On Mon, 2011-06-13 at 16:04 +0200, Arnold Krille wrote: I guess in other countries they are less petty as we are or at least were for broadcasting here in Germany, regarding to what is allowed for airplay. Years ago it wasn't seldom that reports with minor phasing issues were played "einbeinig" (one-legged, muting one channel and using the kept channel for left and right). Regards, Ralf

2011/6/14 Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> then, what's the need for those invalid tests? :) moreover, I would not say old blues recordings are terribly broken, 192Khz super-ultra-mega-super-funky AD/DC to analog converters won't save if the music you are playing/recording with is... brokenù :) regards -r -- *L'unica speranza di catarsi, ammesso che ne esista una, resta affidata all'istinto di ribellione, alla rivolta non isterilita in progetti, alla protesta violenta e viscerale.*

On Tue, Jun 14, 2011 at 8:11 AM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: You DIDN'T SYNC ANYTHING. Or at least, your description doesn't mention any kind of sync. And if you had tried to sync, you would have found that you CANNOT sync two A/D converters unless they are the exact same model (so that their pre-A/D buffer is the same size).

On Tue, Jun 14, 2011 at 10:03 AM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: you're really deeply confused. let me try to illustrate (the diagram is somewhat simplified to try to make the point more clear. the actual reality is a bit more complex in terms of where the clock is actually used, and so on). the section marked "buffer" refers, typically, to some RAM into which a relatively large (say, 32 or more) samples can be stored "in front of" the D/A converter. digital samples-> buffer-> D/A converter digital-side>internal buffer->D/A converter analog side ^ | | sample clock the sample lock says "get another sample", so the device will fetch another sample at precisely the same time (to within a picosecond or so) of other devices using the same clock (if any). this is important. but ... the way each D/A converter circuit works is independent once it has the next sample value available. the amount of buffering inside the D/A converter is variable from converter to converter, as is the actual conversion process itself. So lets put 2 of them together: digital samples-> buffer-> D/A converter digital-side>internal buffer->D/A converter analog side ^ | | shared sample clock | | v digital samples-> buffer-> D/A converter digital-side>internal buffer->D/A converter analog side it makes no different whether the sample clock is delivered from an internal clock crystal, the ADAT cable, word clock via BNC or any other method. once the samples are delivered to each D/A converter, there is no "sync" anymore. this has nothing to with RME, and everything to do with the basics of digital audio and digital-to-analog convertion.

On Tuesday 14 June 2011 16:03:35 Ralf Mardorf wrote: All ADAT (or any other word-clock) syncs is the sample-clock of the processing, it doesn't sync the times the converters need to output the sample as analog audio. And these times depend on the actual chip used. So you can't even expect one adat-io-converter from one manufacturer and another by the same manufacturer to be of the same "latency" because they might use different chips inside with different conversion latencies. To have it exactly in sync, you have to use only one kind of converters in your setup. Or live with the fact that one sample at 48kHz is only some centimeters of distance, and the conversion-latency is usually below that. So its probably okay for normal recording-, live- and playback-usage. Have fun, Arnold

On Tue, Jun 14, 2011 at 6:48 AM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: no, you did make a mistake. your test (to the extent that i understand it) didn't sync anything. you used two different D/A converters without a common word clock. even with the same word clock, they would not necessarily be synced because they may buffer different number of samples internally in front of the actual D/A circuitry. but this is sort of irrelevant because you didn't actually sync anything together at all as far as i could tell. first of all, CD players don't do direct digital audio playback. the audio you get telling the disc to play an audio CD is different than you get when reading it as a digital storage medium. secondly, you appear to have made the cardinal error of trying to do psycho-acoustic testing without using even blind, let alone double blind testing. this means that your assessment is meaniningless. if you don't believe me, try watching (at least the first 2 speakers) of this: http://www.youtube.com/watch?v=BYTlN6wjcvQ (a video that anyone with any pretensions to being an audio engineer needs to watch, assuming they can listen to english). please note that i did not that say that your assessment is *wrong*. but the way you did it is. your conclusion about sound quality might be right, but its equally likely to be wrong.

On Tue, Jun 14, 2011 at 8:31 AM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: there is zero reason to expect them to be synced.

On Tue, 2011-06-14 at 15:23 +0200, Cedric Roux wrote: I worked for Brauner development. You can't replace a Neumann, Brauner, Schoeps etc. by a cheap microphone and an EQ. get a Haun capsule or similar and build a less expensive microphone yourself. I don't have a good microphone myself. Regarding to the music and recordings you do, you don't need a good microphone, but as soon as you'll record more than just your Rock'n'roll vocals by Shure classic sound you need a good = expensive microphone. An EQ only can readjust what is existing, but just is too silent or too loud. You can't readjust response characteristic, directional characteristic, missing frequencies.

On Tue, 2011-06-14 at 21:06 +0000, Fons Adriaensen wrote: FWIW those microphones among other things are that expensive because there's much failures for the capsules. The directional characteristic for large-membrane microphones is set by the amount of the level for the front and backside. For a Neumann and Brauner a capsule's front and backside is as equal as possible. If a capsule hasn't very similar front and back sides they still can be used for cheaper microphones, that e.g. just need a front, but no backside. You can assume that for less expensive microphones they use every capsule without making a selection. For expensive microphones they take care for nearly everything, even the transformer plates are important. The only thing they do not, is selecting semiconductors. Some people build custom versions of famous microphones, with selected semiconductors and voodoo cables. We compared those custom microphones with the standard versions and there was no difference. Just the tubes are important. No china ware! If you take a look at the capsules you will see differences for the wholes. Some vendors do long time trial and error development regarding to this and others just use capsules with no-matter-how holes or they guess it's enough to copy the drill wholes of good microphones, but it isn't that easy. The power supplies of expensive microphones IMO have an unneeded filter overkill, but it's better to do more than is needed than to use a power supply with bad filtering. Those companies pay engineers for doing the soldering, it isn't done by children and housewives, at least it wasn't done some years ago. You might compare the soldering of instrument cables. Some days ago I got some Thomann cables and some Cordial cables, both are cheap cables, but the soldering for the Thomann is very bad and the soldering for the Cordial is very good. Bad soldering soon or later will cause issues. And last but not least, they pay idiots like me to listen to the microphones before being shipped and the boss will listen to those microphones too. Of cause, a cheap dynamic microphone for the vocalist on stage is better than a expensive large-membrane microphone, so the usage is important too. I heard room sound for orchestras done with large-membrane Neumann that was incredible. You only can do this with Neumann and Brauner large-membrane. Btw. even if it should be possible to improve sound of cheap microphones to get a sound near to an expensive microphone, you need expensive anlog EQs or an expensive converters to use digital EQs, if not the EQs won't improve the sound, but rather add new issues. Note! An analog EQ can cause phase issues, so a tip that might work with top-notch equipment by no means is a good tip when using home recording equipment. I never met a guy* from AES or VDT who wasn't a good engineer. *(I never met any engineering women *lol*, neitehr gifted, nor untalented.) If those folks pay for expensive microphones, they do it for good reasons. If those guys claim that you can do a lot with an EQ, they perhaps aren't talking about cheap EQs. It's possible to do a good mastering without compression, you just need very good EQs to place instruments and vocals within the 'room' of the frequencies. It's hard to this at home or with semiprofessional equipment. Recordings can sound as good as reality ex 48 KHz, when they were done with good = usually very expensive equipment. This get lost for CDs, but even when listening to CDs the responding qualities of different microphones are audible. You can hear Peter Gabriel's breath with transparency for his studio recordings, you can't hear Rory Gallagher's vocals on the Irish Tour album that transparent. And here it is again. A studio recording hasn't the energy of a good live performance, hence it's important to have a better sound quality. Rory Galagher live '74 has so much energy that there isn't the need for a sound quality that is needed for synthetic music like it's done by Peter Gabriel. IMO the needed quality depends to the kind of recording. Recoding a Jazz trio is easier to do than recording 30 synth, just because there's more room to allocate the instruments within the 'frequency room'. So IMO there aren't no universal tips, like use the EQ to do this and that, certainly there are some rules of thumb. Hifi speakers get stressed by some frequencies that can't be played by those speakers, so you might cut them etc.. This and similar rules of thumbs are helpful. Oops a novel, but it's just broached. 2 Cents, Ralf

On Tue, 14 Jun 2011 07:22:54 -0400 Paul Davis <paul(a)linuxaudiosystems.com> wrote: Wow. I've been able to listen to about 20 minutes of this and I wanted to say thanks for posting this link. This really is a fantastic opportunity to hear experts in the field clearly explain things that are really important for people who hope to advance their amateur recordings to another level. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa

On Tue, Jun 14, 2011 at 12:48:42PM +0200, Ralf Mardorf wrote: Having measured a lot of RME gear I'd say that if you hear any sonic defects on it using the sort of test you described then the chances are one to a million that you are just deluding yourself. *IF* your hearing is really as acute as you suggest it is, then the first thing you should be able to detect is that all the channels of your Behringer mixer sound different, much more so than the difference between any two pieces of RME (or any other decent brand) equipment. Which BTW is one more reason why your test is completely invalid. Ciao, -- FA