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HotCanadianChick posted:I've got a Delta 1010 that just up and died a few months after I bought it, and I once bought one of their controller KB's that had a couple keys that flat out didn't work right out of the box. I had an old Audiophile 2496 whose MIDI would stutter and drop notes constantly if you tried playing more than a few notes at a time through it (such as sending out data to a couple different synths at the same time). Do they still sell Delta 1010? I was selling them 7 years ago at Guitar Center and they were old then. My Konnekt SK48 from TC shipped with a bad preamp board which was a problem with their first run of manufacturing, and that was a $1000 interface. It's just a fact of life that competition has forced most if not ALL manufacturers to look for cheaper ways to produce their products, and that includes going with cheaper PCB and circuit stuffing facilities, using cheaper component parts, and faster soldering on the assembly line. Some manufacturers are worse than others, but there are very few manufacturers that spare no expense when it comes to avoiding defects. I don't think this is really an issue of spending another $25 on a competitor, if you want truly professional gear that is ultra reliable then you're buying in a much smaller pool of customers, so prices are going to be way higher. It's great that the capabilities of recording are so cheap now, but the market for disposable entry level gear is a bit misleading in that the pro gear is still really expensive like it has always been. It's a small market and the R&D costs are fixed, so even though the components aren't that much more expensive, the manufacturing process and the engineering to test and ensure the product doesn't fail and is roadworthy are added expenses that the makers of cheap poo poo sold to millions don't have to deal with.
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Jul 29, 2011 19:51
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SuperMxyz posted:I've never trusted usb or firewire. I was recording 18 inputs at 44.1/24bit via firewire onto a P3 1ghz with 512mb ram laptop back in 2002 and had no issues. What exactly leads you to "not trust" firewire?
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Aug 1, 2011 19:00
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Well if you're using a laptop then your PCI solution wont work, so it doesn't matter what APPLE puts on their mac laptop. As for thunderbolt, it's an expansion bus like PCI and PCMCIA. It's not replacing firewire or usb, it's augmenting it. As it starts getting more use you'll see breakout boxes with firewire and usb built into them in a few months, they've already been announced by third parties.
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Aug 1, 2011 20:13
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What does the waveform look like in ableton live?
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Aug 7, 2011 01:48
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If the master and channel volumes are at 0 (default) then there is nothing wrong with that waveform, except it may be clipping in a few spots. I would turn the preamp gain down 3dB or so. If that is too quiet then you need to turn up your headphones/speakers. The only other issue might be one of routing your outputs correctly, or maybe there are plugins on the channel or master that are somehow making things quiet. If you master meters are green and at full level (as they should be with that waveform) then you're looking at an issue with your monitoring, not with your recording levels. On the matter of how loud to record, that waveform is just a bit past what I shoot for. I try to make the waveforms as big as possible so that I have tons of headroom when mixing, and so that I'm minimizing any noisefloor in the converters (which is not as big of a concern unless your preamps and converters aren't particularly good, but I don't leave anything to chance since it wont hurt to record a bigger waveform). Doing this has the added bonus of forcing me to bring all of my faders (in Ableton you use select all + drag fader, in protools you select the "all channels" fader group) down to -9dB or more, giving me a ton more room to adjust as I'm mixing. edit 2: VVVVVVVVVV this is bad advice, don't do this please. At the very least it's out of date as radio is less and less relevant every day so the loudness war doesn't matter nearly as much as it once did. RivensBitch fucked around with this message at 23:41 on Aug 8, 2011 |
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Aug 8, 2011 23:17
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Transistor Rhythm posted:..than a "bigger" waveform that's overdriven the ADC on the way in in the first place... If the over LEDs on my Rosetta 800 are flashing, then I'm distorting the waveform visibly in the DAW. Getting a full, large waveform in the DAW with peaks close to but not reaching 0, without any clipping on the converter, is easy to accomplish. So while I appreciate that you're using hard numbers here, my experience contradicts what you're saying. I get the argument that the size of the waveform itself is dependent on how the daw is displaying it, but most DAWs are set default so that clipping is indicated when the waveform is touching the vertical boundaries of the track. So using that as a reference isn't a bad idea. I also get the argument that with 24bits we're arguably way past the dynamic range of the analog noise floor, and that there's plenty of wiggle room. I would still argue that there is no downside in maximizing your dynamic range by getting as strong of a signal as possible without clipping, and I prefer to err on the side of more dynamic range, not less. If you were to argue that a lot of people in this thread might not have an ADC that is separate from their preamp/interface and that they can't meter the actual ADC stage to watch for clipping, I might concede that point, except that in practice I've never hit the peak/over meters on my converter when I use the size of the waveform as a reference. If you'd like to explain to me why I'm wrong, I'd love to hear it, but please don't try to tell me that I'm overdriving something when I'm not (I've recorded a lot of pure sine and square waves in my day, I can tell when something is being recorded correctly). And finally, I can't tell you how many digital mix sessions have come my way (it's in the hundreds) where all of the waveforms were really small and all of the faders were maxed out. It's quite possibly the biggest, most common mistake I see coming from other engineers; it's so prevalent that my most common move when taking over a session is to select all the audio regions and boost by 6-10db, and then reset all faders to -12db. RivensBitch fucked around with this message at 17:18 on Aug 9, 2011 |
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Aug 9, 2011 17:11
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As far as control surfaces are concerned, I'm currently developing a small (8" x 3" x 3/4") control surface that is a 40x2 character LCD screen with 8 channel and 1 master pushbutton encoder knobs, and a few modifier/shift buttons. It's arduino based and will emulate a Mackie Control so it's compatible with all major DAWs. My goal is to make it super compact and affordable.
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Aug 17, 2011 19:13
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I'm currently decoding the Mackie control protocol, in fact I'm building a max for live emulation of an MCU as a functional way of really understanding how the MCU works.  I have a pile of RGB illuminated pushbutton encoders that I'm evaluating to pick the right ones, so each knob will have full RGB color feedback to help you easily see what it is you're doing. Once I have the prototype working and the PCB designed I'll have to do some kind of presale as i've discovered that the RGB encoders are only available when you order direct in quantities of 10,000. Aiming to keep costs as low as possible, but still don't know what the final price will be. A Mackie Control costs $1000 and a behringer BCF2000 costs $175, I'm hoping to be cheaper than the behringer, if not assembled then definitely in kit form. RivensBitch fucked around with this message at 07:14 on Aug 18, 2011 |
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Aug 18, 2011 07:08
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The Electronaut posted:Will you be doing motorized faders? No. Motorized faders have already been done cheaply, check out Behringers BCF-2000. They also take up a bunch of space, they are what they are. There's really no point to reinvent that wheel. My controller will be SMALL, and instead of faders it will have encoders. But unlike the behringer there will still be an LCD to see the channel names and the values of the parameters you are changing (volume, pan, sends, plugin parameters, and a layer of MIDI CCs). Also, the encoders will each have an RGB LED inside of them that illuminates the shaft and the knob. In it's default setting the knobs will indicate fader position (brighter = more gain, or a higher fader position), but I'm also going to experiment with using one color for fader position, and one color for VU metering, with the third color for another parameter. This might seem less useful than a controller with motorized faders, but I think it has plenty of applications for computer based musicians who aren't locked into traditional analog mixing controls. Making the controller arduino based and open source also opens up the opportunity for expanding the uses of this controller, and I'm hoping that will have more appeal to electronic musicians than just trying to make a cheaper version of something that's already readily available. I also think the size is a huge factor, many electronic performers have crowded tables and the last thing they want to do is give up another 12"x12" of space just to make small volume adjustments. RivensBitch fucked around with this message at 22:40 on Aug 19, 2011 |
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Aug 19, 2011 22:37
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(click images for full size)   The colors of the encoders as pictured are meant to signify red meaning the fader position (brighter red = a higher/louder fader), the green encoder is the currently selected channel (intensity still signifying fader position, just in a different color), and the blue master encoder could be blinking at the tempo, or perhaps blue could signify which "page" we're on (volume, pan, sends, MIDI CCs, etc). Other actions could provide feedbacj, like soloing a channel could darken all the encoder knobs except for the soloed channel which would be blue. Colors could also be animated, perhaps green would make a better fader position and then blinking red would indicate a record armed channel. All of the encoders are pushbuttons as well. I'm still developing the interface and how users will interact with it, I like the idea of the non-encoder buttons on the left and right being color coded modifiers. For instance to be in line with the MCU, make them blue, red, yellow, and green for solo, record, mute, and select. Hold down the blue button and press the encoder of the channel you want to solo, hold down the red to record arm, etc. But then the master encoder changes what page you're on, you can use it to flip to pans, sends etc, and then the color coded modifiers would do something else. Use the modifiers when adjusting the master encoder and that's how you reconfigure the controller. And of course using one of the right modifiers would turn the four left modifiers into banking buttons, by single channels or groups of 8. And when holding down a modifier, the LCD would temporarily blank the channels and tell you what that modifier does, so that way you don't have to memorize the whole thing. There's a ton of different ways to do it and I need to build the prototype so I can start using it and see what makes sense, and then make design changes based on that. Laserjet 4P posted:Use mechanical switches (Cherry) for the transport buttons and I will order one. I'm tired of the rubber/tinny clicky ones. I'm torn on this due to the size limitations and keeping costs low, could you give some examples of the kinds of buttons you'd like to see for transport? I could see using the design above and holding down one of the modifiers on the right to turn the modifiers on the left into transport controls, but my current intention is to make the buttons simple tactile switches with caps. I'm open to hearing suggestions though. RivensBitch fucked around with this message at 00:51 on Aug 20, 2011 |
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Aug 20, 2011 00:38
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The shure srh440 smoke the 7506, and I've owned my 7506s for 16 years.
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Sep 5, 2011 23:01
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That's assuming you're using it with line level signals from devices whose levels are reasonably matched, and not with the headphone out of something like an iPhone.
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Sep 20, 2011 07:16
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I'm going to go out on a limb here and say that yes, there will absolutely be an appreciable difference in sound quality between copying the files from the SD card vs re-recording them through the portable recorder's headphone out, through an audio interface into your DAW. Please don't do this unless audio quality doesn't really matter.
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Sep 20, 2011 20:20
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The PG27 and PG42 USB mics from Shure have headphone amps and direct monitoring built into the mic. Also the circuit paths are clean and they have very clean specs compared to the XLR mics in the same price range. The big disadvantage to USB mics is as mentioned above, you're stuck with a single mic and little to no expandability down the road. The instant you want to record with another mic at the same time, you now own a $200 paperweight.
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Oct 2, 2011 06:04
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The advantage of the external drive is mainly to split up bandwidth. If you're using a lot of audio tracks and samples, keep them on the external, and the samples on the internal. If you're just using lots of samples but not a lot of audio tracks, put the samples on the external and keep the audio tracks on the internal.
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Oct 20, 2011 03:24
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himajinga posted:Many of the affordable 8-channel pres have interface qualities themselves (usb or firewire out among other things) so it seems like I'm paying for features I don't need. Any recommendations, or should I just bite the bullet on one that is maybe a little over-featured? The Onyx Blackbird has 8 Onyx pres on it and costs $500, and it can act as a standalone preamp/AD converter if you are really opposed to using it as your firewire interface. It's essentially the Onyx 800R but with a ton more features, and the 800R cost $800 when it was still current product (it's been discontinued FYI) Does the control software in the MOTU 2408MKIII allow you to control it with the Mackie control? I know the other newer interfacers of theirs have that option. If so you've essentially got a digital mixer, since the monitoring is all done on the interface with zero latency.
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Nov 22, 2011 00:15
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himajinga posted:E: Would it maybe be a better idea to just sell the 2408mkIII to fund the Blackbird since they serve essentially the same function? Sure, unless you need the extra inputs
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Nov 22, 2011 18:39
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The "deaf" mix is halfway down the thread in this link
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Nov 30, 2011 18:47
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Hearing damage is a function of volume amplitude and time. Lower levels are still harmful if you listen to them for a long time, while louder levels will cause damage sooner. Take regular breaks and if your session is going to be long, turn it down. If you can't speak comfortably over the playback then you should be careful.
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Dec 23, 2011 20:52
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Acoustic solutions to acoustic problems, it's your best bet.
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Jan 19, 2012 03:48
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Noise Machine posted:Are there books or articles online about beginner's acoustic theory? I'm starting an independent study in engineering this semester, and one of my problems is it looks like I can only track in lovely rooms with lots of walls, not a lot of room, and drop ceilings. Using Rivensbitch answer of "Acoustic solutions to acoustic problems" is there anything I can do to help the situation and get maybe a little bit better sound using some basic fundamentals of Acoustic sciences? I understand that it is literally a science, and some people devote their lives to it, but I would like to be able to walk into a room, listen to a drumset, and say "ok I hear a very boomy overtone in the kick/floor tom, we should hang up blankets/gobos in these specific locations to help it even it out a bit" It actually isn't too tough to find a good room to work in. Here are the basic rules: 1) Avoid parallel walls and right angles. Look for rooms with an angled ceiling, attics and top stories are great for this. 2) Avoid flat surfaces. Rooms that are all glass, drywall, and hardwood are going to have issues. 3) Variety is your friend. Some carpet here. Some drywall there. A window over there. The more different materials you have on each wall/floor/ceiling, the better. If I don't have a studio available, my ideal room to record in is the average large, carpeted and well furnished living room. Ideally with a couch against one wall, a book case full of different sized books, lots of framed photos on the wall, a big flatscreen TV that chops the corner off of the room, and hopefully a ceiling taller than 10 feet. It wont sound amazing like a commercial studio, but it should sound neutral and not have any glaring issues that would keep you from making a great record, provided that your talent sounds amazing.
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Jan 29, 2012 05:16
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wixard posted:even Beta 91s are notorious for sounding overloaded Have you tried the new Beta91A?
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Feb 10, 2012 20:38
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Splinter posted:That's good news. What I read was warning of comb filtering as you move your head horizontally when the speakers are on their side (same thing would happen as you move up and down when mounted vertically). Any opinion on tweeters in or out? I always assumed out.. I'd be more worried about changing the width of the stereo field, does the manual for your studio monitors have vertical and horizontal polar plots to show their coverage patterns/angles?
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Apr 9, 2012 08:58
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yeastiality posted:I'm trying to record some electric guitar at home, and capture the sound "as I hear it". I know this will sound elitist but in my experience it takes a long time to be able to "hear" a room the way a microphone will. It is quite possible that what you are hearing in the room is not actually what it sounds like, because your brain is interpreting more information than what you are recording. You have two ears that are incredibly precise, and as you are playing the instrument, your body is giving you rhythmic and even tonal feedback that your ears and mind use to influence your experience. I was recently recording a choir in a 400 year old spanish mission, and during setup the choir director asked me if I could move the microphones further back in the room so that the bell choir blended more with the vocal choir. For some background, this mission was essentially all flat surfaces, 30 feet wide by 80 feet long with a 40 foot tall angled barn style ceiling. The walls and ceilings were made of brick, cement, and wood. The floor was all clay tile. With the exception of the pews and the catholic icons hanging behind the choir, there was really nothing to absorb sound in the room. When I first started recording this choir years ago, I would record 3 different stereo mic pairs at various distances from the choir. I had a pair very close in front of the choir, a pair about 20 feet away and 20 feet up in the air, and a pair 50 feet away and 20 feet up in the air. I used this setup about 3 times, and after spending a lot of time in post trying to blend the pairs together to get the best sound, I eventually settled on using just the pair 20 feet away and 20 feet up in the air. The close mics didn't blend and had too many individual voices, and the mics 50 feet away were so reverberant that they were muddy and indistinct. They lacked any useable definition, even when used at lower levels and blended with the mics 20 feet away. I've recorded this same choir for 7 years now, and every year they do a christmas concert in this mission. The choir director is new though, and it took a bit of time for me to explain to him why moving too far back into the room was a bad thing. His ears hear the room and he loves all of the reverberation and the big-ness of the sound. Notes carry forever, and sound larger than life. And whats more, when the choir performs his body is moving, he's involved in directing the swells and diminuendos, and his body and mind are deriving joy from the experience, which alters his perception of the sound. This is why I cary an extra set of headphones, so in between takes I can give him feedback. It's not that his or your hearing is wrong. It's just that a microphone distills the sound, and reveals details that our mind glosses over. Even the best treated rooms in the best recording studios are still not immune from this. It's only with a lot of experience that we can start to honestly listen to a sound and a room, and force our minds to not let these other factors smooth out the details that microphones do not forgive. A good rule of thumb when you are still learning these skills is to record dry with a single mic. Try to dampen your rooms reverb as well. The reverb that sounds great to your ears is probably going to cause more harm than good in the recording process, especially in your acoustically untreated apartment. If you've ever heard the term "mud" used to describe a tone or sound, often that can be caused by excessive or poor reverberation. The suggestion above of using headphones is great too. I prefer earbuds as they block outside noises. Have a friend play your guitar, and plug the mic in so you can hear it in your headphones. Move the mic around and see how things sound. This is a great exercise in hearing how the room actually sounds. And above all, spend more time experimenting and listening, and trust that experience more than anything you read online or in a magazine. Write down your experiences when you're finishing a project, keep a journal of what you did, what worked, and what didn't, and go back to those notes months after you're finished so you can review them with fresh ears. RivensBitch fucked around with this message at 23:37 on Jan 2, 2013 |
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Jan 2, 2013 20:27
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Fiendish Dr. Wu posted:What is best for reducing those spikes? Find the instrument that is causing them and turn it down at those spots. Automation can work here. Also look at the waveform and listen to the spikes at those spots, how clipped does it look and how does it sound to your ears? The later is more important than the former. Compare your mix to other music tracks of a similar style, how is the volume level in comparison? Keep in mind that those tracks are professionally mastered, and it takes a really talented mastering artist/engineer to get those last few dB from a mix without killing it, so don't swing for the fences here. Get close and let your listener turn their system up just a little more if they want it louder. Or, save up $75-150 per track and get them mastered. Or cut corners and squash it yourself, and risk your mix sounding harsh and/or flat.
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Feb 20, 2013 02:22
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ashgromnies posted:Downside of the Mackie is that only the top of the line one supports individual hardware channels being represented in your computer; the others only let you record a mix down and sub sends. This is wrong. I own both the blackjack and the blackbird, both give you discrete channels. Where did you get this misinformation?
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Dec 18, 2013 08:40
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sether01 posted:Do the 3.0 drivers work on windows 7? I got a blackjack for my dad 2 years ago for his birthday and the drivers were poo poo and crashed constantly. It took Mackie like 9 months to update them to 2.1 and they still didn't loving work on windows 7. That was a about a year ago. Worst 150$ I spent, since it was useless and wouldn't work for longer then 40 minutes without crashing. They even closed their forums because so many people were calling them out on their lovely drivers. I'm not sure what to tell you, the Blackjack works fine both on my mac and my virtual machine running win7. Are you running 64 bit? I remember having terrible luck with 64 bit windows and audio in general, but that would have been in 2007 before I made the switch.
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Jan 21, 2014 02:58
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ashgromnies posted:Awesome! That was the interpretation I had taken from reading the descriptions on their site. Good to know you get discrete channels, I may have to get one then. Yes if you're referring to the mixers, only the 1640i has 16 channels of output, the rest are just 2 output channels. I know some people like mixing down on an analog console but I've never thought that using a Mackie Onyx mixer was the way to do this.
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Jan 21, 2014 02:59
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DawntoDust posted:Does anyone know whether there is any weird thing I should know about the difference between a matched pair of AKG C451s and a matched pair of Shure KSM141s? Their spec sheets seem pretty comparable, but that isn't real life. Am I making a mistake thinking of using either as drum overheads? TIA. Fun fact: Shure manufactures their mics to such strict standards and tolerances that you can take any pair of KSM141s and they are a matched pair. Shure never even uses the phrase "matched pair". You can even do this with an SM58, or any other mic that they make.
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Jan 21, 2014 03:04
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strangemusic posted:I have to give major love to putting LDCs in front of a kick (NEVER INSIDE!) Pussy. There are plenty of LDCs that you can put right up on that kick beater and have at it. KSM32 comes to mind.
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Dec 18, 2014 01:38
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strangemusic posted:Firstly because it's a bad habit, and secondly because it's not actually fixing the issue. Your mastering engineer will still have no room to work with, it will just be quieter overall. My workflow usually includes sub-master buses for all the instrument groups, and when setting up my session I'll usually place these at -6db each as a starting point. I also start with my channel faders around -9dB, turn up my monitor level and begin my mix there. Lots of headroom to bring up instruments as needed. All of that said, depending on your DAW bringing the master fader down should be functionally the same as bringing all of your channel faders down the same amount. I believe ProTools uses a 64bit bus, meaning you can bring the master fader down something ridiculous (like 40dB) without losing any headroom.
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Aug 30, 2015 02:30
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My vote - mix to 0. It's been our digital gold standard from the beginning. But we should all be required to only work with "black face" ADATs. (Is that racist?) https://www.pgtonstudio.ch/Adat_Manual.pdf
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Sep 22, 2017 18:33
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Your ceiling fan has a motor inside of it, and when you connect it to the circuit the power surges, which you hear in the speakers because they're on the same circuit. It's likely that the sudden surge of power on the circuit to the motor in the fan is drawing enough power that for a brief moment it's as if your speakers are turned off and then back on.
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Oct 6, 2017 05:10
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Hey guys how's it going?
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Jan 4, 2018 10:00
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