&gclid=CjwKCAiAxreqBhAxEiwAfGfndF6oZlOlBMdJplVPFUZMyaH6SCTLDeuAVscGdiO3_JkecFrYcTvX-RoCAg8QAvD_BwE">An alternative to the Chifi class A amplifier would be Schiit Aegir.</a> The power is just right. I'm not going to consider such a thing until I've sold an equivalent amount of stuff on eBay, starting right now. But it looks pretty cool. The design goal is correct, there is really no need for a class A transistor amplifier. But high bias is generally good (this amp is quite high bias for the power, and features their special circuit which sounds good in the description and the Amp scores well at Audio Science Review...in fact about the best thing I can see for the price, as it is surrounded by $5k amplifiers in the upper end of the ratings). The upper bandwidth is huge and there are no coupling capacitors or servoes. The one non-purist thing is a relay, but those are fairly ubiquitous, and this one is CPU controlled.</p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-28060054528147364822023-10-07T08:59:00.005-07:002023-10-07T12:40:34.679-07:00Creating Music playlist and playing in RoonHere is a 30 second video showing me clicking on the "MakeMusic" script icon (built upon my universal playlist generating program <b>mplay) </b>which generates a playlist called Music (it takes about a second, the script window stays open for 5 seconds so you can see it worked) and then playing this playlist in Roon.<div><br /></div><div class="separator" style="clear: both; text-align: center;"><iframe allowfullscreen='allowfullscreen' webkitallowfullscreen='webkitallowfullscreen' mozallowfullscreen='mozallowfullscreen' width='320' height='266' src='https://www.blogger.com/video.g?token=AD6v5dyEJycoE_M8DdaiULkhVkU-YnSzEFBRTEkuyLDRDLXtak-iQ4CzaO2W14vTAU5jC60tPKpa9zCBmAaLkLYmYg' class='b-hbp-video b-uploaded' frameborder='0'></iframe></div><br /><div><br /><br /><div><br /></div></div>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-74992381879564921242023-08-23T12:21:00.001-07:002023-08-23T12:21:11.338-07:00mplay Generation 2: playlist generating programs<p>I've been working on automatic playlist generating programs now since January 2021.</p><p>I originally had just one program, <i style="font-weight: bold;">mplay</i> (make playlist).</p><p>Now there are more programs: <i style="font-weight: bold;">splay </i>(shuffle multiple playlists together), <i style="font-weight: bold;">tplay</i> (tell about what actually played, and truncate playlist or play history), and <i style="font-weight: bold;">shufflelinks (</i>shuffle multiple "playlist" folders-of-links together).</p><p>These programs can be compiled in scripts, making playlist generation either more intelligent (such as removing the items that didn't actually play from the playlist history before creating a new playlist) and/or combining playlists (or folders of links that serve as playlist for programs that don't handle playlist) together in different ways. (Playlists can also be simply concatenated together using the system command <i style="font-weight: bold;">cp</i> or equivalent so I didn't write a program for that.) </p><p>Also <i style="font-weight: bold;">mplay</i> itself has advanced considerably since the 2022 software release on Sourceforge. I hope to release this new version before the end of 2023, <span style="font-size: xx-small;">possibly fixing some of the current "gaps" between what it does and what it really should do.</span></p><p>One key new feature is the ability to specify ALL of the files in the specified folders. As in "just make a playlist with all of the files in these folders." You don't have to guess or predetermine the total number of files that would be included. Since the playlist history is unchanged by this operation, it is neither read nor updated (<span style="font-size: xx-small;">however, it might be better if it reads to playlist history first, to put unplayed items at the head of the playlist, currently it just ignores if itms have already been played or not</span>).</p><p>You can also apply an file age criterion in order to make a playlist consisting of only files newer than a specified number of days. That can be combined with the ALL option to easily create a playlist of ALL the new files. A script can concatenate this <i style="font-weight: bold;">premier</i> playlist at the beginning and/or end of a main playlist. (<span style="font-size: xx-small;">Another useful feature not yet implemented would be to make a playlist of ALL the as yet unplayed files.</span>)</p><p>But I've found it useful also to mix the new files into a playlist of older files. There are two basic approaches to this kind of mixing:</p><p>1) <i style="font-weight: bold;">Dense Shuffling</i> selects one (or some specified number) of items from each playlist, which are then ordered randomly (or not) into a new segment of the output playlist, with additional segments added until all the items from every playlist are included. If any playlist is exhausted early, it is reset and reshuffled. This works very much like <i style="font-weight: bold;">mplay</i> itself (though currently only for non-audio files, since audio files are currently treated as if they were all in one big folder, with no other option).</p><p>2) <i style="font-weight: bold;">Sparse Shuffling </i>mixes a smaller playlist into a larger playlist so that each item in the smaller playlist is included exactly once (or some specified number of times) in the end result, using random placement.</p><p>Currently splay does only Dense Shuffling, and shufflelinks does only Sparse Shuffling. It would be good to have both available in both programs, and/or to combine the programs to work on any combination of playlists and links-folders, to generate either a new playlist or links-folder.</p><p><i style="font-weight: bold;">tplay</i> is very useful in the case where I've been playing an audio playlist in Roon, but switch to playing other items. If I go back to the playlist in Roon, it does not remember the last item played but simply starts all over from the beginning. Using tplay, I can remove the items that have already been played from the playlist. Or, I can remove the items that were not played from the playlist history and create a brand new playlist.</p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-48493595813549159842023-07-15T11:58:00.003-07:002023-07-15T12:25:00.797-07:00Thinking about HDCD again<p></p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhBfZDJH3sA8zbIacIFRjfKzQZlU5UlEm_4VEAJlrhvU_oyiH8MTA1EFY02TdrOcOHM5Y_YK1QQqXouLIU9INVKaktThP-OdJqRiFDxNEYgY7YRuJsCQrJX8YjFYxV09LNPrL-NtJVFR3-8LMwGEur70Vqi2eUt9TnK0bNIC3u9uddtnWK1zVwpMjxdylU/s4032/IMG_1151.jpg" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="3024" data-original-width="4032" height="480" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhBfZDJH3sA8zbIacIFRjfKzQZlU5UlEm_4VEAJlrhvU_oyiH8MTA1EFY02TdrOcOHM5Y_YK1QQqXouLIU9INVKaktThP-OdJqRiFDxNEYgY7YRuJsCQrJX8YjFYxV09LNPrL-NtJVFR3-8LMwGEur70Vqi2eUt9TnK0bNIC3u9uddtnWK1zVwpMjxdylU/w640-h480/IMG_1151.jpg" width="640" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Using DVD-5000 to decode HDCD on played on PD-75 as digital transport</td></tr></tbody></table><br /> I've been testing a pair of the DVD-5000's I originally bought for use as living room DACs. They are both for sale on ebay as I write this. I am reminded of the fact they are very good, and according to objectivist standards should not be audibly different from the very best DACs. They also have the ultimate R2R chip made by Burr Brown, the PCM 1704, which was discontinued in 2012. By objectivist standards, it should be no different than a good Sigma Delta chip. But it is <i style="font-weight: bold;">different</i> in objective ways, for what that's worth. I love the DVD-9000 for HDCD decoding but don't use it for anything else. I would have been tempted to use DVD-9000 for midrange DAC until I discovered that I needed 3 DACs of exactly the same design for my tri-amplified system. Even if I adjusted the delay times for one particular sampling rate, when I changed sampling rates those adjustments would no longer be valid, because the latency of nearly every device varies with sampling rate.<p></p><p>Faced with that problem, and the (at the time) skyrocketing price of DVD-9000's, I opted to get 3 DVD-5000's instead. Those would be basically the same thing, I predicted.</p><p>But for probably no good reason, I just never fell in love with the DVD-5000's the way I did for the DVD-9000. And it was much more convenient to have multiple DACs with level controls, and smaller. So I ended up with 3 Emotiva Stealth DC-1's, which are quite fine also (technically they measure much better than the DVD-5000's, but once again that difference should not be audible).</p><p>I still use DVD-9000 for decoding HDCD's (because my Oppo BDP-205 doesn't do that). So I tried using the DVD-5000 (which doesn't actually play discs, but connected as a DAC to the digital output of a Pioneer PD-75, optically) and I found it does indeed do very well with HDCD's, just like my DVD-9000.</p><p>But it is different. When playing an HDCD the output level never exceeds the output level from an ordinary CD. The DVD-9000 actually plays the level expanded portions of an HDCD up to 6dB louder than an ordinary CD. In contrast, the DVD-5000 does what every other HDCD player I've ever tested (other than the DVD-9000), it lowers the average level, just so so the most expanded peaks of HDCD reach the maximum CD level, but no more than that.</p><p>When my brother-in-law George first heard the HDCD on a DVD-9000, he was (unusually I'd say, because he never matches levels by <i>measuring) </i>shocked and appalled that that HDCD boosted levels 6dB above normal CD levels. He felt that was unfair. HDCD was all just a cheat, he insisted.</p><p>Well, as I've started to explain in many other previous essays, it's much more complicated than George and almost everyone thinks, because of inter sample overs.</p><p>(And of course, George was also wrong that HDCD players boost the level beyond that of CD's. Though for the longest time I wondered if earlier HDCD players did do that boosting, and it changed when players stopped using the PMI chips. I had it reversed. The DVD-5000 uses PMI chips and doesn't do boosting beyond CD levels. The DVD-9000 uses a software implementation--like all later players--and does boos the level. So it does not necessarily have anything to do with using the chip, it was just a particular approach the designers of the DVD-9000 took, which possibly was available to the users of the chip as well, but I haven't seen one that does.)</p><p>The ISO's on regular digital recordings can match those of even peak level expanded HDCDs. And what's more, HDCD's don't seem to have such as big ISOs. So it comes out about as a wash, the HDCD is just giving a kind of engineered peak, and regular CD's are giving us extrapolated peaks. That dynamic range in the analog output was there and needed anyway, HDCD's (when boosted) are just using that dynamic range for real music dynamics, rather than extrapolations from what may be just high frequency ringing.</p><p>Now I also know, that SACDs may in fact be the worst offender of all. They can have the highest peak output above the nominal 2V level.</p><p>But I'm wondering if it was the boosted HDCD level (compared to other players) that led me to fall in love with the DVD-9000 in the first place.</p><p>(I think HDCD done as intended (and with the filter control*) would have been a great idea...as an open standard, and would have enabled a way to bypass the loudness wars, using the HDCD encoding to provide the uncompressed version which for which the compressed version would be heard without it. Now we're simply stuck with HDCD as necessary for reproduction of a significant number of fabulous sounding recordings, which are better heard when available in high resolution.)</p><p>(* The filter control somehow seems less necessary as HDCD's don't seem to have the transients that provoke excessive ISO's. Somehow they achieved that effect without a post-filter change. It will be necessary to do more investigation.)</p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-66037097521378814362023-07-07T20:03:00.001-07:002023-07-07T20:03:30.684-07:00What's going on with Poulenc in DSD<p>I've always loved the Linn recording of the Polulenc Concerto for Organ. There is no question, it's very dynamic (as a whole album, much less so in individual tracks). The SACD layer has a marvelous sound. I have not listened to the PCM layer in awhile.</p><p>It has by far the highest peaks above the nominal 0dB RMS level of any disc I have encountered (+7.5dB).</p><p>I have been accounting for these peaks as Inter Sample Overs. That might not be the correct description of what is happening on SACD's.</p><p>On PCM recording, the highest peaks seem to be associated with the leading edges that produce the most pre- or post- ringing. This makes sense, as they basically represent very high level high frequency content which 'propels' an interpolation of the signal to go way beyond the normal boundaries. Only high frequency content can do that. Even when reaching 0dB, low frequences just reach the top slowly, over very many samples, and don't change enough from one sample to the next to cause an ISO. (That was why I measured 0dB with 880 Hz, a low enough frequency not to produce ISOs.)</p><p>But on DSD...there is no pre- and post- ringing. That is of course the beauty of it. You get smooth looking curves that visibly look like the original wave forms. (The eye is being deceived. In actuality, those smooth looking curves are obscuring vast high frequency noise, which is often making the curves look smoother than the real thing.)</p><p>Here the entire Poulenc album on the DSD layer, recorded at 96kHz. Notice that the very highest peaks which give the album very high dynamic range only occur in the last track. Just a couple other tracks have higher peaks at all.</p><p><br /></p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiHeVdWq21kc3qNtVjjxtMaR-togszai2H_bLJ05SprZ8eGYooXPwc4btVVKMhYPVNC6xEKfgla8j_w7HUMVSzZf4J6rYmHB80JDwwHOspJJfD0aEnxtW7oiQ-45G_MaJeqEp9Dz2t8pphSzgDIGZflgGSUg7B0vAK4J2Ma4picF3wLTxHAE9iHj6Qg1kU/s1920/Screen%20Shot%202023-07-07%20at%209.27.37%20PM.png" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="360" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiHeVdWq21kc3qNtVjjxtMaR-togszai2H_bLJ05SprZ8eGYooXPwc4btVVKMhYPVNC6xEKfgla8j_w7HUMVSzZf4J6rYmHB80JDwwHOspJJfD0aEnxtW7oiQ-45G_MaJeqEp9Dz2t8pphSzgDIGZflgGSUg7B0vAK4J2Ma4picF3wLTxHAE9iHj6Qg1kU/w640-h360/Screen%20Shot%202023-07-07%20at%209.27.37%20PM.png" width="640" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Poulenc (entire album)</td></tr></tbody></table><br /><p>Now lets take a look at what may be the highest peak (there are actually a bunch of them as you zoom in) in the last track:</p><p><br /></p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgl_ozoGkXnf5c_pabJ8YJSws3pNFlFNfIMrGRz8Ogw-kB-kcsjnTHf9_LTPrXxJS_YsJcoWE73x1d8VzDj7f-ju4s60gePHsd8g5H8JEgajP35Goi_XIRGp9AxzKykt5gQTmlqCxrXWX43MF8jL69vCte6GUGwvHEmN4MkDU5ShwJMfCWRvl8dJ2V5czo/s1920/Screen%20Shot%202023-07-07%20at%209.34.02%20PM.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="360" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgl_ozoGkXnf5c_pabJ8YJSws3pNFlFNfIMrGRz8Ogw-kB-kcsjnTHf9_LTPrXxJS_YsJcoWE73x1d8VzDj7f-ju4s60gePHsd8g5H8JEgajP35Goi_XIRGp9AxzKykt5gQTmlqCxrXWX43MF8jL69vCte6GUGwvHEmN4MkDU5ShwJMfCWRvl8dJ2V5czo/w640-h360/Screen%20Shot%202023-07-07%20at%209.34.02%20PM.png" width="640" /></a></div><br /><p>Mostly, it doesn't look that extraordinary. There's no sharp edge, no pre- or post- ringing. However, there is an interesting notch on he bottom channel at the very peak (near the cursor). Given the scale, that represents very high frequency information. It may be a telltale sign that some sort of limiting or other thing was done just here. But the notch itself is tiny in comparison with the overall peak, which doesn't look out of place in its surroundings.</p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-9190654072206004932023-07-04T10:16:00.008-07:002023-07-06T12:11:42.729-07:00Dynamic Range in Recordings is not a well defined concept<p>Dynamic Range in audio equipment seems relatively easy to understand. It is the difference between the peak level and the noise. </p><p>OR, it's the diffrence between the peak level and the lowest resolvable signal, which can sometimes be 20dB or so BELOW the noise level. (Signals can be resolved from noise using spectrum and/or <b><i>gestalt analysis</i></b>, and the human auditory process does both.) This is a little less well defined. How capable is the spectrum analysis? And the gestalt analysis is not very well defined at all. Gestalt analysis is how we can identify discriminate different sound sources by other aspects of their quality than frequency, such as by their rhythm or randomness or apparent location.</p><p> Either way, dynamic range in audio equipment is now specified as being done in the presence of signal, thereby requiring the signal to be filtered out from the product before analysis. This is not hard to do now that we have ready access to things like FFT.</p><p>Sometimes this kind of Dynamic Range is called <i style="font-weight: bold;">Signal to Noise Ratio</i> in the context where we are looking at the noise of a particular component instead of an entire system. And also perhaps when were are not measuring it in the presence of signal, which was the traditional way of measuring it (and what I usually do on my test bench).</p><p><b>But where it really gets thorny is when we are talking about the Dynamic Range in audio recordings.</b> It bugs the heck out of me when people just don't get how poorly defined this concept is (even though we now have 'Standards' and tools for measuring it) and how much, therefore, how much it depends on assumptions, heuristics, algorithms, psychoacoustic research, and the like.</p><p>For example, you could rightly claim that nearly every DDD recording has at least 96dB dynamic range, if you defined dynamic range as the difference between the peak level (which is almost always near the maximum level, defined as <i style="font-weight: bold;">0dB</i>, on digital recordings) and the lowest level in the recording (such as in a fade out near the end of a recording). </p><p>Also, since sound is wave-like, and audio is based on alternating currents, there will be zero crossings in every small section of recorded audio. So it's always going down to zero <i style="font-weight: bold;">somewhere</i> and actually <b><i>quite frequently!</i></b></p><p>Even if you removed silence-before-the-first-downbeat, silence or mechanical fade outs, and zero crossings, you may often find tiny near-silence gaps even within in highly <i style="font-weight: bold;">compressed</i> program material. Even ruling out the two considerations above (fade outs and zero crossings) we can just look at the low level those relatively small gaps (for example 250ms, enough for 250 cycles at 1000 Hz and about where we hear gaps as gaps and not just clicks) and say "see, there is plenty of dynamic range between these 250ms intervals and the peak levels."</p><p>What the whole family of standards, methodologies, and tools for measuring "Dynamic Range" in recordings is really about is determining how much compression has ruined particular recordings, or alternatively how much compression can be used without making things sound bad.</p><p>It's a matter of heuristics. They do this by comparing some quantity of peak levels to some quantity of less-than-peak levels.</p><p>But the quantities involved are not something that it intuitively obvious, but rather related to what is found to be 'interesting' in that it correlates to things sounding over-compressed.</p><p>So it is that some Dynamic Range methodologies use <i style="font-weight: bold;">histograms</i> and others use <i style="font-weight: bold;">percentiles</i>, things that are not intuitively obvious to self appointed audio gurus.</p><p>For example, the famous DR scale is based on a comparison between the peak level (fine, that's intuitive enough) and the average of all levels that are in the 20th percentile of highest levels or above (rather counter-intuitive). IOW, it's comparing <i style="font-weight: bold;">loud</i> vs <i style="font-weight: bold;">loudest</i>, whereas you might think that dynamic range should compare loudest and softest (but then we are back to the 96dB of digital recording systems, etc). But the presumption seems to be that "almost of the recording is at or below this level" and it should have at least one much higher peak to be "dynamic." It seems to me there are other ways of being dynamic.</p><p>Why not the 50th percentile, or the 80th percentile??? Apparently because choosing the 20th percentile makes the tool identify recordings that <b><i>sound </i></b><i style="font-weight: bold;">overcompressed</i> as compared to the genre of recordings they come from (which score differently even if no compression were used).</p><p>Furthermore, the DR tool analysis uses short non-overlapping segments of 3 seconds each. You could get significantly different results if you defined those segments as shorter, longer, or overlapping. You could argue for the merit of 250 msec intervals, which could produce a radically different result. Likewise for 10 second intervals.</p><p>You might also think that something like Equal Loudness spectral curves (such as the famous <i style="font-weight: bold;">Fletcher-Munson)</i> ought to be applied...and indeed some tools use that...but the DR tool does not.</p><p>I was inspired to research and write about this after reading a poster (<i>Tank</i>) at Hoffman's discussion site talk about <a href="https://forums.stevehoffman.tv/threads/king-crimson-2009-re-issues-cd-dvd-stereo-5-1-all-inclusive-thread-part-3.266653/page-2">the difference in "dynamic range" between the 30th and 40th anniversary reissues of classic King Crimson albums</a>. I very very much like the 40th anniversary series, released on high resolution DVD-Audio, but this poster claimed the dynamic range on many albums in the 30th anniversary series was higher (and he like them better). (I have not listened to the 30th anniversary series, and it's not "high resolution" so I might not bother.)</p><p>It really bugged me that this simpleton poster pushed back against a recording engineer (<i>Plan 9</i>) who was arguing with him, claiming that the dynamic range issues were easy to understand and the engineer was just full of BS. The poster showed tiny graphs of different songs and said "See!" But anyone who has done any amount of audio editing knows that things can look very different at different levels of zooming in. </p><p>Sadly the engineer didn't do a great job spelling out his case, and after being denounced by Tank he just shut up. But he did mention <i style="font-weight: bold;">micro-dynamics</i> and <i style="font-weight: bold;">macro-dynamics</i> and as far as I can tell, these are real things and not just BS. Right now I couldn't define either one and I still need to read and understand more in this area.</p><p>[I previously wrote similar but less complete comments about the R128 used by Roon just a few posts back. R128 does make slightly more sense to me than DR, perhaps just because I misunderstood DR. It still seems to me highly arbitrary, though Roon's system works pretty well in practice for keeping a constant "level," which I noted then is not immediately obvious to me how that is done from the R128 dynamic range rating which is about range and not level.]</p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-7715371181766166912023-06-30T11:08:00.008-07:002023-07-05T23:01:50.333-07:00Inter Sample Overs vary with digital process ?<p>[Correction: Subsequent data has called most of the theoretical speculation below into question. It turns out that SACD's do not have consistently low ISOs, they can possibly have the highest ISOs of all, as I detected with the the 'Poulenc' SACD released by Linn. This is an orchestra with a large pipe organ, and if I had associated low ISOs with higher quality, let alone previously believing all SACD's to have low ISO, I would have believed this to require a minimal 2.0dB headroom for ISOs. But in fact it set my current record...it required a a mind boggling 7.5dB headroom. And it was somewhat inconsistent. It might work with as little as 6dB headroom....or it might not. It apparently produces a peak around the same size as the 96kHz sampling interval, and how high it registers depends on how much of the peak occurs in any one sampling interval. So there is a large random element to how much headroom is required. I've also noticed that on some DVD-Audio discs with 96kHz sampling rate.</p><p>It does still seem that Reference Recordings and 'Discipline' (the brand behind the 40th Anniversary King Crimson DVD-Audios) produce consistently low ISOs. I would venture they seem to know what they are doing. But the peak ISO level seems not to depend on DVD-Audio vs SACD. Nor does it even have a fixed relation to music genre (rock vs classical) or does it necessarily have anything to do with the recording "Dynamic Range" or how much compression was obviously used---though it's possible some compression was slipped into the Poulenc at key moments and that's what gave it such high ISOs.]</p><p><br /></p><p><br /></p><p>I have begun fairly systematically copying all my SACD's and DVD-Audios into 24/96 copies for my hard drive. And I'm noticing a few weird things.</p><p>Remember that I set my gain structure so that with an 880 Hz tone recorded at precisely 0dB (generated by Audacity) a +4 level would be the highest possible before clipping.</p><p>But for actual recordings, I have to set the level between -2.5dB and +2.5dB, giving them at least 1.5dB extra headroom and as much as 6.5dB appears to be needed in some cases. This headroom is required for <i style="font-weight: bold;">inter sample overs </i>(ISOs) where in between the samples the signal peaks higher than 0dB.</p><p>Benchmark claimed that ISOs could be as high as 3.01dB and that many digital decoders failed to allow sufficient headroom.</p><p>BUT I am finding 3dB headroom is way insufficient.</p><p>EXCEPT, in some cases it isn't. Some discs seem to cluster around 2dB headroom required, up to 2.5dB in some cases. Which discs are these?</p><p>1) Reference Recordings HRx (174kHz/24bit) seem to only require 2-2.5dB headroom. These are some of the best sounding recordings ever.</p><p>2) SACD's generally only seem to require 2dB headroom (I would have expected SACD to be the 'wildest,' but in actuality, it's the 'tamest.')</p><p>3) The DVD-Audios in the King Crimson 40th anniversary DVD-Audio boxes, which seem only to require 2.5dB headroom. These are truly spectacular in audio quality (or in the case of Court of the Crimson King, merely way better than ever before).</p><p>On the other side?</p><p>1) Many hot sounding recordings on DVD-Audio, including Elton John, and especially Steely Dan. (I've long noticed that. I figured Steely Dan cranked up the compression so high it's bleeding out the ISOs). </p><p>I wrote this off for awhile, but now I'm getting a pretty clear feeling that the recordings which require the least ISO headroom are the ones that sound the most natural, laid back, and 3D. The recordings that require the most headroom sound highly processed.</p><p>You may be astonished to see some of the pictures when I post them. Having the huge ISO's means the rest must be scaled back in a recording, losing dynamic range in the midrange.</p><p>I think the excessive ISO's occur when the the anti-alias filtering is insufficient, and full scale high frequency garbage gets into the digital encoding. THAT's what's causing so much overshoot.</p><p>DSD has such a high rate it captures and controls all the HF crap by design. So SACDs do not suffer from excessive ISO's.</p><p>Likewise, the PMI analog to digital encoders used by Reference Recordings, which must have superb filtering.</p><p>And whatever King Crimson was using in the 40th anniversary DVD-Audio set.</p><p>Now, it would be interesting to know what Steely Dan used in such things as the Everything Must Go DVD-Audio, that produces such high ISOs.</p><p>Perhaps it's not the digital process, but the amount of processing (including compression) used before digital encoding that is the culprit. But the biggest ISOs look too big for just that, IMO. Still, the sonic variation might just be coincidental, the least processed recording just happening to use the digital processes which produce the least ISOs.</p><p>This is also a function of the Oppo BDP-205, which is apparently not headroom constrained itself. But I've found the height of the ISOs not to change with different reconstruction filters. I think they are inherent in the data and and the degree of oversampling used, with higher amounts of oversampling 'revealing' the true ISO height (because you are filling in more of the points in between the points).</p><p>******</p><p>Update:</p><p>Now it appears that that the ISOs in DVD-Audio discs are all over the map. They can be < 2.0dB, for example, in Queen's <i style="font-weight: bold;">A Night At The Opera</i><i style="font-weight: bold;">.</i> They can be < 2.5dB such as in every King Crimson 40th Anniversary DVD-Audio box. Or, they can be as high as 6.5dB, as in Steely Dan's<b> </b><i style="font-weight: bold;">Everything Must Go, </i>Elton John's <i style="font-weight: bold;">Goodbye Yellow Brick Road</i>, or Frank Zappa's <b><i>QuAUDIOPHILIc.</i></b></p><p>SACD's still appear to be consistent at around 2.0dB, and everything from Reference Recordings is there too.</p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-87360697420533217142023-06-08T13:03:00.000-07:002023-06-08T13:03:05.140-07:00The Audio Hobby<p>The audio hobby means different things to different audiophiles, and isn't that wonderful?</p><p>Not always, perhaps. It can be made awful in many ways. People can fall victims to unfounded beliefs that cause them to waste time and/or money. It can lead to the virtual inability to listen to music anymore, in extreme forms of <b><i>audiophilia nervosa.</i></b></p><p>It can lead to endless <i style="font-weight: bold;">bullying.</i> I think generally the best attitude is non-judgemental: <i style="font-weight: bold;">live and let live.</i> However all the same I believe audio is filled with frauds of many kinds, both on the big and small levels. You probably already know I side mostly with the <i style="font-weight: bold;">audio objectivists</i> as to what kinds of audio beliefs are founded and which are not. All the same, I don't take it as my mission to change anyone's mind. I suppose, in some cases, even I could be wrong too.</p><p>Anyway, nowadays I think generally it's a waste of time to compare good amplifiers, DACs, cables, power conditioners, or anything of that ilk.</p><p>All have been tested endlessly by audio objectivists getting only null results. With my lack of patience, I'm unlikely to do better, if I follow all the proper procedures. If I don't, then the result may be meaningless anyway.</p><p>My normal result with good amplifiers is this: I start the auditory level matching process by matching the apparent loudness with both amplifiers to make it identical in fast A/B testing. Once I have matched the level, I attempt to match the <i style="font-weight: bold;">quality</i> of the sound. If one seems to have more bass, or highs, I assume that's because it's actually <i style="font-weight: bold;">louder.</i> In this second phase I make only the smallest adjustments, 0.25dB at at time (I'm fortunate to have a first generation Emotiva Stealth DC-1 with 0.25dB adjustment. That means, on average, the best case is within 0.125 dB of being exactly correct, which is close enough to pass the 0.1dB minimum in my experience.) Ultimately, on every amplifier I've tested, I can make them sound identical simply by matching the level that closely.</p><p>Now I did not do such procedures when I thought for several years my then go-to amplifier, the Aragon 8008 BB, was sounding a bit harsh. Back in those days, I found myself avoiding listening to the FM for very long. It drove me insane.</p><p>I later found the distortion had risen to 0.7% because of low bias. After bias adjustment, I got it back to 0.07% and sounding fine.</p><p>But I know that from years of experience and not A/B tests. And measurements which make that experience believable.</p><p>And that's another thing. One should often believe measurements, when they are meaningful and honest. Not necessarily specifications.</p><p>Anyway, distortion can be a factor down to 0.1%, so products having higher than 0.1% should generally be avoided. (In electronics, anyway, where it's easy to do better. There's hardly any speakers that can do as good as that.)</p><p>In audiophile land, there are often electronic products with higher than 0.1% distortion. And sometimes they <i style="font-weight: bold;">sound better.</i> I think what's happening is that in some cases products with predominantly 2nd order distortion may fix recordings that were made with high amounts of 3rd order distortion.</p><p>Furthermore, boosting the amount of 2nd order distortion, which tends to occur with zero feedback designs (feedback works best at suppressing 2nd order distortion) can add additional "spaciousness" and air to recordings lacking those things because of poor production.</p><p>Things like this may work on some recordings and not others. It may work best on recordings that are fairly simple, like a few instrumentalists. Not on works of great complexity, like a full symphony orchestra.</p><p>Euphonic adjustments are like that. Generally it's best to stick with low distortion, wide response, low noise, because that's the combination that works overall best on everything. Basically what the objectivists say.</p><p>Other claimed magic requirements in design, however, are sold on the basis of faulty comparisons, typically failure to match levels very well.</p><p>Anyway, if feedback free amplifiers, and electrically charged cables, or whatever makes sense to you, go for it.</p><p>The best is when we're not bullying people over such things, one way or the other.</p><p>Even being forced to make a decision is a kind of bullying.</p><p>We are generally not designed to discriminate among audio reproduction systems.</p><p>We don't have a 'memory' that works very well for making such comparisons. We don't store 'experience A' in anything like the raw form that would make for a good comparison with 'experience B.'</p><p>To be reliable at all requires, as the objectivists always say, instantaneous A/B switching. Other than that, perhaps exhaustive training.</p><p>Furthermore, always being assigned to make the equipment comparison detracts from the process of having the most enjoyable and enlightening experience from the music, appreciating the music itself rather than arcana of possible sonic differences caused by different audio equipment.</p><p>Fine, the uber subjectivists say, just see which piece of gear gives you that most <i style="font-weight: bold;">transcendent</i> audio experience.</p><p>That's basically impossible, because each time you listen to the same piece of music you get a <i style="font-weight: bold;">very different</i> experience. Symphony orchestras often like to prove this by playing a <i style="font-weight: bold;">Premier</i> (first ever) performance of some work <i style="font-weight: bold;">twice</i>, sometimes even without warning. Few guess it was an identical repeat. The identical music doesn't provoke an identical response. And for a very important reason.</p><p>The <i style="font-weight: bold;">you</i> listening to any work the second time is now older and wiser, having already heard the music before. The brain has already stored memories and made new connections. Usually that opens up entirely new realms of experiences. While closing down others.</p><p>(Curiously a work with more 'movements' can be heard more times without seeming repetitious.)</p><p>As many have often opined, that is what audio should be mostly about, <i style="font-weight: bold;">the music</i>, and quite often isn't.</p><p>(However in line with non-judgmentalism, I prefer to say that any mode of enjoying an obsession with audio reproduction is fine. If your thing is building amplifiers that test conventional theories -- fine. If your thing is arguing about whether such things can or do make any difference -- fine. As long as I'm not to much detained by your obsessions.)</p><p>But along those line, I fear the excessive denigration of standard audio engineering practices, as often occurs in the writing of audio subjectivists (including the one I always loved to read anyway, Harry Pearson) tends to promote rather than discourage <i style="font-weight: bold;">audiophilia nervosa</i> and therefore lack of being able to enjoy music.</p><p>But some people swim in one thing or the other, so whatever works for you. Doing 'comparisons' is also a way of just hearing things twice, which may itself be beneficial.</p><p>I have dubbed such demonstrations <i style="font-weight: bold;">magic shows</i>, and typically enjoy them even when (that is, all the time) my core beliefs are unshaken. </p><p>I myself only drifted into near audio objectivism in my late 20's, after doing carefully constructed experiments I felt would 'confirm' my beliefs in tweakdom. That's the path of many noted audio objectivists. First they were fully taken in, then they decided to do some tests.</p><p>After the recording, DAC, and amplifier problems are 'solved,' is there anything left? Two things, the loudspeaker/room (or headphone) interface, and the selection of the music itself. Neither is a problem that will ever be 'solved.'</p><p><br /></p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-19465879684521039682023-06-02T21:20:00.002-07:002023-06-03T06:54:11.935-07:00R128<p>Here is the EBU R128 standard for measuring "dynamic range" that Roon uses:</p><p><a href="https://tech.ebu.ch/docs/tech/tech3342.pdf">https://tech.ebu.ch/docs/tech/tech3342.pdf</a></p><p>Dynamic range can mean different things. In the context of technical measurements of an amplifier or transmission system, dynamic range is pretty similar to "Signal to Noise." What is range from the lowest signal that can barely be resolved to the maximum. So a dynamic range of around 120dB is where many of the best units are (chips can be as good as 130dB).</p><p>But this has nothing to do with they "Dynamic Range" of program material. But what has long bugged me is, what does this mean anyway, because every signal, no matter how high the peak, ultimately has to go back through zero again. And how close it gets to zero depends on how finely you can measure it, assuming it's a continuous waveform. So this is back to to the "dynamic range" of amplifiers again.</p><p>But this is not what audio/music people mean by the Dynamic Range of program material. Their window of analysis is not 1 uS, say, the resolution of a decent digital scope. Their resolution is the loudness in a 3 second time window which must be overlapping. The spec is unclear to me, but in specifying "dBFS" it is clear to me they don't mean instantaneous levels but something like average or probably RMS levels...that is to say levels related to the equivalent sine wave measured with RMS.</p><p>And then the Dynamic Range is specified as the difference between the 10th percentile and the 95th percentile of these 3 second time windows.</p><p>Now it's still not clear to me how Roon uses this in level normalization. There must be other parameters such as the maximum level, etc, because the R128 I've just described only relates to relative and not maximum levels.</p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-49201508254001565942023-05-26T12:21:00.000-07:002023-05-26T12:21:12.275-07:00My Simple Surround System<p></p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjXkPJgXiz3GBTRHSU_Oh2uYfpFcTnd55FlVgtSKG6LVUxcQQzlBs1fdyR-HUXYuMRHgAVeN5b7XRPiqgj8n1An2jbWcmTqVyl71BNVh-fb82tkpyQ_Nywh8svzYvV9r6_9jZ9GQuod7eaTVmKVuhDwmobEtHB5HYPxL0AOZFt8QvgXGxPiZcE2CGHa/s4032/IMG_1041.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="3024" data-original-width="4032" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjXkPJgXiz3GBTRHSU_Oh2uYfpFcTnd55FlVgtSKG6LVUxcQQzlBs1fdyR-HUXYuMRHgAVeN5b7XRPiqgj8n1An2jbWcmTqVyl71BNVh-fb82tkpyQ_Nywh8svzYvV9r6_9jZ9GQuod7eaTVmKVuhDwmobEtHB5HYPxL0AOZFt8QvgXGxPiZcE2CGHa/s320/IMG_1041.jpg" width="320" /></a></div><br />I started fooling around with Surround Sound in 2005, and that's about all it was, fooling around. I had purchased 5 speakers from my brother-in-law, and I experimented with placing them on the back counter of my kitchen, where they quickly got in the way and had to be removed.<p></p><p>The current system really got going around 2018 or so, but it's still based on my 2005 Yamana HTR-5790 receiver, which has always worked very well for stereo and everything. It's got very good amplifiers (for 7 channels!) and good performance everywhere. It can decode common surround signals from coax and optical connections. What it doesn't have is HDMI inputs and the ability to receive surround sound via HDMI.</p><p>That hasn't been an impediment, since I can either take the coax low res digital straight from my Oppo BDP-95 for movies, or the seven channel analog output from the Oppo for high resolution audio multichannel discs. Normally I just use the analog for everything, so all my digital decoding is done by the Oppo. The only way this is a "problem" is that I'm not getting high resolution (from high resolution audio discs, not movies) from the few high resolution discs I have in pure digital form for the fancy processing in the receiver. But that doesn't matter much, since the Yamaha receiver doesn't do fancy processing on the seven channel analog inputs. Instead, I do all the basic digital processing required in the Oppo, which has menus for setting the delays and levels for each channel. So the only way that this is a "problem" is that I'm not getting all the fancy digital processing applications like Audessy, that may correct for EQ and phase. And I wouldn't even want Audessy much, except the later versions which allow you to do the fine tuning on your phone. And that set of Home Theater Processors are still very expensive even used, because they have that feature.</p><p>Anyway, I'm getting the seven channels and routing them to amplifiers in the Yamaha and thence to speakers. With levels and delays set in the Oppo. That's pretty much all there is to it.</p><p>Except, what about when there is 5.1 channel content? I could play it back just as that, which is theoretically fine. But because my side speakers are not optimally placed, I often find it better to play 5.1 channels on my 7.1 speakers by duplicating the side content in the back. When I do that, I lower the level by 2dB so it still balances OK. I use a rotary switchbox (made by dB systems no less) to switch from pure discrete 7.1 to 5.1 expanded to 7.1. And then I have a preamp for the sides having a level control with two marked positions. I use the higher position for discrete 7.1 and the lower one for fake 7.1.</p><p>My project to add optimally placed side speakers has gone nowhere in 5 years since I bought the required wall hangable small speakers, currently still in my bedroom to remind me to install them.</p><p>(I have tried many other solutions for the 5.1 to 7.1 conversion. For some time I used a well known box from the 1980's. After using it a year I found it distorted the sound intolerably. Then I tried using various delays and EQ's. Finally I decided that simply duplicating the sides in the backs, and lowering the level by 2dB, worked better than anything else. I also tried a historic Integra processor, but found it did nothing useful, and was basically a pain in the neck because it only produced distortion if input or output levels exceeded 1 volt. The Yamaha is good at least to 3 volts on inputs and outputs.)</p><p>I could solve this 5.1 to 7.1 conversion problem problem "better" with a fancy Home Theater Processor that might cost $5000 (do they still make those???) or more, but it's not been worth it to me when I could have afforded it (and now it's simply unaffordable).</p><p>Anyway, my simple idea is implemented with the units on top of my kitchen rack. The bottom DEQ box is for the rear speakers and it only does level adjustment for the backs, plus level and spectrum displays (so you can see at a glance if the backs are doing anything). The level adjustment feature that this box actually does could just as well be done by a preamp (like the upper two boxes) but I happened to have the DEQ and not another preamp. (It inherently adds about 10mS of additional latency, which I have compensated for in the Oppo adjustments. When used for the fake 7.1 the additional delay is a bonus that makes it sound a bit better--like an even larger room--but isn't that important either way)</p><p>The upper DEQ box is for the subs and currently does nothing more than level and spectrum display. (I thought it was also doing some eq but it appears not. At some point in the last couple years I bypassed it. I'm not sure that was out of design or necessity--such as it might have been adding hum). Currently the upper DEQ box is non-functional, it appears to have the usual power supply issue. But since I'm not doing any processing there anyway, it doesn't affect the sound. But it was very useful to have the spectrum display on the bass because I could measure ground loops exactly and work to eliminate them. The spectrum display on the subwoofer signal enabled me to solve and fix about a dozen ground loops over the years. The complicated kitchen electronics (my central computer, video, and audio for the whole house, plus TV and radio) are prone to those. Now the box that did the helpful bass spectrum displays needs fixing.</p><p>The upper two preamp boxes are for the sub and sides. It allows convenient setting of the levels, which I need to do for the sides when changing from discrete 7.1 to fake 7.1. Also historically I used to mess with the sub level a lot depending on recording. But I've had it dialed in pretty well for everything recently and hardly mess with it at all anymore. Still I like have "controls" that I can just reach up and control as opposed to complex apps which may not do anything until you take another measurement.</p><p>The stereo frequency response (including subs) is quite flat, though I think some of my low frequency optimizations that used to live in the subwoofer DEQ would have made it flatter there. Still it sounds pretty good.</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhmhwDp9Qj3HvFLBv6hItPiUxvrAb8F245RXk9ZI9T0Bq0PN5R3QNVtH8NhWqQVnlWhkSHXoCHEzqktSh6asbAQUocshEX_lcOVTOac7o9gDM4-XcyxOn_oOEUhB2Au6T3stYMv9JxNl1UUmU6VvA92qEMyqhAHsgNuZRVyAsGf6_-d6u9lAhaUpBqs/s2208/IMG_1040.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1242" data-original-width="2208" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhmhwDp9Qj3HvFLBv6hItPiUxvrAb8F245RXk9ZI9T0Bq0PN5R3QNVtH8NhWqQVnlWhkSHXoCHEzqktSh6asbAQUocshEX_lcOVTOac7o9gDM4-XcyxOn_oOEUhB2Au6T3stYMv9JxNl1UUmU6VvA92qEMyqhAHsgNuZRVyAsGf6_-d6u9lAhaUpBqs/s320/IMG_1040.jpg" width="320" /></a></div><br /><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-56823961718795207842023-05-16T20:52:00.008-07:002023-05-24T17:04:04.566-07:00Rethinking the Oppo BDP-205 Filter Choice<p>My last post got 'updated' with a long discussion and testing of the Oppo BDP-205 filter choices. I had been using the Linear Fast filter, thinking that was the best of the best. But it seemed to induce about 5 clipping events beyond +4dB headroom I allowed for inter sample overs (ISOs), which shouldn't be happening at all (or at least Benchmark seemed to say it was only necessary to allow 3.1dB headroom for inter sample overs). This is not necessarily the flaw of the Oppo or any of it's digital filters, but a deviance from my previous expectation and a seeming objective variance between the different filter choices. (See update below. I have subsequently determined that my counts of clipping events was wrong. It looks like all the fast filters generated the same number of clipping events. They just looked different.)</p><p>The other choice that looked best to me at that point (on technical considerations and examining Archimago's measurements) was the Brickwall. I found that only induced 1 clipping event above +4dB, thereby seeming objectively better, plus having the best numerical specifications on noise, distortion, and loss at 20kHz.</p><p>I'm NOT going to choose any of the slow filters even if they eliminated such clipping events (in fact, I'd expect they might) because of their leakage effects.</p><p>But there's one other filter that might be the best of all, and it's actually Oppo's choice for the default, so it's apparently what they thought to be the best. It's the Minimum Phase Fast filter.</p><p>This filter has the curious effect, like all other minimum phase filters, of moving the ultrasonic ringing past any transient, rather than being on both sides (acausal). Intuitively this seems better to most, including me. I was brushing it off last time as "unnatural" but in fact it is the natural thing you could get with real circuits rather than digital simulations, if you could make those circuits well enough (which in practice isn't possible).</p><p>We'll I'm long past thinking about ultrasonic ringing in the first place. I'm more interested in low noise, extended response, and those being equal I'd consider the phase response.</p><p>The minimum phase fast HAS higher noise and distortion than the brickwall...but also it appears to have more ultimate bandwidth (a big plus) at least according to Archimago's spectrum graphs (which didn't look right, because the brickwall had the lowest loss at 20k, but on the graph it was cutting out well before 20k steeply). So given the possibility that min phase fast has the widest bandwidth, and nearly as low distortion, that could make it the best.</p><p>Anyway it occurred to me it might not have these these above 4dB inter sample overs in the first place, and if so it would be an obvious choice.</p><p>But the test shown below shows it has the same single clipping event over +4dB as the brickwall. So it's "equal" in that respect, and better than the linear phase which had 5 such events.</p><p>But...looking at that actual clipping event...it makes far more sense than with any of the other filter, to my analytical eye. In fact it makes so much sense, I'm inclined not to "repair" it at all, as there's no repair that would preserve the underlying high frequency transient it is apparently trying to show (which previously I insisted had to be electronic...and it might be...but that hardly matters here) without smearing it.</p><p>It looks to me best just left very slightly clipped, for it's clear the clipping is right at the upper bound of where it's going to be anyway. There's likely so little difference made in the single clipping event it's not worth lowering the recording level -0.5dB to avoid it (though a test might be warranted). The clipping looks to be hardly making a difference...the peak would only reach a microscopic amount higher anyway, judging from the trailing ringing which is now concentrated on the "after" side making it easier to understand. The other filters give very messy looking results that defy repair altogether, compared to how this looks.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi1XaamA6jnPmGSyqCmjmPiMNy105KFlB3_xf8-9Lulbhthsk9nEsuEOOPik7OofQcl6GBtI8eWl6zNY4nxZF9GLncbCbMLwsjnBAbXz93qZczr4gRiEuNIOW7AVT-nCZAk07D9CFwsnwr3O7iF_gvNMNbYjgkHPXZWukBc3YL6c1hewi2vHXs0uXbh/s1920/Screen%20Shot%202023-05-16%20at%2010.23.53%20PM.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi1XaamA6jnPmGSyqCmjmPiMNy105KFlB3_xf8-9Lulbhthsk9nEsuEOOPik7OofQcl6GBtI8eWl6zNY4nxZF9GLncbCbMLwsjnBAbXz93qZczr4gRiEuNIOW7AVT-nCZAk07D9CFwsnwr3O7iF_gvNMNbYjgkHPXZWukBc3YL6c1hewi2vHXs0uXbh/s320/Screen%20Shot%202023-05-16%20at%2010.23.53%20PM.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Min Fast +4db clip (look for the ringing)</td></tr></tbody></table><p>I'm now leaning towards leaving the Oppo at the factory default Min Fast setting.</p><p><br /></p><p>Update:</p><p>My counts of clipping events may have been wrong. I was just checking visually. This can be misleading and depend on how much time is being displayed and also when exactly it starts. I discovered this by cutting the first section of the Min Fast recording and discovering that there were no clipping events at all, instead of just one. The one had disappeared because of the differing start time. Then when I magnified and scrolled, I saw several events. I probably made that exact same mistake with Brickwall. Lets assume for now they all have the same number of clipping events. (That also explains a recollection of another file I didn't report. It was also Brickwall but had several clipping events, just like linear fast.)</p><p>So this number of clipping events was an entirely bogus analysis. What still looks good is the concentrated and natural way the Min Fast makes each inter sample over clipping event look. They just look right, whereas all the others just looked awful.</p><p>Right now I have no evidence that the factory default filter isn't the best, and one subjective guess that it looks best. (I doubt I could hear the difference...especially in a double blind test.)</p><p>Update May 17</p><p>I listened to Min Fast filter last night and thought it sounded great. Pure, harmonic, and no digital artifacted sound.</p><p>But now it appears that my bit about the look of the ISO clipping events (once again, the fact that there is clipping is not Oppo's fault, it's mine) and it's damned hard to tell even which one looks best. The Linear Phase Fast filter, which I now see is Archimago's preference, does give very short ringing, shorter even than Brickwall, though it has that annoying pre-ringing also. Archimago simply argued on merits, that linear fast is like previous Oppos and most players, and admitted that he's never heard a difference among filters nor was a difference found in earlier testing of similar filters.</p><p><a href="https://www.audioholics.com/blu-ray-and-dvd-player-reviews/oppo-udp-203-udp-205">The Audioholics tester (Gene Dellasala) also tried very hard and could not hear a difference among the filters. His advice was leave it at the factory setting (Min Fast) and worry about more important things.</a></p><p>The difference in noise among the Brickwall, Min Fast, and Lin Fast filters is negligible. The Brickwall noise level is slightly better with -119.0 instead of -118.8. The THD numbers are <i style="font-weight: bold;">identical</i> at a barely measurable 0.0008%<i style="font-weight: bold;">.</i> That indicates none of these three filters have significant leakage. The Brickwall apparently cuts off a microscopic amount faster somewhere above 20kHz resulting in the tiniest bit of extra noise reduction but the benefit is so small, one might as well use the more "natural" looking Min Fast, and maybe that slight added bandwidth is a good thing (especially for someone like me, using supertweeters). The Min Fast may have the widest bandwidth as it does have the lowest loss at 20kHz (-0.19 for the Min Fast vs -0.21 for the Linear Fast vs -0.20 for the Brickwall), and since that's the one and only superior spec for that filter, perhaps it's why Oppo chose it as the default.</p><p>Actually none of this matters to me anyway because I only use the Oppo to play high resolution discs, with sampling above 44.1kHz, or SACD's. Standard discs I simply rip to my computer and send the CD quality bits into my system and the DAC in the Oppo doesn't matter at all. With high sampling rates on high resolution discs and SACD's, any of these filters is way more than good enough. I could even go with the slow filters and not suffer significant alias leakage.</p><p>I think I'm going to follow Gene Dellasala's advice, which I was trending to anyway.</p><p>I also don't have any MQA discs (do they exist?) and when I stream the Oppo DAC isn't involved (and I no longer use a streaming service that supports MQA either). Some apparently like the Apodizing filter with MQA. But for other uses, the Apodizing filter is unappetizing to me in general because it adds a tad of high frequency ripple (almost certainly inaudible, but why have it anyway).</p><p><br /></p><p><br /></p><p><br /></p>Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0tag:blogger.com,1999:blog-8394402209602657003.post-75392901838991966922023-04-22T08:16:00.035-07:002023-05-12T22:48:41.062-07:00The Lavry AD10 Microscope<p>My 12 year old Lavry AD10, which had been left running 24/7 since I bought it (I won't be doing that any more) died when during some power surges when my home foundation was being repaired.</p><p>I sent it to Lavry, who repaired it for $600 by replacing the main board. So basically I have a brand new Lavry AD10 (which appears to still be in production). It might even be better than the original unit (purchased in 2010) ever was, but I never did complete measurements of the Lavry before. I figured I didn't have anything good enough to measure it with.</p><p>The Lavry AD10 has a signal to noise specification that appears to be about 4dB better than the AD converters of the Tascam DA-3000 recorder (117dB vs 113dB). Since the DA-3000 is a newer device, and noise specifications depend heavily on test protocols, I really didn't know which was better. But I figured the Lavry was probably better, and I was right. My measurements yesterday suggest the Lavry is indeed just about 4dB quieter than the DA-3000.</p><p>I've long said that digital converters are about the best thing made in audio because a lot of attention has been focused on them. This may be less true of analog to digital converters nowadays (up until about 10 years ago, typical analog to digital converters were better than the digital to analog converters, somewhat counterintuitively, but now consumer DACs can have better than 130dB S/N which is as good as megabuck ADC's).</p><p>I've often said they were so good, digital converters are often and generally better than preamps. Perhaps even my Emotiva XSP-1, which I envision as about the quietest digitally controlled preamp you can get under $10k (above which perhaps Mark Levinson makes even quieter digitally controlled preamps, though FWIW the Emotiva specs are better than all but the most expensive Levinson "Reference" models and about the same as those Reference models, so we're already getting about as good as it gets with the Emotiva XSP-1, in performance anyway).</p><p>Digital converters are probably better than nearly all non-digitally controlled preamps too, which in some cases (probably not many) might be still quieter. But I like the digital controls for setting levels precisely and also ensuring perfect stereo tracking. I got so frustrated by channel tracking I switched to digital preamps in 2000, my first being a Classe CP-35, not only because of the perfect level tracking but to my ear* it even sounded better than passive and non-digital preamps I had been using before. I've found no need to go back to non-digital preamps since then. (I found the XSP-1 to be far better sounding* than even the Classe).</p><p>Now I have studied this issue in some detail. It appears that there is an unimportant flaw in the noise spectrum of the Emotiva which is made invisible by the noise of the DA-3000. But like a microscope, the Lavry AD10 clearly shows this flaw, an ultra low frequency noise in the left channel only (both of my two Emotiva XSP-1's show this same identical flaw).</p><p>It's worth recounting how I came to use the Emotiva XSP-1 as my living room preamp for all "analog" sources (including the analog outputs of DVD-Audio players which preserve the full 24bit resolution while the digital outputs truncate it to 16 bits). For many years I was using a passive switchbox to switch among such sources, and the level adjustment on the Lavry to set levels for digital encoding for my downstream digital processors. But the Lavry level adjustment lever, which is great for long term settings, it a pain to reset on every disc. A big pain.</p><p>Then I also discovered that the noise level of the big XSP-1, which I had originally purchased for my less high end bedroom system, was even lower than the tiny XPS-1 phono stage I was using in the Living Room, which itself was lower than my dB Systems high gain preamp (which stunned me, because the dB systems preamp and the XPS-1 and the XSP-1 all use the same low noise preamp chip for phono pre-amplification, but somehow it was about 10dB quieter in the XSP-1 as compared with everything else).</p><p>So I needed an XSP-1 in the living room just for the phono preamp alone. And all my bench measurements of the XSP-1 suggested it was about as perfect as I was able to measure, with S/N better than 110dB and distortion below 0.005%.</p><p>So the combination of convenience and performance led me to use the XSP-1 not just for phono preamplification, but also "preamplification" (mostly downward level adjustment) for the special digital sources (DVD-Audio**, SACD, and HDCD) that cannot be output in their full resolution through SPDIF which I need for my digital crossovers and equalizers.</p><p>Now using the "Lavry Microscope" I can see a flaw in the XSP-1 more clearly. This same flaw was invisible amidst the mere 4dB higher noise level of the DA-3000.</p><p>I obsessed over this Emotiva flaw for at least one day. But although I toyed with the idea of using a passive switch rather than a preamp for the digital sources again, I have once again concluded the XSP-1 is "good enough" not to bother with that. Slightly higher noise levels below 5 Hz (and still below -120dB) are just not that important. That's probably what the designers of the XSP-1 thought too.</p><p>[Pictorial section being expanded.]</p><p>First I wanted to re-measure the setup I've been using, Oppo BDP-95 into Emotiva XSP-1 into Tascam DA-3000. I also measured the Lavry with open inputs (not shorted, an oversight). By itself, with open inputs, the Tascam had about 1 dB less noise than with the full chain. That suggests that the Tascam was generating more than half of the noise, which would be about what I'd expect. </p><p>The picture below shows about 5 minutes of recorded noise amplified digitally by Audacity by 94dB. So we're taking a close up look at the noise, about as close as we can get as there is only about 0.3dB of headroom on the right side, which is the full chain of equipment, whereas just the Tascam by itself is on the right.</p><p><br /></p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhsZF0W44XbJcNjRYw7eQPm6tw35ve4kRBAr8L7MLw5hqg-xJaI7hpiwtuA6aQKq4fJUsVp2F9CXygOD-x1lEzrVum7McTUelKBmv7YJy8Ovh7XObV1lAxetf0e627dznrQdMg9dVfK-7Xc-AVfo8hzGRtn_paQQPaS3bCBf2QF23-J7tY07_-VKZm0/s1920/Oppo:Emotiva:Tascam%20vs%20Tascam.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhsZF0W44XbJcNjRYw7eQPm6tw35ve4kRBAr8L7MLw5hqg-xJaI7hpiwtuA6aQKq4fJUsVp2F9CXygOD-x1lEzrVum7McTUelKBmv7YJy8Ovh7XObV1lAxetf0e627dznrQdMg9dVfK-7Xc-AVfo8hzGRtn_paQQPaS3bCBf2QF23-J7tY07_-VKZm0/w400-h225/Oppo:Emotiva:Tascam%20vs%20Tascam.png" width="400" /></a></div><br /><p>Things didn't look so good with the Lavry inserted in front of the Tascam to do the A-D conversion (using the Tascam wordclock signal for synchronizing):</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg5oq47zxjOsctZkHjhX4Pp7B4GUj8MrmgpZsWIhUwNnlHmVEixMelByXE3nJQitXUTibVTj0Hu9JSfMma-PTbvHK3Ir9xJxi819xRJ_51Yxhb3LvVrws4FYVg0QS8JWZeR8PuNZFLosgwwRDuW-OGvdHDrzloQx_e60Ub2eXEN9QMPPNwAwjyF4iH0/s1920/Oppo:Emotiva:Lavry:Tascam.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg5oq47zxjOsctZkHjhX4Pp7B4GUj8MrmgpZsWIhUwNnlHmVEixMelByXE3nJQitXUTibVTj0Hu9JSfMma-PTbvHK3Ir9xJxi819xRJ_51Yxhb3LvVrws4FYVg0QS8JWZeR8PuNZFLosgwwRDuW-OGvdHDrzloQx_e60Ub2eXEN9QMPPNwAwjyF4iH0/s320/Oppo:Emotiva:Lavry:Tascam.png" width="320" /></a></div><p>The (top) left channel noise looks "noisier" somehow. Notice that the right channel noise is significantly less than when recorded directly by the Tascam in the previous picture. Almost 4dB lower in fact, just as the specs of the two AD converters suggest. And if you carefully compare the right channel to either of the Tascam direct recordings, it's actually lower, but not as much as the right channel, because of some extra noise that's being added somehow to it.</p><p>My first concern, in fact the main reason I was doing these tests was to be sure that the Lavry, just back from an expensive full board replacement, was now working properly. And this first measurement didn't look good. And possibly because I go about things in a more round about way than necessary, I didn't fully prove that the Lavry was fine and good until about ten measurements later. Well I was also concerned about my Emotiva and Oppo which might be just as expensive to repair.</p><p>And it turns out that the problem itself is no big deal. But I've decided to tell this story mostly as I experienced it, so it becomes clear that way.</p><p>It seemed to me that the first and easiest thing to do would be to reverse the channels. The right side of the picture shows the normal connections and the left side shows the reversed connections.</p><p><br /></p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhnOKLVR6V8--3SwHrhLaLMrPdx2cbAvBoYNEZ89YVYqZ0szsncqpBty8OVt9Kr5EeH-nqdSBWrdlR6Un_H2_-dssv-tQYVvClsTcI_4ZkwP1ARiEh8u6Gd2OAdVQEO0fsF1XsfbuqgnUnCTnr96nx1Tx7XWrQfsPNOsBLlcCfmw29o55ZHrMs_UfPe/s1920/ChannelReverseAtLavry.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhnOKLVR6V8--3SwHrhLaLMrPdx2cbAvBoYNEZ89YVYqZ0szsncqpBty8OVt9Kr5EeH-nqdSBWrdlR6Un_H2_-dssv-tQYVvClsTcI_4ZkwP1ARiEh8u6Gd2OAdVQEO0fsF1XsfbuqgnUnCTnr96nx1Tx7XWrQfsPNOsBLlcCfmw29o55ZHrMs_UfPe/s320/ChannelReverseAtLavry.png" width="320" /></a></div><p>The extra noise in the left channel moves to the right channel after I reverse the cables. So therefore, whatever it is that is causing the extra noise in the left channel with normal connections must precede those connections. It possibly comes from the cables themselves, so I then tried putting the Lavry connections back to normal and reversing the output connections at the Emotiva:</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg64Bdd0-H6l4Rk91UoZRiYv5BmzIbtp2hkHsNQKOPLqlrkA1wGUG66RzoBZbenRGEALAS-8Qah6EFYUPeQt0E_oQ1ScWXmgG5sO5u2bCPnQrlTadRyBUrSbmzyQfZCm_fnb8cquVUyV3kUnOUToy_csuSGIMEz_j1Rj-cAM0ZM2xJfrZIVC48nTWQ6/s1920/ReversingDifferentEnds.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg64Bdd0-H6l4Rk91UoZRiYv5BmzIbtp2hkHsNQKOPLqlrkA1wGUG66RzoBZbenRGEALAS-8Qah6EFYUPeQt0E_oQ1ScWXmgG5sO5u2bCPnQrlTadRyBUrSbmzyQfZCm_fnb8cquVUyV3kUnOUToy_csuSGIMEz_j1Rj-cAM0ZM2xJfrZIVC48nTWQ6/s320/ReversingDifferentEnds.png" width="320" /></a></div><p>After all the messing around with cables I had done just to get the audio XLR cables moved from the Tascam to the Lavry I worried that I might have damaged them. But it was clear that it made no difference whether the cables were left-to-right reversed at the output or the input, the result was exactly the same. So the XLR cables between Emotiva and Lavry were exonerated from causing the added noise. </p><p>At this point I decided to "mute" the Emotiva preamp. I'm not sure whether I dialed the volume all the way down, or turned it off. (My recollection is that I was going to turn the volume down, but I made the recording without even turning the Emotiva on. So I called that "muting the preamp" in my too-brief notes.) The strange extra noise in the left channel went away completely. From this point on, I was convinced that the problem was not in the Lavry, however I didn't fully verify that until a later test in which I put shorting plugs in the Lavry inputs.</p><p><br /></p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgkOtDOZA-YpYOm64YRcp2BIgPcQSlJa_mG4uxVMRQ4dfWDOUb1ZvCivwLnTEHX8B_uAjLW49xhgXL1El9BnROOzF40LAKgaTbYuoJrhoi6WFYzGUV-IuTEkwheil7oq1Jf9dgFpprjOWzOlFCKToQ4ar1YHZTPjsuFcQBUFALqzfvnBj1AgRExfSjJ/s1920/EmotivaMutedOff.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgkOtDOZA-YpYOm64YRcp2BIgPcQSlJa_mG4uxVMRQ4dfWDOUb1ZvCivwLnTEHX8B_uAjLW49xhgXL1El9BnROOzF40LAKgaTbYuoJrhoi6WFYzGUV-IuTEkwheil7oq1Jf9dgFpprjOWzOlFCKToQ4ar1YHZTPjsuFcQBUFALqzfvnBj1AgRExfSjJ/s320/EmotivaMutedOff.png" width="320" /></a></div><br /><p>It was at this time that it occurred to me that prior to having the Lavry repaired, I had taken great pains to connect the AC power to the same power strip as the Emotiva and all the front end components. It looked impossible to make this change without disconnecting and moving heavy equipment like the almost 50 pound Denon DVD-9000 from the rack in order to get at the power strip behind it. But somehow I managed to get it done anyway. I even used the same 3 foot SJT power cord I had been using before.</p><p>The results seemed like a significant improvement when I was first examining them, but on checking them now, I'd suggest they were not any change at all. However in either case the extra noise in the Left channel did not go away. That part had not been fixed.</p><p>Then I removed extra cable after extra cable from the preamp in a whole series of tests I won't bother you with here because they were all the same.</p><p>Finally I went all the way with this sort of test by removing every single cable from the Emotiva except AC power and the output cables connecting to the Lavry. I selected a balanced input (2) which was shorted with XLR shorting plugs. The extra noise did not go away (note that the channels are still reversed because the output cables were reversed at the Emotiva). The Emotiva left channel (bottom) noise looks different from previous images because now we are looking at just one minute instead of 5 minutes. After spending hours doing these tests, I decided I could just as well get by with 1 minute recordings. Now it's becoming clear that the extra noise is a very low frequency baseline shifting around -98dB in level (these pictures are amplified 94dB by Audacity, so a full scale noise would be -94dB down):</p><div class="separator" style="clear: both; text-align: center;"><span style="text-align: left;"><br /></span></div><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhsqTIAcOUyblOet2ZvbBLppiCTmNgQxwglkQzxW4Wm6bEz36t2w9BcZoM4sX2RbpDNeb-zg9YHttR5Ml4CTCIiD50skn4-rfPgmP7XGNdi_h6PipMzIc35MmokNdbsLFqluzPRAV9WjFj6mI-xlOJG4H10G2FeQYZvow9ICndy800l_Gpo4YLxlkpe/s1920/Emotiva%201%20Fully%20Disconnected.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhsqTIAcOUyblOet2ZvbBLppiCTmNgQxwglkQzxW4Wm6bEz36t2w9BcZoM4sX2RbpDNeb-zg9YHttR5Ml4CTCIiD50skn4-rfPgmP7XGNdi_h6PipMzIc35MmokNdbsLFqluzPRAV9WjFj6mI-xlOJG4H10G2FeQYZvow9ICndy800l_Gpo4YLxlkpe/s320/Emotiva%201%20Fully%20Disconnected.png" width="320" /></a></div><br /><div><p>Now that I had the XLR shorting plugs out, I plugged them straight into the Lavry. The result was the lowest noise of all with no extra noise in the left channel whatsoever. To compute the peak unweighted noise level, I bring up the Amplify dialog one more time, and it shows me how much more amplification is possible. In this case it is showing 5.78dB more amplification is possible Since the starting amplification here is 94dB, the signal to peak noise level is 94+5.78 = 99.78dB. Applying RMS adjustment and A weighting would probably improve this to almost 120dB or maybe better.</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh7LDjdIKDnupL0Yvi4ckVzkph5QjQeA03IirNZET6eYhRSwN4Y-zdgC3Aeu6kZ06o6QBOIn3PYLOjSREX9BG06Dh_QtEnXWxKrd8w7TIPJtB7sXbWnvQmLVdHvgBJGB5rrjv-c-mI7gC6WlTf_2ZNECI0mabRdnal_c3mkLKu3PAsnz_t3mrkVVFNK/s1920/Lavry%20Input%20Shorted.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh7LDjdIKDnupL0Yvi4ckVzkph5QjQeA03IirNZET6eYhRSwN4Y-zdgC3Aeu6kZ06o6QBOIn3PYLOjSREX9BG06Dh_QtEnXWxKrd8w7TIPJtB7sXbWnvQmLVdHvgBJGB5rrjv-c-mI7gC6WlTf_2ZNECI0mabRdnal_c3mkLKu3PAsnz_t3mrkVVFNK/s320/Lavry%20Input%20Shorted.png" width="320" /></a></div><br /><p>I concluded there was something "wrong" with the Emotiva XSP-1 preamp that was causing a small but measurable extra noise in the left channel.</p><p>So the next day, I moved the pile of mostly spare equipment away from the side of the rack so I could swap the living room and bedroom XSP-1's. The living room XSP-1 measured above was purchased in 2018. The bedroom XSP-1 was purchased in 2014 (it's also the second generation btw) but was repaired (basically refurbed) in 2019 by the factory. As part of the repair they do a full AudioPrecision test which verifies it meets all specifications. Since the repair, it has not been used very much (and I was careful to keep it turned off when not in use, something I should have been doing from the beginning because even if nothing else the display gets rather dim in about 5 years of on time).</p><p>As it turned out, my other XSP-1 was almost identical to the one measured above. The added noise in the left channel looked almost (but not entirely) identical. The noise in both both channels was just over a half dB lower, so I decided to keep this other XSP-1 in the living room from now on (and it won't get unwanted wear now that my home control system turns it the living room preamp on and off using the trigger signal).</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgsdeBO5vE311xKeyOmy1Dj-_CEh8LCOR8KB0EdWhp-105Rbw0ZJjY1slvp4gvFCFu_u4_UMYvcCtqJA4ozIY5srYcr3iGRxA7vGOeGo6cGrX0sQ1JZDr8B7FEXcpnIrAwOVKVtu65ShM1gfatvJJMeNoqtUHVHl8b3XtKmLuqjK1_WQKeCztsj8Prq/s1920/Second%20Emotiva%20Disconnected.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgsdeBO5vE311xKeyOmy1Dj-_CEh8LCOR8KB0EdWhp-105Rbw0ZJjY1slvp4gvFCFu_u4_UMYvcCtqJA4ozIY5srYcr3iGRxA7vGOeGo6cGrX0sQ1JZDr8B7FEXcpnIrAwOVKVtu65ShM1gfatvJJMeNoqtUHVHl8b3XtKmLuqjK1_WQKeCztsj8Prq/s320/Second%20Emotiva%20Disconnected.png" width="320" /></a></div><br /><p>It was only now I started to probe the noise itself. You can see recurrent peaks in the range of 2-3 seconds (see the selected range in the above picture). That means the noise is in the range of 0.5 Hz to 0.33 Hz. We're talking about very low frequencies here. And we're also talking about very low levels too, certainly 97 dB or so below peak level. That's good, but not as good as the midrange noise floor which seems to reach down to -150dB or so (better than I should be able to measure, though it seems I can). Here are the spectrum plots of the two channels, first the better one. Remember to subtract 94dB from the levels shown in the spectrum.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjRzu7N8koM0hy23-PS4pkB90RFkwlHgbnOdLvPgIm4oZDzGtavQ0A8oWze-WzhM4lh214Gyo8dVznfXX7c6IOl6lvJZJ0rorIL6KqR5IRig4IHmGjdI7lWKuBVGkIREu1kQ4V56jqBBgcY607LshydZKpwVTBs8yMhuiGI_9gfywVA6GzpcJAb-BOb/s1920/Second%20Emotiva%20Better%20Channel.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjRzu7N8koM0hy23-PS4pkB90RFkwlHgbnOdLvPgIm4oZDzGtavQ0A8oWze-WzhM4lh214Gyo8dVznfXX7c6IOl6lvJZJ0rorIL6KqR5IRig4IHmGjdI7lWKuBVGkIREu1kQ4V56jqBBgcY607LshydZKpwVTBs8yMhuiGI_9gfywVA6GzpcJAb-BOb/s320/Second%20Emotiva%20Better%20Channel.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Emotiva Right (better) channel<br /><br /></td></tr></tbody></table><br /><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi9d-2Pji6zfBLr_cvjigD3OmtWuiJXEynzfZqZdv3tNB0HeRF0ydY6BgISs6np295Ide-CupsdKyse8NPo3zd9jbb8duq8UIzytJycCFs45J1O7nrsiyQ4YnzGxehkkD2lq0U6b6votsEexn75iyJVvHB-4d-hoWkU8V3ZF3rSTlTNy5bdMQFVjrdb/s1920/Second%20Emotiva%20Worse%20Channel.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi9d-2Pji6zfBLr_cvjigD3OmtWuiJXEynzfZqZdv3tNB0HeRF0ydY6BgISs6np295Ide-CupsdKyse8NPo3zd9jbb8duq8UIzytJycCFs45J1O7nrsiyQ4YnzGxehkkD2lq0U6b6votsEexn75iyJVvHB-4d-hoWkU8V3ZF3rSTlTNy5bdMQFVjrdb/s320/Second%20Emotiva%20Worse%20Channel.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Emotiva Left (worse) channel</td></tr></tbody></table><br /><p>The main difference here is that the noise seems to rise below 4 Hz faster in the left channel. In both cases the midrange and high frequency noise level is extremely low. The signal to noise in the midrange is around 154dB (94+60).</p><p>This low level ultra low noise (ULF) noise in one channel would not even appear in an "A" weighted noise until you were many decimal places out. Very low frequencies like this simply aren't audible and usually aren't measured either. Listening to the noise (amplified by 94dB by Audacity) the left channel with the ULF noise if anything sounds softer and more pleasant than the right.*</p><p>Many preamplifiers are going to have input capacitors which completely filter such low frequencies out. And sooner or later it is almost certain some component will filter it out. And it's very low in level (-97dB or more) to start with.</p><p>A nearly identical very low frequency noise in just the left channel is a flaw in both my two units, including the second which was sent back to the factory for repair 2 years ago and hasn't been used much since. It looks to me like a design flaw that Emotiva isn't concerned about, and truly it isn't of much importance either.</p><p>In fact, it's rather surprising that the Lavry AD 10 has frequency response which extends down to 0.33 Hz, but it appears like it must. Whereas the Tascam DA-3000 does have about 4dB more noise, as the specs suggest, which partly covers up the Emotiva ULF noise, and partly it may have low frequency filtering which also hides it.</p><p>So at the end of the day, the difference seems to be that the Lavry captures the ULF noise of the Emotiva both because the Lavry is so quiet, and also because it has low frequency response that seems to extend to something like DC, whereas the Tascam does not.</p><p>But just because you can see something with some kind of measurement, doesn't mean it's important. I can't imagine a single good reason why such a ULF noise would be important not to have. At this point, I don't even have any data about whether other preamps have similar issues. With most tube preamps, noise would overwhelm such small effects, and in my experience with tube preamps they flopped around their DC levels not just in microvolts but in millivolts if not volts. So there you could say the XSP-1 has a bit of that kind of "tube character" in very attenuated form.</p><p>Though my theory is that it's the output servo loop of the left channel being closer to the power supply or computer or something like that. An issue that could be resolved with yet another board layout revision. But the XSP-1 is beyond revisions now, it's been discontinued (which I'm not happy about either, I think it is a very fine preamp and I don't believe Emotiva or anyone else has a close enough replacement for me now, though fortunately I don't need a replacement now, my two XSP-1's are working fine, good enough in my opinion--did I mention the midrange noise floor looks to be in the vicinity of -150dB, that's the kind of thing that actually counts).</p><p>The next day I decided to test an alternative theory, that the problem was being caused by a ground loop in the (coaxial) clock signal from the Tascam to the Lavry, possibly causing jitter. (They are plugged into different AC power strips and that is almost unavoidable.) I found that using the internal clock on the Lavry and the Sample Rate Converter on the DA-3000 made no difference. (My notes are unclear if I also disconnected the clock, which wasn't entirely necessary for these tests but would rule out a ground loop on the input circuit.) Also I shorted the left input on the Lavry but using the usual clock cable, which eliminated the extra noise in the left channel. And remember the channel reversing experiments and shorting experiments above, which were also all done with the usual clock cable. It seems to be disproved that the clock signal is in any way involved in this extra ULF noise.</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEglQfqufKEdiQU5F9i82Xzh5RyXgWwiOV65UUojGpNbwuNAuXzY-WTxXVeyxwzFdc2ULxdIDsACxjCMgRNFWHakKl6S3gvt3N2qr41NlFUkLjW_MwxKgeQ-wzZw1cz7sYyu7zjB1mPmoT-Vlj0Gtxra2TXVJTRZILtHLsrC380s1Gfdete8U_Cp44dr/s1920/Left%20Input%20Shorted.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEglQfqufKEdiQU5F9i82Xzh5RyXgWwiOV65UUojGpNbwuNAuXzY-WTxXVeyxwzFdc2ULxdIDsACxjCMgRNFWHakKl6S3gvt3N2qr41NlFUkLjW_MwxKgeQ-wzZw1cz7sYyu7zjB1mPmoT-Vlj0Gtxra2TXVJTRZILtHLsrC380s1Gfdete8U_Cp44dr/s320/Left%20Input%20Shorted.png" width="320" /></a></div><div><br /></div><div>Here the left channel is on top, but the Lavry left input is shorted. The bottom right channel shows a little ULF noise, but unchanged from the right channel in previous measurements and far less than the left channel.</div><div><br /></div><div>Unhappy that I had failed to note if I had removed the cable or not, I decided to do another followup test this time making sure I removed the cable. By this time (and also in the previous set of tests) I had already figured out a small optimization. I reduced attenuation on the Lavry by 3dB so that peak level is now -8 (around 5.5v peak) instead of -11 (3.8v peak). Then I increased the gain on the Emotiva to +4. The Emotiva can just as easily handle that balanced output voltage. This optimization could in theory increase the S/N by 3dB. (I set the Emotiva level to be just before the level it causes clipping on the J-Dunn test. Since I reduced the Lavry sensitivity by 3dB I might have just raised the Emotiva to +3, but it now seems like +4 is the correct level to reach closest to 0dB on peak signals.)</div><div><br /></div><div>The following tests showed that removing the clock cable (which requires setting the Lavry to internal oscillator and enabling the SRC on the Tascam) makes no difference at all to the ULF noise in the Left channel. </div><div><br /></div><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhVsoJ8El7J0L6zcEzBaxzoiQYuznUB5-dwKp-c-ZfcMY-VogQIChCf8NpBmuGDj1BPuZXsa-h9TMM4P4nGK3b3YfksFH1tdBG4MEmhBsYUk_BJ2QbPzO2dr2qjX-Y0rWRSboHhZh1SML8cYP2-kyWDYentyNmZy1krR8r6p8HGSAGa50YHKy6BQ6Bp/s1920/Normal%20clock,%204dB%20higher.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhVsoJ8El7J0L6zcEzBaxzoiQYuznUB5-dwKp-c-ZfcMY-VogQIChCf8NpBmuGDj1BPuZXsa-h9TMM4P4nGK3b3YfksFH1tdBG4MEmhBsYUk_BJ2QbPzO2dr2qjX-Y0rWRSboHhZh1SML8cYP2-kyWDYentyNmZy1krR8r6p8HGSAGa50YHKy6BQ6Bp/s320/Normal%20clock,%204dB%20higher.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Clock Cable being used, +4dB higher level</td></tr></tbody></table><br /><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgMahMAitVaD_fyjJB-IhLP6wu5x4L1QBJGbwYCnnai7Aj6fpYDCSUDi-7mOPLliNvYV8r6JfMDkYKxzrESC_dgBmWPsl2zTJtfROYydeDkdiK0L3VIpB3KE3J1KxM-jOiV94Y4EUyNaUoBvluvpNsz3VD_BOZ6-p3v0vlObzxWPYvHeqCia2gyfGdU/s1920/Clock%20Cable%20Disconnected,%204dB%20higher.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgMahMAitVaD_fyjJB-IhLP6wu5x4L1QBJGbwYCnnai7Aj6fpYDCSUDi-7mOPLliNvYV8r6JfMDkYKxzrESC_dgBmWPsl2zTJtfROYydeDkdiK0L3VIpB3KE3J1KxM-jOiV94Y4EUyNaUoBvluvpNsz3VD_BOZ6-p3v0vlObzxWPYvHeqCia2gyfGdU/s320/Clock%20Cable%20Disconnected,%204dB%20higher.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Clock Cable disconnected, 4dB higher level</td></tr></tbody></table><br /><p>It is looking like the level readjustments may also have reduced the ULF noise from the Emotiva. The new noise levels (with clock connected as usual) are amazingly good, peak noise -99.2dB in right channel and -98dB in left channel. A weighted RMS S/N would be in the vicinity of 120dB. This is the full chain including Oppo, Emotiva, and Lavry.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiyUWystnU7xbUNa1nj9-5kK3dz1FATTOv9hqWmWE1jVy--Kyr_hsPXYsNyUZkU46tkk_unT1ReEp6drjRJBvFX50-zngjWuN9xpiCQv-MYh6ev8nXv1A1-nXaG5ZoLsp4JYlEE9quoI71WRx0jQVNOAimFjr5Q3p9lwkY8HgYR6X22TKNQG1ze_EXJ/s1920/Right%20Channel%20Noise%20New%20gain.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiyUWystnU7xbUNa1nj9-5kK3dz1FATTOv9hqWmWE1jVy--Kyr_hsPXYsNyUZkU46tkk_unT1ReEp6drjRJBvFX50-zngjWuN9xpiCQv-MYh6ev8nXv1A1-nXaG5ZoLsp4JYlEE9quoI71WRx0jQVNOAimFjr5Q3p9lwkY8HgYR6X22TKNQG1ze_EXJ/s320/Right%20Channel%20Noise%20New%20gain.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Right Channel Noise</td></tr></tbody></table><br /><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhTBIVHyVn4EfBgMB8X9r5PZrFvNtQjH3QcRypxAncHsV6EWjnhM3p7JIGG0bkd467WE4xxPcnS6gxJ8qQaaKDMCkK_kc2LGjXq0IUhJ9WaUGC9OyJkkepLt5aAdolE90hNw550HhlNT7MdlAtSbdDb8sfwQjPR2QozQM7H_EheT_W4xvq_HJGvV9fq/s1920/Left%20Channel%20Noise%20New%20Gain.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhTBIVHyVn4EfBgMB8X9r5PZrFvNtQjH3QcRypxAncHsV6EWjnhM3p7JIGG0bkd467WE4xxPcnS6gxJ8qQaaKDMCkK_kc2LGjXq0IUhJ9WaUGC9OyJkkepLt5aAdolE90hNw550HhlNT7MdlAtSbdDb8sfwQjPR2QozQM7H_EheT_W4xvq_HJGvV9fq/s320/Left%20Channel%20Noise%20New%20Gain.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Left Channel Noise<br /><br /></td></tr></tbody></table><p>Now I wanted to see how much the gain changes were making this better, so I went back to the old gain setting on the Lavry (-11dB reference level, ie +11dB gain). In fact, the gain changes (previous set of measurements) were making a pretty big difference. The new noise level (with Emotiva at +1 to optimize J-Dunn headroom as was done in last two measurements) is -95.5. So there has in fact been about a 2.5dB improvement in lowering the ULF noise from the Emotiva by lowering the Lavry gain 3dB. The other channel noise level has not changed as much, presumably because it's mostly being caused by the Lavry at this point.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjeJ4TgAhsTymRImVgkKAlTZkyNtY2n8G4uw_rQgkZgRjz0WNCD44cQUqV-py2JfvwY-yeDIUXwQlc-yqJSa9cWWoQOCeBqblToOKdDrm1tRq_fICnB20XfAqC3qrMVW1th9UDeGHcjubAfxRtBNDDoQOfohE9WsNyTK6oaMjqtQaSIChaoJT56tTak/s1920/Emotiva%20+4%20Lavry%20-11.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjeJ4TgAhsTymRImVgkKAlTZkyNtY2n8G4uw_rQgkZgRjz0WNCD44cQUqV-py2JfvwY-yeDIUXwQlc-yqJSa9cWWoQOCeBqblToOKdDrm1tRq_fICnB20XfAqC3qrMVW1th9UDeGHcjubAfxRtBNDDoQOfohE9WsNyTK6oaMjqtQaSIChaoJT56tTak/s320/Emotiva%20+4%20Lavry%20-11.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Emotiva at +1dB, Lavry at -11</td></tr></tbody></table><p><br /></p><p>Most of the prior measurements were done with Emotiva gain at +0. This was artificially making the S/N look "better" than it actually was. Here is a replication of what I was doing before. It "looks" like the peak noise level is -95.9 but that is misleading because full scale cannot be reached.</p><p><br /></p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgSsUmMblH_8zrBhkn7gOOgFWNML9Dzi7BLj29hY4_QgHvJs37yfoTJMSeL8r9xzoRyudM5OhqnG_JO5P5yKaX_ktiQs9JlkqXkhKBbPaWEK31id8-O2VmEVcDFya5N_9q64tLyPTOEj_VHRtdVnw_JLF39fUY8CDNAe4-BVkgWkOTVZTxrZ1J2Zsux/s1920/Emotiva%20+3%20Lavry%20-11.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgSsUmMblH_8zrBhkn7gOOgFWNML9Dzi7BLj29hY4_QgHvJs37yfoTJMSeL8r9xzoRyudM5OhqnG_JO5P5yKaX_ktiQs9JlkqXkhKBbPaWEK31id8-O2VmEVcDFya5N_9q64tLyPTOEj_VHRtdVnw_JLF39fUY8CDNAe4-BVkgWkOTVZTxrZ1J2Zsux/s320/Emotiva%20+3%20Lavry%20-11.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Emotiva at +0, Lavry at -11</td></tr></tbody></table><p>Because I'm not boosting the level, I wanted to see how much effect this has on distortion caused by the Emotiva which may be putting out more than 2V balanced into the Lavry. I used a CD on which I have recorded digitally generated (by Audacity) 880 Hz at maximum level (0dB) but no clipping. I changed the Lavry gain to 0 for these tests. I tested this "maximum output" CD at Emotiva gain levels of +0, +4, +9. and +10. At Emotiva gain level +11 this signal was clipping the Lavry (even at 0dB gain on the Lavry so I could not even make the measurement).</p><p>At +10 (which is 6dB higher than my new standard level) there were no visible peaks above the -90dB bottom of the Audacity spectrum (but is this misleading?) and the bin value was -101dB but the peak value was -28.8dB. That would mean about 4% distortion. My guess is that the "peak" value is more representative of the harmonic distortion peak but it might be exaggerated. All I can say for sure is that somewhere above +8 we turn a corner and distortion starts rising. </p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgyZA2hoHqX8sC0mDE2rb-LQR5XMwy9eqt9yoaXv6Wlb4zc6jspFTg7-SSUJeDHv0-9U9dNllP0Bk8RahhQvEmGa07Bbpa0m34GJ3nLpAYHX0HSUGWgHs97gtNfv5rK6wwnHFt7uPpm_LAcQn0eB88Bv7cMhS3fOUq843v2pwCQLop2UdKgL-HeEqLD/s1920/Emotiva%20+10%20max%20880.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgyZA2hoHqX8sC0mDE2rb-LQR5XMwy9eqt9yoaXv6Wlb4zc6jspFTg7-SSUJeDHv0-9U9dNllP0Bk8RahhQvEmGa07Bbpa0m34GJ3nLpAYHX0HSUGWgHs97gtNfv5rK6wwnHFt7uPpm_LAcQn0eB88Bv7cMhS3fOUq843v2pwCQLop2UdKgL-HeEqLD/s320/Emotiva%20+10%20max%20880.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Emotiva at +10 with max 880</td></tr></tbody></table><p>The very best harmonic distortion measurements were with the Emotiva at +8. At that point, the peak level of the second harmonic is is -110.6dB, corresponding to THD of 0.0003% or better.</p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjmjhRYbt2TdWQuncGBV3LB2nrGY3S0QT3J6zZSDTn1htc6g7PCCOfoQBSqdHXSKcOOXRp_gA1OjuuvpSX9j7OGI3TDgRY-8WV_ioDsoZ14LBljKBGMPtyvRBUsHWkQok6PLe7_2s-42lp9fBDQP3J0jnlUgsjq_Qw85Uj7jNuHR7aGF-C-U1jJVwZZ/s1920/Emotiva%20+%208%20max%20880.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjmjhRYbt2TdWQuncGBV3LB2nrGY3S0QT3J6zZSDTn1htc6g7PCCOfoQBSqdHXSKcOOXRp_gA1OjuuvpSX9j7OGI3TDgRY-8WV_ioDsoZ14LBljKBGMPtyvRBUsHWkQok6PLe7_2s-42lp9fBDQP3J0jnlUgsjq_Qw85Uj7jNuHR7aGF-C-U1jJVwZZ/s320/Emotiva%20+%208%20max%20880.png" width="320" /></a></div><p>Backing the Emotiva down to +4dB gain, the distortion rises slightly to -107.7, or about 0.0006%.</p><p>Someone lacking knowledge would just set the level to +8 (and the corresponding -4dB reference level on the Lavry--as it's clear the numbers have to add up to 12 to just avoid clipping the Lavry above 0dB).</p><p>However, I've long known about inter-sample-overs. In between samples, the signal may rise above the 0dB level when rendered with oversampling--which must fill in the points between the points. My sampler is likely to read these inter-sample-overs at least some of the time (or maybe nearly all of the time if I'm sampling at a higher rate than the original, which I need to do for decoding HDCD's for example).</p><p>I've long assumed that inter-sample-overs could reach as high as 6dB, though I've only ever actually measured about 4dB. If in fact inter-sample-overs could get that high, I'd have to set the Emotiva gain no larger than +2 and the Lavry at -10 to avoid having them clip the Emotiva. (But read on...)</p><p>Now, if you're not understanding inter-sample-overs and why I seem to be setting the MOL from the Emotiva at 6dB lower than the optimal distortion level (+8) for no good reason, read <a href="https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings#:~:text=Intersample%20Overs%20are%20a%20Common%20Occurrence%20in%20CD%20Recordings,-We%20have%20frequently&text=This%20intersample%20overs%20contain%20dynamics,by%20most%20D%2FA%20converters.">Benchmark's description of inter-sample-overs</a>.</p><p>Benchmark sets their headroom for inter-sample-overs at 3.5dB. Perhaps I'd measured 4dB because of rounding in the low resolution readouts on the I've been using. Benchmark says the maximum theoretical seems inter-sample-over is 3.01dB around 11kHz.</p><p>If only 3.5dB headroom for inter-sample-overs is needed, since I can't adjust in fractional units, I'd have to allow 4dB headroom below the optimal distortion at +8, thereby putting me right back to the adjustments I was assuming in my first "optimization," <b>+4 gain on the Emotiva and -8 "reference level" (ie 8dB gain) on the Lavry</b>.</p><p>So that now does look like the optimal adjustment, taking both headroom and noise considerations into account. No only will no real signal cause rising distortion from the Emotiva, but no inter-sample-over will either (and many converters don't even handle those well...perhaps the true deficit in CD reproduction from the beginning). In fact with the +8 setting I've allowed at least 0.9dB more headroom than necessary for inter-sample-overs, and possibly more (as I never bothered to measure the +9dB setting on the Emotiva).</p><p>(Note that the Oppo BDP-205, which was playing but paused during the noise measurements, and played the 880 Hz maximum level signal, has 2V XLR output. Some say this is "all that's needed for any amplifier" but in fact my Krell FPB-300 required 2.8V and much higher to reach the true peak levels. Can't always rely on inter-sample-overs to bring you there either--those depend on high frequencies usually. I would have expected standard XLR level to be twice the RCA level, so 4V balanced. Anyway, with the Emotiva at 0dB it will also be putting out the 0dB level of 2V which is pitiful I think for a balanced output. By boosting that 4dB, it's being boosted to 3.2V, and allowing another 3.5dB of headroom would put us around 4.7V. Surely the Emotiva balanced outputs are still in their low distortion range at that point! In fact, in the <a href="https://hometheaterhifi.com/reviews/amplifier/preamplifier/emotiva-xsp-1-balanced-stereo-preamplifier/">Secrets of Home Theatre and High Fidelity measured the lowest distortion at 5V</a> ! (It bothers me they didn't probe the question "how high does it go", but as it turns out, 5V is all I need. The 8dB gain setting on the Lavry officially corresponds to a voltage of 16dBu or 4.89V.)</p><p>Now I sort of remember that I'd come up with this setting (at least the -8 reference level) many years ago, driven by the Two Against Nature or Everything Must Go, both Steely Dan albums had incredible inter-sample-overs. I figured out this same setting (and I think it was this one) empirically driven by the need to play the a Steely Dan DVD-Audio, because when I first played it, it clipped my sampler like hell. This was how I personally discovered inter-sample-overs before I even read about them. Benchmark refers to a particular Steely Dan track on Two Against Nature in their discussion and then analyzes a few others on that album and a few other albums notorious for the highest inter-sample-overs.</p><p>Then over time, I couldn't remember my results, and I drifted back to the obvious (-11).</p><p>Update May 12 2023</p><p>Oops, I wasn't thinking clearly. While the "-8" reference level (ie 8dB gain) on the Lavry is still correct (that means the Lavry clips before the Emotive begins to distort...as things should be) the corresponding +4 gain is ONLY correct if there are no Intersample Overs (ISOs). Only if I played a recording without ISOs, like my 880 Hz 0dB test disk, would that be OK. Albums with a lot of ISOs would need to be played as low as +0.5 on the Emotiva, if Benchmark's claim that the maximum ISO is about 3.1dB.</p><p>I got plenty of ISO clipping when I tried to copy one of my all time favorite DVD audio discs, <b><i>Pulse</i></b>, by The New Music Consort. It clipped with the Emotiva set to +4dB gain, clipped at +3dB, and so on. Actually, it was still clipping at +0dB, which suggests maybe even Benchmark was wrong, ISOs may occur even greater than 4dB. In this case, I was recording a 24/96 disc at 24/96. I would think that would minimize ISO but maybe not.</p><p>The clipping at 0dB made me think that maybe my filter selection on the Oppo was in not optimal so I checked that. I was thinking perhaps I was using one of those audiophile 'slow' settings. But I found that I was using Linear Phase Fast. That should be fast, in fact I'd think it was the fastest and best so that's why I chose it.</p><p>I looked over all the settings again. I wouldn't want any of the 'slow' settings since those always leak aliases. <a href="http://archimago.blogspot.com/2018/06/measurements-oppo-udp-205-part-1-output.html">I looked over Archimago's investigation of the different filters.</a> IMO the only settings worth considering are Brickwall, Minimum Phase Fast, and Linear Phase Fast. The Apodizing filter is weird having rippled HF response. Minimum Phase Fast is Oppo's default, and it puts the ringing entirely after a pulse and the ringing is quite long, I consider that weird too. The 'Corrected Minimum Phase Fast' is actually a slow filter having similar high frequency cutoff as the slow filters. My subjective judgment of all the graphs and statistics is that the Brickwall filter is best. It has the lowest distortion and noise, and second to the flattest response at 20kHz (according to the Rightmark evaluation) with the Minimum Phase Fast filter being 0.01dB flatter at 20khz, hardly worth sacrificing any noise or distortion for.</p><p>Audiophiles are inclined to dismiss brickwall filters I suspect mostly if not entirely by suspicion. But this is not your grandfather's brickwall filter, this is a very precise brickwall filter. (Archimago has no actual test of high frequency phase, which might show some way it could be inferior. But the response is so flat and extended, phase errors must be pushed out in very high frequencies too. And this is probably achieved without using any of the non causal FIR tricks that the linear phase filters use.)</p><p>Anyway, I tried it, and it might just be chance (caused by dithering or something) but it had fewer clipping events than the Linear Phase Fast filter. With the Brickwall filter I simply had to repair one clipping ISO and there was almost 0.5dB of additional headroom left. For the Linear Phase Fast there were 5 clipping events having to be repaired, including the weird wider than usual one shown below.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiHFleWT5jckNNS-o5V61cgUfZTt9kMVT1OxHehBy9cG6yJHg2tnG92-ZQL7qAI2HUAZ3Zd_GGR-KbOZvHkI8Y_l9BA5YDRvZmz3cp98xHwUA7EC_q2r2ftQt3wwpvPneTlmK5pLvOH97QlXI4q0r6t-fpP4KhIL1_2KpbfcOH7tY0VbjKsJyrt1fP1/s1920/Brickwall%20vs%20Linear%20Phase.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiHFleWT5jckNNS-o5V61cgUfZTt9kMVT1OxHehBy9cG6yJHg2tnG92-ZQL7qAI2HUAZ3Zd_GGR-KbOZvHkI8Y_l9BA5YDRvZmz3cp98xHwUA7EC_q2r2ftQt3wwpvPneTlmK5pLvOH97QlXI4q0r6t-fpP4KhIL1_2KpbfcOH7tY0VbjKsJyrt1fP1/s320/Brickwall%20vs%20Linear%20Phase.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">Brickwall (top) vs Linear Phase Fast</td></tr></tbody></table><p>The Brickwall doesn't clip with this apparent digital error on the recording shown above, whereas the Linear Phase Fast does cause clipping (above +4dB headroom allowed for ISOs). And the Brickwall ringing is slightly more compacted and seems to make more sense. Generally, I liked the way the ringing on the ISO's looked with the Brickwall, in addition to having fewer of them clip. I think I'm going back to Brickwall (it had been my option until a few months ago). My feeling is that the Brickwall is less tricky and more honest than the other filters. Among the filters, Brickwall may well be the mathematically least complicated. Many of the others HAVE to be implemented with FIR digital filters because they are acausal. They are approximations of things that are impossible to achieve with ordinary circuits. Brickwall can in principle be implemented with IIR filters (but the Oppo may well use a FIR to approximate a more perfect Brickwall).</p><p>Indeed all the ISO clipping events on this recording are caused by ringing on very rare and unusual short impulses which look like digital errors. Filtering out those pushes the peak level down to -1.8dB. But even THAT peak level is still being caused by ISOs. With no ISO's, the level should be -4dB, allowing my LF sinewave derived level setting of +4dB gain. Here is what another of the ISOs looked like with the Linear Phase Fast filter. Note that it appears to stem from a transient less than 4 samples wide at 96kHz (actually the impulse itself is less than 2 samples wide, or about 48 kHz, and therefore unlikely to be of acoustical origin). Repairing some of these peaks may make them MORE audible as the inaudible high frequencies are replaced with audible lower frequencies which probably shouldn't be there either. All the better to have fewer ISOs to repair, as with the Brickwall filter.</p><p><br /></p><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhYt8Gr-jBx63dPiPQ2yg-DLqIfyeFJo1Wu9jvao3mnzXr4v3ksHqEsfIJ8AzrVuqAkwFCG-8WwsTtZHldR2T2xRYGh_RxOmjx8-dSX4VpcQpmH0Mozm66bfqcrlTDaYzEYGVx_BUDhd4Gx8aT4E1iq_zlelD9ZeytJxH2IadYo0PgOOZ6qCmo6qBPt/s1920/Four%20Samples%20Wide.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhYt8Gr-jBx63dPiPQ2yg-DLqIfyeFJo1Wu9jvao3mnzXr4v3ksHqEsfIJ8AzrVuqAkwFCG-8WwsTtZHldR2T2xRYGh_RxOmjx8-dSX4VpcQpmH0Mozm66bfqcrlTDaYzEYGVx_BUDhd4Gx8aT4E1iq_zlelD9ZeytJxH2IadYo0PgOOZ6qCmo6qBPt/s320/Four%20Samples%20Wide.png" width="320" /></a></div><br /><p>I made the last Pulse recording with the TV that I had used to set the filter choice still running. To see if that made any difference, when the recording had completed one more pass (it keeps repeating endlessly on these Classic Audio DAD's) I stopped it, and kept recording for a minute, then turned off the TV, and tried to run for another minute (but it was shortened to 40 secs because the audio file had reached maximum length). Amplifying the noise by 94dB, the point at which the TV was turned off is simply not visible.</p><table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto;"><tbody><tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjaoUaTLIVN8WpAW00TK_lJ4pU9qKeVH6LksvmjXB0Ie-emQXOB_PRpF5a1CZNOJ97lqotRoPk-kCCSPrlPOBEmV7s2QZ74FHGtTSE7utF6m_Jk1iGxlslLo0nvgIBP3oaY1OgCxHzDVZSrvDu0p5FXMQ7FRPUQQlEPXcYhoHQjurfbbequLObgVpwU/s1920/TV%20ON%20then%20OFF.png" style="margin-left: auto; margin-right: auto;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjaoUaTLIVN8WpAW00TK_lJ4pU9qKeVH6LksvmjXB0Ie-emQXOB_PRpF5a1CZNOJ97lqotRoPk-kCCSPrlPOBEmV7s2QZ74FHGtTSE7utF6m_Jk1iGxlslLo0nvgIBP3oaY1OgCxHzDVZSrvDu0p5FXMQ7FRPUQQlEPXcYhoHQjurfbbequLObgVpwU/s320/TV%20ON%20then%20OFF.png" width="320" /></a></td></tr><tr><td class="tr-caption" style="text-align: center;">TV turned off midway here makes no difference</td></tr></tbody></table><div><br /></div>That noise level, around -99dB peak unweighted*** (so approximately -120dB A weighted) is way lower than on the recording itself. When I amplified the initial part of the recording, including before the Oppo started playing the disc, you can clearly see (at 36dB amplification) where the disc starts playing, the noise just rises out of nothing (the earlier -97dB peak unweighted noise from my chain of equipment is invisible at this level of amplification):</div><div><br /></div><div class="separator" style="clear: both; text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhaVAEGdU5xIyghcNBD9UyVN1EGwBqvtbhZbAaAIGPaZJa73i5Cq669qNRhuqt2tvr0AFyrZx_fRkavYM-zwxMYsCzEtzS3H-Te1ZGqAKLIWxq3CLkWTqsilRhJyfzuPkuGZVn5H2EE5z5lmtU_E27hddUolPiqFP9eHsN-YO7suYUI21tAgJhR8_r6/s1920/Screen%20Shot%202023-05-12%20at%2011.24.32%20PM.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="1080" data-original-width="1920" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhaVAEGdU5xIyghcNBD9UyVN1EGwBqvtbhZbAaAIGPaZJa73i5Cq669qNRhuqt2tvr0AFyrZx_fRkavYM-zwxMYsCzEtzS3H-Te1ZGqAKLIWxq3CLkWTqsilRhJyfzuPkuGZVn5H2EE5z5lmtU_E27hddUolPiqFP9eHsN-YO7suYUI21tAgJhR8_r6/s320/Screen%20Shot%202023-05-12%20at%2011.24.32%20PM.png" width="320" /></a></div><br /><div>*** The actual value of noise measured may depend on how long you measure, when you measure peak noise. There is sooner or later always a higher peak. This is especially true given the ultra low frequency noise found in the Emotiva left channel...it is the primary driver for this noise dispersion, despite being at a tiny -103dB peak level itself. In one second that low frequency level has barely changed, but in one minute it can do a lot of up and down dancing.</div><div><br /></div><div>So the numbers I'm seeing now from the Emotiva at +0, the Lavry at -8, taken from the interval shown in earlier picture before where the TV was turned off, are these</div><div><br /></div><div>1 second<span> <span> </span>-100.5dB</span></div><div><span>10 seconds <span> -98.7dB</span></span></div><div>1 minute -97.7dB</div><div>2 minutes -96.7dB</div><div><p>An advertising dept might see how small of an interval they could possibly go... But nobody uses peak noise for specifications anyway, they use average noise weighted, but I can't do that as easily with Audacity.</p><p>I might establish a "10 second standard" for measuring this, though brag about the 1 second level.</p><p><br /></p><p>(*In sighted listening tests.)</p><p>(** I know that DVD-Audio discs can be played through HDMI which preserves the full original 24 bit resolution, though it's still not possible for either HDCD or SACD on my system. But it's still way better to have PCM computer files on my hard drive for automated playlist playback, and no files can be created from HDMI because it inhibits digital copying. Unlike some, I believe 24/96 digital encoding and decoding is essentially perfect for any audio purposes, and the present results support this by showing digital conversion to be good enough to show flaws in an excellent preamplifier. I see no reason to store DSD files as they would later need to be converted to PCM anyway for playback through my digital crossovers and equalizers. So I might as well do the conversion (SACD, HDCD, or DVD-Audio to analog to digital 24/96) before storage, and it saves me a lot of hassle later too, when I'd be using the exact same components to produce 24/96 for my crossovers and equalizers anyway. )</p></div><br /><br /><div class="separator" style="clear: both; text-align: center;"><br /></div><br /><div class="separator" style="clear: both; text-align: center;"><br /></div><br />Audio Investigatorhttp://www.blogger.com/profile/17831968195627932789noreply@blogger.com0