danbronson

I don't know if this is valid, perhaps someone here could comment, but I've thought about hanging carpet a bit out from the wall so the sound wound be dampened a bit as it passed thru the carpet before reflecting off the wall, then attenuated a bit more as the reflected sound passed thru the carpet again. As mentioned above, this would cut down on the reflections but not provide proper 'treatment' Yes, this is the best way to do things. With carpet, an inch or two should be fine. The result is an extended (lower) frequency range that can be absorbed. It'll be more significant with thick bass traps though. One other thing to mention... If you're going to hang carpets, make them the thickest carpets you can find. If you have a lot of them, rolling a few up and placing them in the corners of the room can make for a decent on-the-fly bass trap. It is usually more important to deaden the lower frequency range than the highs. High end resonance in a room just sounds like reverb. Low end resonance can create huge comb filtering problems.

Hanging a carpet or two will definitely deaden things in the higher frequencies (which may or may not be a good thing, you'll have to figure that out for yourself by experimenting because every room is different). The result will sound less 'live', though any low end and midrange problems (which any room in a typical house will have) will not be fixed. The uncontrolled low end will cause monitoring problems mainly, though your recordings will suffer if you set up room mics too. Low end/midrange issues can be treated somewhat effectively with bass traps, though that is somewhat expensive and/or time consuming (I have made a few) and they rarely solve a problem completely. To really stop these problems, you need a room designed with acoustics in mind. Acoustics are one of the most important aspects of recording (all the rack equipment in the world won't help if you record/mix in a shitty room) and one of the most difficult/expensive to get right. Not trying to sway you from setting up a home studio though. Looks like you already have realistic expectations.

Like I said, the change in transient response, smearing and (what sounds to me like) low pass filter are what is making the clips sound different on a whole. That's what I'm hearing, not performance differences. I paid extremely close attention to how I performed each test to get my performances as alike as possible and the mic positioning has not changed. If my playing were inconsistent, you would hear it within each clip. Even if inconsistent playing could make the changes you hear, it would be very unlikely to affect each entire clip in a completely different, yet consistent manner. For you to assume that it has downright stupid, hence my frustration. Go ahead and reread that paragraph until you understand every word, because it's certainly not getting through to many people here. So there's that, but there's also the fact that I performed three separate tests (which you've heard two of, the first was deleted because I wanted to do it again to verify what I heard the first time, therefore minimizing the possibility of error), all with the same results. I also recorded a pass at 96 kHz and added it to the guitar clip, just before the 48 kHz pass (so it is first). You can hear that they are very similar sounding (96 sounds very slightly better than the 48 to me, just my opinion), while the 44.1 kHz clip without r8brain conversion sounds dull, muddy and lifeless. http://dl.getdropbox.com/u/1395881/96%20then%2048%20then%2044.wav So that's seven seperate performances in total (4 on the cymbal, 3 on guitar). And every time, the recordings that started at 44.1 sound dull and crappy compared to recordings with higher sample rates. You'll have to take my word for it on the cymbal clip you didn't hear. I deleted it because I wanted to use the same file names for the second test. Needless to say, the 44.1 clip suffered from the same dull sound. Obviously we're dealing with two variables here (performance and sample rate) and it's not possible to isolate them. But if the test is repeated more times with the same results, the accuracy of the findings goes up. The only problem is, I'm not going to repeat this test a hundred times to satisfy you guys. Playing this stuff consistently, seven times is enough for me. So let's leave it where it is. Nothing more needs to be said. If you want to argue more, read through the posts I've already made because I've said all I need to say. Make up your own mind about the findings I've posted here. /last post in thread

If your built-in SRC algorithm truly sounded bad, then I promise you that it would impossible to improve it by running an additional SRC algorithm on top of it. This is assuming I accept your theory about AD conversion requiring a built-in SRC algorithm, which I do not. I do appreciate your post, though. I just don't agree with the assumptions you made going into your test. They're not assumptions. Do a little research. I'm not going to spend an hour or two reading and quoting just to respond to you accurately (even if I'm generally right, I would probably miss a detail or make a generalization at some point that would open up a whole new bag of {censored}) so feel free to do it yourself. Or, if you're certain I'm wrong, feel free to explain why. EDIT: Looked into it a bit now that I have time, and I've been misinformed. I was under the impression that all converters worked like DSD audio (1 bit/2.8224 MHz). But all that is really beside the point, which is that regardless of the way you get to 44.1 kHz, there will be filters applied to the audio that can potentially sound terrible. Therefore, two different ways of getting to 44.1 kHz can potentially sound different. And in this case, they do.

I'm in a bit of a rush right now so I can't go into detail or double check what I'm about to say but... When an A/D converter puts out a digital signal, it doesn't start as 44.1 kHz or 48, 96, etc. I think (I could be wrong) but it starts out as a 1 bit file with an extremely high sample rate. The waveform is represented as accurately as possible in an extremely primitive form (1 bit = just 1s and 0s, on and off). The A/D converter isn't done yet though. It uses a built in sample rate converter (the same thing you'll find in software, like r8brain) to send the audio to the computer at the desired sample rate. The built in sample rate converter can potentially sound terrible. That's why it can be beneficial to record at a higher sample rate (where there will be more allowable detail and the filters will be much further outside the range of human hearing). In my case, it is beneficial. I encourage everyone to check with their own converters. Voxengo r8brain is free and very high quality. Do a test like the one I did and listen for yourself. If you don't hear a difference, then the built in sample rate conversion in your A/D is good enough for you. If you do hear a difference, use the method that sounds better to you.

I hope you can pardon a perhaps overly broad analogy but that seems like trying to choose a paring knife by comparing paring knife A cutting an orange with paring knife B slicing an apple. I recognize you're not trying to do a scientific test, but I'm afraid I just don't see the point of a comparison where you cannot isolate the factors that you're setting out to test. Feel free to propose a way to do this. I would love to remove that one last variable. Until then, it is what it is. I feel there's a lot to be learned from this comparison. But maybe I'm the only one who bothered to even listen to it. Perhaps next time I will just keep my findings to myself unless I happen to run them by CERN first. I only posted so that people on the forum could learn what I have just learned. But if you guys want to waste your time chasing impossible testing standards, be my guest. I'll be over here improving my ears and recording skills in the mean time (starting with using higher sample rates on the way in and converting later).

So basically, this is not a sample rate conversion shootout. It very much is. It compares: -TC Studio Konnekt 48's conversion to 44.1 kHz -TC Studio Konnekt 48's conversion to 48 kHz + Voxengo r8brain's conversion to 44.1 kHz Note that it is not possible to compare the r8brain on it's own to the TC SK48, as the SK48 must do some conversion if it's doing A/D conversion. Therefore the best way to test is to use the SK48 at a higher sample rate (which will be more transparent) and use something else to bring it back down to 44.1 kHz. Put plainly, this is a comparison of two different ways to get audio to 44.1 kHz using the same A/D converters.

I'm not arguing with your opinion, I'm just curious why a different sample rate would affect transient response. I'll listen to the clip tonight and give my opinion. Yeah I didn't think it would be able to do that either, but check out the link blue2blue posted. It becomes a little more obvious how transient response can be effected by sample rate conversion.

Call me lucky I guess, I was taught scientific method in school. I'd use it if I thought something like this was important enough (not that I can anyways), but I guess I don't take myself that seriously. It's just a sample rate conversion comparison after all. I stand by what I said about a good ear being able to discern what sounds different due to performance differences, and what sounds different because it's making an entire take sound cloudy and dull. :/

At the risk of offending, that is such a frustrating cop out of an answer. :/ A good ear (with good monitoring) should be able to hear which differences exist due to performance, and which exist due to sound quality differences. For example, if a low pass filter were applied to one clip at 5 kHz, the lack of top end in just one clip would obviously not be caused by the fact that one is a different performance. This situation is not far from that example. Are the performances different? Of course. Are those differences going to make one clip's overall spectral balance and clarity noticeably different than the other? Hell no. Don't kid yourself. Actually listen to these things, the way they differ could never come down to performance... To me, the differences in these clips are not subtle. I'm surprised nobody has said anything. Even my girlfriend and parents reported almost the exact same findings in blind tests (and with no communication with each other). And frankly, I agreed with every word they said. No cop out answers, they just said what they heard. What I'm hearing is this (since nobody else has the guts to voice an opinion )... The first halves of the clips sound good to me. The top end is quite apparent, making things perhaps a bit thin sounding. Nevertheless, things sound clear and punchy to me. I listen and feel like the sound is actually happening right in front of me. Cool. The second halves of the clips sound very, very dull and 'smeared' in comparison. The transient response is basically gone, making it seem like the sustain to attack ratio is higher and therefore things lose a lot of punch and also sound more compressed. Also, it sounds like there is a low pass filter on the top end (and indeed, there probably is since the differences here are in the sample rate conversion method). The lack of high end means the lows/mids seem much louder in the second halves, and it's here that the 'smearing' sound is so obvious. There is just an overall lack of detail in the whole frequency range when compared to the first halves of clips. It all sounds like dead, muddy {censored} comparatively! For what it's worth, I did the cymbal test twice (but only posted the second one) to make sure that the results would be consistent between both tests (and of course, further tested with an acoustic guitar for three tests in total). What can I say? Exact same results every time. So, since nobody's playing...the first halves (bright and lively, remember) are recorded at 24 bit/48 kHz, exported from Cubase 5 (24 bit/48 kHz) and converted to 24 bit/44.1 kHz with Voxengo r8brain free. The second halves (dull and dead) are simply recorded at 24 bit/44.1 kHz in the first place and exported that way as well. Then they are all brought into the same 24 bit/44.1 kHz Cubase project where they are level matched and the volumes raised (without gain reduction) in Voxengo Elephant on the master buss. They are also dithered to 16 bits with Elephant and exported as a 16 bit/44.1 kHz wave file. This sound quality difference is a pretty huge revelation for me, as it means I can get a much better clarity in my recordings by recording at a higher sample rate and converting later with r8brain.

Here's a handy SRC comparator: http://src.infinitewave.ca/ It takes a second to figure out the interface -- it's set up to select one SRC on top and a different one on the bottom and then compare aspects of their tested performance. I can't vouch for the rigor or methdology but they've been updating this thing regularly for some years now and no reason to not think it's as informative as it looks. Funny you'd post that, this site was the inspiration for these tests! I guess people either aren't hearing a difference in these clips or are too worried about being 'wrong' (not that there is a 'wrong' here) to post, so I'll spill the beans on what the differences are. Until now, I've always just recorded at 44.1 kHz and left it at that. The files are smaller and they don't have to be sample rate converted to become CD quality, just bit truncated. I figured this was saving me some audio quality loss but I guess I never really thought it through till now, because even if I'm not converting in software, my A/D conversion is still sample rate converting to 44.1 kHz! So you can't really get around it. Either way, you're sample rate converting at some point. Knowing this and seeing how differently software can test from the site you just posted got me wondering... I changed Cubase (and therefore my converters) to record at 48 kHz. Then I downloaded Voxengo r8brain (which tests extremely well on that site) and used that to convert to 44.1 kHz. I compared that to a clip just recorded at 44.1 kHz as I have always done before. That's what's going on in the clips I posted. Can you hear it? Which one do you think is which?

The biggest difference is the performance. It is definitely unfortunate that I couldn't use the same performance somehow. But I'll say this, they're not as different as you might think. I found that one method of recording completely changed the transient response (or at least that's what my opinion is, I want to hear everyone else's!), making one seem as if I were hitting harder and/or in a different place on the cymbal. But in truth the performances are very, very similar. I guess you'll have to take my word for it though. Still interested to hear what people's opinions are. It's not a trick or anything, I promise! Just let me know how you think the clips compare. It'd be nice to get some discussion going here, even if people disagree with each other (especially if they do!). Here is another clip. The order (concerning the way these were recorded) is the same as in the first clip. This time it's two acoustic guitar clips (about 6 seconds each). The two clips have been RMS level matched to within 0.01 dB of each other. This'll show how an instrument with a wider range reacts to the two methods of recording (I will explain them once I get more opinions, but I don't want to bias people and I've already said too much). Hopefully this makes it more obvious than the cymbal? http://dl.getdropbox.com/u/1395881/SRC%20Shootout%20Acstc%20Gtr.wav

Thanks for listening. Anybody else?

Hey, I have a 24 second long CD quality wave file here (it has been dithered from 24 to 16 bits with Voxengo Elephant, no limiting however). There are two different performances on a ride cymbal (first at 0:00, second at 0:12) that were recorded/sample rate converted in different ways. Their RMSs have been matched to within 1 dB of each other. That's all I'll say for now. http://dl.getdropbox.com/u/1395881/SRC%20Shootout.wav I want to know what you think of the differences. I've already made up my mind, but since I can't do a blind shootout on myself, I want your opinions as well. Are there big differences? What are they? Which half of the clip do you prefer? Maybe the differences are smaller or not noticeable to you? Please let me know! Thanks!

OH NOES!1 quick! tell him he's a stupid head!

Let me preface this with the first rule of recording: #1: Never pass audio through anything with Behringer written on it. Okay, now that that's out of the way... The best decision I've made in recording was to only go with the highest quality gear I could get away with. This means I have a lot less gear than some people, but I can make a better recording in the end. And I'll spend less in the end because I'm buying gear I may never replace. I hate excess {censored} that I'll never touch. If I had a $60 condenser, it would very likely become excess {censored}. So for me, it would be a waste of money. So that's my perspective. Now decide where you stand on this.

are the guys I would contact first... then there is Dave Collins, Ted Jensen, Howie Weinberg, Bob Ludwig, blah, blah, blah, blah, blah. These are seasoned professionals who can take your demo to come a hell of a lot closer to sounding like a record that you will be able to do at home on your best day. If nothing else, spend the money to book a session with one of these guys and have them at least begin to explain the process to you. They have decades of experience, are exceptionally good at what they do... but are also really cool guys who will be happy to take some time and at very least give you some of their knowledge with which you can begin to experiment. Peace. I master all my own stuff because I enjoy it. I know I'm fighting a much harder fight, trying to impress people with sound quality when I've got a fraction of the gear and experience, but it's fun for me and I love the challenge, even if it's impossible at this point (though again, I'm sometimes surprised that my masters sometimes sound better than pro ones...that's just a little sad). But let's face it, if all the guys you just mentioned had just 'left it to a professional' instead of getting their feet wet years ago, you wouldn't have mentioned them today. Not that I expect I'll be one of them, just making a point. Everyone starts with nothing.

Oh, I should also mention that the cheapest useable software limiter I've ever used was Voxengo Elephant. Go easy on it (it gets pretty gritty under heavy limiting, but not necessarily in an ugly way) and you will get good results. There are a few free ones out there but none that I've tried were worth the time it took to download them. Cubase's built-in limiter is not great either. Elephant and Ozone are both capable of musical results.

What you need is a limiter. My advice: -read up on wikipedia to figure out what a limiter is exactly and how it is similar and different to regular compression. -read everything on this page and download the plugin (it's free) -download iZotope Ozone (this is the highest performance to price ratio available for mastering software) Use Ozone's loudness maximizer (that's the limiter), but also figure out what everything else is and how to use it for best results (the multiband compressor is especially helpful in my experience). Try to avoid using presets. Place the SSL X-ISM after Ozone. Run through a song before exporting it while watching X-ISM. You will probably need to set Ozone's output levels to -0.1 dB or so to avoid peaks. When everything's good, disable X-ISM and hit export. Bam, instant amateur mastering. If your ears and monitoring are good, you will be able to churn out better masters than some pro mastering engineers. No, I'm not kidding (let's face it, some commercial mastering is abysmal).

No way Dan. According to Ethan, you couldn't possibly know that unless you've done a 100% scientific test. Some of you may be interested in the results of this thread: http://www.gearslutz.com/board/so-much-gear-so-little-time/395087-kenny-gioia-ethan-winer-test-converters.html Despite the fact that I'm curious about what the results will be, I think I'd rather debate with Creationists all day than read through all that!

Lol, I forgot about this thread. I finally upgraded from my Firepods to a TC Electronic interface. Holy {censored} does my D/A sound better! You'd have to be deaf not to hear the difference.

No kidding, and the graph below from my Audiophile Beliefs article shows this in no uncertain terms. The two responses were measured only four inches apart in a 16 by 11 by 8 foot untreated room, yet they're so totally different you'd never guess this is even the same room and loudspeakers! --Ethan Hey look, a trump card for every a/b test ever. Fwiw, I tested my Firepod vs. analog thing with headphones as well (because their bass response goes much lower than my speakers). Same result. The loss of bass was more obvious actually.

Exactly. Everyone understands and accepts that the placebo effect is real, but for some reason audiophiles don't think it ever happens to them. For all I know the Firepod really is a total POS. I already said I don't have one, but I have heard and used plenty of mid-level sound cards that are transparent and indistinguishable from high-end converters. Here's a blind challenge for everyone who is certain that all prosumer grade converters are crap. Below are two short files for you to identify. One was recorded through a $25 SoundBlaster X-Fi card, and the other through a $6,000 Apogee. This is one performance captured through one microphone that was split after the preamp, then sent to both converters and recorded at the same time. Anyone here care to identify them? Clip 1 Clip 2 After enough people have posted their guesses I'll reveal which is which. --Ethan If someone set a tiny model of the Eiffel tower up a few inches away from you, and right beside it in your field of view was the real thing hundreds of feet away, it may be impossible to tell which one was real if you had to look through a window so dirty it mangled everything you saw through it. Same with converters or any other piece of audio equipment. I could never tell you which is which because I'm listening on converters that make everything sound like junk. Now take into account the fact that, personally, I don't have any experience with Apogee or that SoundBlaster and I will gladly tell you I cannot pick which is which. Unfortunately, a reasonable test would be very difficult to do. A reasonable test would be comparing both the SoundBlaster and the Apogee to their all-analog equivalents to see which one is the better representation of what that signal is actually supposed to sound like. How would you compare the way those converters stack in a dense mix (for example 25 converted tracks all played together)? What D/A would you use anyway? The only way to know which is better is by having extensive recording and listening experience with both units. Now for all I know, the SoundBlaster is better than the Apogee. It's highly unlikely but it is possible. The truth is I have no experience so I can't comment. And it's not something I'd be asinine enough to speculate on. All I know (for the last time) is that when you throw my Firepod's converters into an otherwise good sounding signal path, things start sounding bad.

Dan, I actually think its good you can bypass the converters and compare the two. You can actually compensate for some of the alteration in the process by goosing the source a bit. Totally. Since I've realized what's going on, I've learned how to compensate. I have a few tricks up my sleeve now so I can fake it a bit. Still not perfect, but closer to a 'real mix'. In any case the differences may be great from what I'm using but it doesnt stop anyone from creating great music. Even with its limitations an 8 track board can do wondorus things if creative minds are being used. You bring up something we should all remember. At the end of the day, just enjoy making music. There's nothing wrong with trying to get the best sound you can, but when you get lost in the details, it's not worth doing anymore.

* I see on the page above that the FirePod has an input impedance of only 1,600 Ohms, which is awfully low. So depending on what device Dan has feeding the FirePod, it's very likely this is the source of the loss of fullness. Especially if the previous device is "prosumer" type gear. * Another factor for loss of fullness (or clarity at high freqencies) is level matching. When doing a proper A/B comparison, whether blind or not, it is absolutely critical that both A and B be level matched. A difference as small as 1 dB can make the louder version sound fuller and clearer. The standard for proper testing is to match levels to within 0.1 dB. --Ethan Preamps used were the Firepod's. Like I said, the converted signal was louder. To some this might make the converted signal sound better. It was clearly very bad sounding though by comparison. It was louder in a very "oh god please turn that down" kind of way. The all analog signal path was much more ear pleasing. It sounded wider and the response seemed flatter, more like what I feel the amp I'd mic'd actually sounds like. You say specs don't lie. Obviously they don't. But companies sure do use them to make their product seem better. My last speakers were M-Audio BX8a's. They advertised a very flat response. On paper, they look like the perfect speaker. I bought into it. After a while, my ears couldn't take it anymore and I ponied up for a set of Adam A7s. The difference was astonishing. Speaking of bass, the A7s (6.5" driver, 46 Hz - 35 kHz overall response) have a much deeper, more accurate, tighter, better sounding bass response than the BX8a's (8" driver, 40 Hz - 22 kHz overall response). The specs seem to disagree. Ethan, you're implying the Firepod's converters are transparent (or close to it) and that the difference I heard is because the analog signal path has a problem with it...a very nice, natural sounding problem. Right? Please. And of course, you also state that any $100 converter can perform as well or close to any other converter in any price range. Why? Because it looks accurate on paper? Or is it because when people skip the expensive interface, they'll have more money to spend on acoustic treatment? It boggles my mind how you can fly in the face of logic, evidence and black and white personal accounts and say what might as well be "the sky is not blue". But then again, I guess you have a motive.