Harmony
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Posts posted by Harmony
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Been meaning to post, but I finished reading it a while ago and I enjoyed it. It's good as an intro. Some of the examples were very easy to understand, I was able to learn a few things, and now I've got yet another place to point people towards who have basic questions about audio and mixing.
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I don't hear a vast difference in the loudness of your track or sixto's. If you're still concerned about getting it louder, as gwilendiel said you need to focus on mixing. If your pre-limited mix is very peaky, the limiter is going to have to do a lot more work than it should. The mid-level sounds will never get as loud as they could have, had you mixed them properly in the first place. There is also the issue of EQ. Certain frequencies, even if they're at the same amplitude, sound louder or quieter to the human ear.
Bottom line, if you're "close" to what you want, then you need to work on other issues to solve the loudness problem. Limiting is not the end-all solution. And it is not your sound card that's limiting you. If you get a new one, get it for the right reasons.
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Yep, agreed. There's your big advantage for going the 64-bit route, but it's potentially full of headaches right now, so I'd heed Yoozer's advice.Like Yoozer said, a 64-bit OS will allow you to access more RAM. A 32-bit OS allows you to access only 2 GB of RAM per process (unless you do the 3 GB switch, which I've heard is unstable sometimes).
well....no, not really. It can be confusing, but I think you're mixing up terminologies. Sonar's 64 bit architecture means that the program can read memory addresses that have (at most) 64 bits (slots for storing binary info). This gives the program the ability to understand numbers like 8,796,093,022,208 (the number of bits in a terabyte) where a 32 bit architecture wouldn't have enough "places" to understand something that large. It is also important for being able to understand very small numbers. Sonar isn't doing the sampling though, your soundcard is. That's where recording bit depth comes in.If your DAW is also 64-bit (like Sonar), it allows more headroom for recording and drastically reduces your chances of clipping. 64-bit will more accurately reproduce audio because it measures the waveforms amplitude many, MANY more times than 32-bit.First, to clarify, bit depth doesn't specify the number of times your amplitude gets measured. That's determined by the sampling frequency (for example 44,100 samples per second). Bit depth determines how accurately each one of those 44,100 samples in a second can be represented. As an illustration, let's say that the amplitude of one of your samples is actually 0.987795739237492 in whatever units. A bit-depth of 4 would represent that as 0.9 (it can essentially only represent 1 decimal place). A 24 bit recording would record that as 0.9877957. As you see, you have the ability to represent sound more accurately. As you said, you get more leeway before clipping with higher bit depth recording. Why? Well let's say you have two adjacent samples of value 0.987795739237492 and 0.987796987564456. Even though the values are actually different, the 4 bit recording would record those as 0.9, 0.9. Get enough of the same amplitude values in a row (even just 2 or 3 can be audible) and you have clipping. The 24 bit recording would give those two values as 0.9877957, 0.9877969. Two distinct values and you're less likely to have an audible artifact.
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I absolutely disagree. While it's only my opinion that could computing is where we're all headed at some point in the near future, it's fact that graphical work for certain types of projects has been done effectively through an internet connection. I used to run fluid dynamics simulations from Maryland on computers in Arizona. Transferring the hundreds of gigs of data back and forth for processing and pretty-pictureizing just wasn't feasible. The people at the University I was working with wrote a visualization and graphical editing software (similar to TecPlot, if anyone is familiar with that) that was virtualized in the same way that we're talking about here. If "could computing" can handle processor intensive graphical computations on large datasets, it can handle replacing heads in Photoshop.Cloud computing can't replace the desktop computer. There are far too many things that can't currently be done feasibly through the internet (video editing, graphics editing (beyond MS Paint level, anyway), gaming, and countless other tasks.What cloud WILL do is open up possibilities for old, cheap technology to be repurposed into useful machines; specifically in mobility. I think you'll see this taking off on netbooks, since they can pretty much can the hard drive beyond a gig or two of slow flash rom, and toss in absolute bare minimum specs without any major performance hits.
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He used a Pentax system. I'm not too familiar with them, but it looked like either a K200D or a K20D body. I noticed a couple of different lenses, but he seemed to favor a fixed 50mm/1.4 lens (not sure who made it). I saw him take the wedding rings shot with that lens.Nice photos, can someone possibly tell me what camera was used to take these and the lens/es?
Oh yeah, I've been looking forward to those. We can't let it fall too far behind! Let me know if you want any help with simple tedious stuff like slicing the footage into individual performances.Oops, no - not yet. I haven't gotten around to going through all the footage. Still working on the photo album, to be honest. So much to dooo! -
I've used the ASIO drivers on my ProFire in Sonar 6 without issue, but I've switched back to my beloved WDM drivers. If the Lambda offers them, give them a shot.
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(double post)
Alright, I found audiodg.exe, but it only runs when I route some of my outputs on the profire to any of the software returns. If I send everything through the onboard mixer, the way it was setup yesterday, audiodg doesn't run at startup.
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Yeah, that's the same link I found when looking up this process. I'll check again when I get home, but that process wasn't in my task manager and I didn't see it in my running or stopped process. It was 2:30am, maybe I overlooked it. I didn't bother looking for the physical executable to see if it actually exists on my pc, but I'll try that when I get home too.
What version of Win 7 are you using (I'm in Ultimate)? If it exists, have you disabled the mobo's integrated audio device?
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You might try increasing the gain on your master track by 4dB or so, so the limiter only has 0.7 dB of headroom to remove instead of 4.7dB. Depending on what's actually causing the pumping, that may help. You could equivalently do this by increasing the levels on all of the tracks feeding into the master by enough such that the final result is an increase in level on the master track by 4dB...but who wants to do that?
And just so we're clear, the "gain" is not the same as the "level". Gain affects the signal level before it gets to the plugins. You can think of it as a global input level for the plugins on that channel. So changing the gain on a channel would change the output of many FX, such as compression or distortion. "Level" affects the output after any FX have been applied, and thus changing the level has no effect on the output of your FX.
Limiting a rock track should be similar to any other track, although a loss of dynamics isn't usually as critical as in an orchestral piece.
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Downloading...in the meantime:
Those settings sound good. I tried to get your track louder before I left for work and I used almost the same values.
Listening....in the meantime:
I also applied a multiband compressor just before the limiter for the purpose of controlling some of the peaks in lows which were triggering the limiter unnecessarily. That allows a little more volume increase with the limiter without squashing things. You can grab something like T-Sledge and put it on a mastering preset to get started with something like that, if you don't have a mb-comp already.
The track sounds good. I can't compare it accurately to the old ones here at work, but sounds louder and less distorted than your original attempt with W1. Nice work
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For underground, I've gotta recommend OCR's own myf, aka Navi the Swami. I've always been a fan, but I caught him at a show last week and was highly impressed, especially with his production.
Also good from the same show was Flex Matthews. Not as interesting production, but still good from a lyrical standpoint.
And for the mainstream people I like for production, I agree with those that have mentioned Mos Def, or more specifically, people who have produced for him like The Neptunes, Kanye, and Hi-Tek (although lately Tek has moved away from the northern style hip hop I think you're after, into the southern club rap that you're probably not after). Also, Talib Kweli's producers on his album Quality were great. Kanye did at least one of those tracks.
EDIT: I like what you've got there Andy. It's got more of a syncopated feel than I'd associate with most hip-hop, but the artists that push boundaries moreso than others, like Lupe Fisco, would thrive on something like that.
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You can use the limiter wherever, but the master track is the usual place for it.
So here's the deal. When you're composing and mixing, conventional wisdom says to adjust the levels of your tracks so that the peak level on your master track doesn't get anywhere near 0dB. That's called leaving headroom: if your peak level doesn't go above -5dB, then you have 5dB of headroom. I think it's absolutely critical to do so when recording live instruments, but it may not be as important when you're working with samples and synths. To each his own. The point is, headroom may help at the mixing stage, but when it comes time to master and distribute, you want NO headroom. It's just wasted space at that point.
According to my meters, you have 5.2dB of headroom on the original track. Good, if this isn't your final output. One primary use of a limiter is to remove the headroom by boosting the signal to, but not above, the "ceiling" value. These types of compressors are often called a "brickwall" limiters for this reason. The track you posted with the limiter applied still has 2.2 dB of headroom, which tells me you most likely set the ceiling at -2.2dB. Bad. Or rather, pointless. It means your track could be 2.2 dB louder -- that's nearly 60% louder -- without introducing any clipping or changing the dynamics at all. To correctly use the limiter for your purposes, first set the ceiling at 0dB (or -0.1dB if you want to be safe). Leave the release time at the default 200ms (increasing it is fine, but don't decrease it). Now, lowering the threshold increases the gain on your track until the peak output level reaches the ceiling value. Lowering it beyond that point applies compression with a huge compression ratio and a fairly fast attack. Since the loudest sound doesn't get any louder as you decrease the threshold, the only thing that can happen is the increase in volume of the quieter sounds. This is what increases the perceived volume. One obvious danger is that your final track will lack dynamics since the quieter sounds are now closer in volume to the louder sounds. That's where the art of proper limiting comes in. On an orchestral mix, you especially don't want to overdo the limiting.
I think one problem you may have is that you're not looking at the waveform. Immediately after opening the wave in Audacity or Sonar I was able to see that you're wasting a lot of audio space. Grab something like this, put it on you master and actually LOOK at the sound you're pumping out. Is it peaking at, above, or below 0dB? Is there a lot of headroom that I could get rid of? Is this waveform too squashed compared to other tracks that I like?
So, always get rid of headroom on the final track. A limiter will do that for you. Proper use of the limiter will also get you increased perceived volume. If you want more volume without completely killing your dynamics, fancier plugins can analyze your entire track, find peaks in terms of their frequency content and apply some optimum settings to eek every ounce of volume out without just squashing everything to hell. You can do some of that manually with a multiband compressor, and while it's a little more tedious than the single limiter approach, the results can be fine tuned a lot more, and therefore are potentially better.
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If you're talking about just pure perceived loudness, grab a simple maximizer and push the threshold way down. It will sound louder. Assuming you've already tried this, you should post examples.
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Yeah, I noticed the unchanged driver version right after I posted. I have 4GB installed, 3.12GB usable (stupid ASUS mobo randomly sucking up the rest).Heh... well, it's still 2.05, but it might be a newer build. Out of curiosity, how much RAM do you have?
I can't find that process/service on my setup. The closest that I find are services AudioSrv and AudioEndpointBuilder, both running at start. Is there something that I'm missing?the root cause was a driver issue stemming from the audiodg.exe process. -
OMG! If the same goes for firewire, maybe that's what has been happening with the ProFire 2626. If I leave the computer on for 30 mins or so without doing anything, the interface goes into a standby mode and I need to power on/power off to get it back. I thought it was just an annoying "feature" of the interface, but maybe it's this. I honestly don't remember whether or not it happened in XP.One irritating thing about Win7 is that, even under the "Performance" power config, it still has selective USB suspension (or something to that effect) enabled. So my Lambda would randomly power off and I'd not be able to power it back on without a disconnect-and-reconnect. Once that's disabled, everything's golden. If you've been having similar issues, I'd suggest checking that out. -
The only problems I've had have been related to trying to get 32 bit plugins to run correctly, but that's more of a 64 bit SONAR issue. My two M-Audio interfaces have worked problem-free. I'm using the Beta drivers on both now, but for the first few days I used the Vista 64 SP2 driver for the ProFire 2626 and it worked fine too.Really? No occasional BSODs or anything? I've been having really intermittent crashes - once a week, in some cases, and I just conclusively narrowed it down to the ProFire 610 beta drivers.Wonder what you're doing that's different than me? How did you trace the cause of the crashes to the interface drivers?
Also, the Win7 drivers for our interfaces are now out of Beta and have been officially released. Maybe the new ones will work better for you.
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M-Audio ProFire 2626 and M-Audio FastTrack Pro working fine under Win 7 64 bit (and XP 32 bit).
Under Win 7, try other non Win 7 drivers, e.g. Vista drivers. They seem to work well with my interfaces.
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very cool idea.
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"unsupported" doesn't necessarily mean "not functional". I've gotten all of my software/hardware/plugins to work under Win 7 after upgrading from XP. I'd go ahead and try it out, since Win 7 makes it incredibly easy to dual-boot it with XP (or Vista). That way, you don't lose you're old solid XP setup while you're trying out the new Win 7 stuff. If you need something to partition your drives in Win XP for the dual boot, I use Partition Wizard
Since Win 7 and Vista are very closely related, keep in mind that trying some of the Vista drivers may work for you. Even though M-Audio has beta drivers for most of their newer hardware, I initially tried the Vista drivers and they worked fine.
When attempting to install something, if the install doesn't finish properly, Win 7 attempts to diagnose the problem and automatically change the compatibility mode of the installation file to work properly. You can also manually change the compatibility mode (to XP mode, for example) in the setup file's properties windows. That has gotten a few items to work.
Also, especially if you're coming straight from XP and are not used to working in Vista, don't forget that sometimes it's necessary to run installations in admin mode by right clicking and choosing "Run as Administrator". This assumes that your user is already an admin. Some older programs, like Cool Edit Pro 2000, required me to do that to get it to run.
Finally, if all else fails, Win 7 Enterprise, Pro and Ultimate offer a full virtualization of Windows XP SP3 for free! I tried this out the other day and it works flawlessly. The full version of WinXP runs as a window and you can install all of your old Windows XP stuff there. Anything installed in the virtualization windows actually shows up in Win 7 (on your start menu/task bar/desktop) and you can run it without thinking about the virtualization behind it all. Biggest issues with that for me were that I couldn't get 64bit mode to work and there's obviously a CPU hit. For me it was about 10%.
Most of the problems I've had from the upgrade have stemmed not from compatibility, but with the change of all of my software from 32 to 64 bit. Some stuff just won't work in 64 bit, and while that sucks, that's progress.
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Great, that would be much appreciated. I can help if needed. Suggestion: Differentiate sites offering free, commercial, or free+commercial material in some way. Maybe differently colored links?So it's been a while since Andy posted this. I think sometime over the next couple weeks, I'm going to do an overhaul of his OP and try to get it a little more up-to-date for everyone. -
Thanks, I love videos like these!
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Kanthos summed it up. I'll add that aside from having a few extra features that may be helpful for really large or complex projects, the bigger more expensive sequencers (SONAR, Cubase, Logic, etc) potentially will give you fewer headaches when trying to work with third-party software and hardware. Does Kontakt (a popular software sampler) work with SONAR? Absolutely. Does it work with Mixcraft? eh...maybe...it should...right?
Everyone's different. I take weeks or months to finish a full song. The length is partially due to the fact that I work with a lot of live instruments, and setting them up to record isn't always a fast process. But I'm also a perfectionist when it comes to my own music, not because I (or anyone else) need to be to get good results, but because for me it's funI you don't mind me asking a personal question, how long does it take for you to compose a single project now that you are a pro with music related programs? What about the "gods" of the composers? Some people crank out the hits it hours or days. There's a music competition over at ThaSauce called the One Hour Compo in which musicians have a single hour to finish a song. The quality some people get in an hour is amazing sometimes. So yeah, it just depends.
No problem, and thanksThanks a lot for the education! I guess musicians DO have great attitude. I wish other forums also have members like you
Can anyone recommend a sound card?
in Music Composition & Production
Posted
http://ocremix.org/forums/showthread.php?t=26031
Chapter 1 gives a good explanation.