Harmony

And this is why I just have to chuckle when people frown on heavy sampling as being "uncreative"

Hmm, that's a pretty good idea for MAGfest-wear.

Epic. You guys are making me want to beat my budget up and get one of the EWQL vocal packages! Oooh, I really like the swells that sharply cut and end with a huge bass drum (e.g. 0:27 and 0:47). They make the song feel like more than just an auditory experience...I'm more engaged as a listener.

Hmm...maybe you didn't catch this part Crazy mind. There is a mini FREE version of Groove Bias that is great. You can take the kick/snare samples that GB comes with and use them directly in your project, in any sample player, or use the FREE Kontakt Player to load up the full free kit. I just tested GB Mini in Kontakt player and it sounds great (there may be demo limitations on using GB in Kontakt Player, but I haven't run into any yet). You're not going to find legally free samples much better than these EDIT: There is a 30 minute limit on using GB in Kontakt Player and you can't save. Nevertheless, you can use the raw sample files that GB comes with however you want.

I approach the mastering process by thinking that if I mix every track perfectly, all I'd have to do at the mastering stage is take out the headroom and gel stuff together with a little compression. Of course it never works out so perfectly, and I end up just like Rozo playing the damage control game by fixing stuff with EQ, and adding harmonics that I think are lacking, and using multiband compressors to tame a wild kick, and re-adjusting the mixing levels, and etc etc etc. As far as mixing goes, unless it's a very specific effect I'm going for, I generally wait until the very end to automate levels. I like to tweak levels often during a mix and having to move or redraw curves everytime I'm paranoid that something is 0.1 dB too loud would get tedious. Conversely, unless it's a very specific effect, I usually set and forget the EQ on each track right up front. I may tweak it again towards the end of the mixing process, maybe automate a fader or something, but that's about it. Effects like delay and reverb always come before levels for me. Since those effects can affect the output level of the track, it's just easier to keep a handle on things by having them in place before I start messing with the levels. Of course none of this is really hard and fast, especially for people like most of us who are still growing as musicians and producers, and who haven't really found their specific style/workflow yet.

Old skool funk acoustic kit. My first choice for that sound these days is Groove Bias, which is part of a great sale next weekend. Check out virt's demo. Sounds similar to the vibe you're looking for, and with the right eq/compression your could really nail the sound of the clip you posted. The full version is only $79-$89, but check their site to find the free mini version which may work for you too. And if that doesn't work, you could just sample the kick and snare from the audio Go find a higher quality version of the song (it's probably only pennies on Amazon or something) and bam! you're in business.

Bay window...neutral walls...big screen...gear,gear,gear. Nice setup Malcos! I'm a little confused though. Is this a single photo, or are 2 photos stitched together? And where is the desk ,or better question, where is the keyboard/mouse?

Happy 10th OCR! Thank you for the friends I've made, the wonderful experiences I've had working with the site, the musical education and inspiration I wouldn't have gotten otherwise, the constant laughs, the sanity you provided throughout grad school, and most importantly, THE REMIXES! Let's keep it rolling for many many years to come

Teeeempting....

Add some stylish holiday socks to that and my Christmas list is done.

Dude, you've got some sweet stuff there. I'd love to read a rundown of what's in the picture.

As far as I know, Kore Player works in recent versions of Fruity Loops, but if you genuinely can't download the 300MB file then you can always go for soundfonts. Of course there are specific latin soundfont kits, but you could also find a small general midi soundfont and use the latin portion of the drumkit (cowbell, bongos, clave, shakers, etc) sf2midi Soundfont site. Lots of good stuff here Darksword's soundfont collection. the Drums Congas soundfont has a few open tone samples Freesound.org is my favorite sample spot. They have TONS of samples of latin percussion, if you don't mind working with individual samples. The more specific your search, the better. instead of "conga" search for "conga slap" or "quinto slap"

Kore Player has an excellent free Latin Kit http://www.native-instruments.com/#/en/products/producer/kore-player/

For you, that was the most important part.The bottom line was that louder is actually not preferred when you consider listening to an entire track or better yet, an entire album. The Pepsi Challenge gives a person two unmarked sips of soda (one is Pepsi the other Coke) and they're then asked which one they prefer. Fact 1: They usually choose the Pepsi. Fact 2: More people buy Coke than Pepsi. Fact 3: Pepsi has a higher sugar content than Coke. Facts 1 and 2 seem contradictory. If in the sipping taste test people usually pick Pepsi as the better tasting drink, why do more people buy Coke? Because of fact 3. If you put more sugar in the drink, one sip of the soda will probably taste better than one sip of something with a similar taste but less sugar. BUT, when you drink the whole can of soda instead of just a sip, the added sugar turns a significant number of people OFF of that soda. That's what happened with the New Coke product which was an attempt to up the sugar content of the original Coke to appeal to people in the short run. It did -- people liked it more than Pepsi. However, that doesn't matter in the long run. People eventually (whether they knew why or not) rejected the soda because it was too sugary. It's like Mitch Hedberg said about Pancakes "You love 'em at first, but by the end you're fuckin' sick of 'em". Again, a couple of bites of the sugary pancakes is great, but too much sugar just isn't appealing to most people. The musical analogy to be drawn is that when listening to a portion of a track, or even the whole track, you may initially think that a louder version sounds better. But on repeated listenings, that artificially added loudness --added sugar -- will turn most people off of the music. That's what happened with that Metallica CD. I haven't heard it, but I'm pretty sure that there were no audibly clipped peaks, in spite of what a clipping analyzer would tell you. That would have been a technically bad mastering job and at that level of the game, that rarely happens. The Metallica CD was an artistically bad mastering job. Like you're attempting to do with your tracks, the master was pushed to the limits of perceived loudness and while the final result may have been "better sounding" on a first listen to many people, when people started playing it on repeat dozens of times a day (you know how you Metallica fans are ) they got sick of it, like pancakes and New Coke.

Bam, nail on the head. I'll add that knowing what "sounds good" is part of the problem. People can listen to an album and know that it "sounds good" but have no idea why, and thus can't translate that to their music making.

Haha, yeah that was my favorite part

Of course I could be wrong and maybe I'm missing the obvious, but it sounds exactly like Yahtzee with a different mic (or maybe different audio production), I don't see anything in the credits that remotely alludes to a different person (the Johnny Cash ref is what you're referring to perhaps?), and if it actually is someone else, imo they didn't make the 'joke' obvious enough for it to be funny.As for the episode itself: lol

I'm a little confused at what you're confused about As you said, it's a gate/expander, compressor, equalizer and a tube emulator (aka saturator or harmonic exciter). Use them where you want them. The first three are standard audio functions. If you've never run across the 4th one before, a tube emulator attempts to reproduce the sound you'd get by running the audio through an analog circuit containing vacuum tubes. These circuits have characteristic ways of altering the frequency spectrum of the audio that is usually thought to give a warmer/fatter/richer sound. I use them on drums to help them cut through the mix, and I use them on vocals as a more musical way of increasing the higher harmonic frequencies without using EQ (which boosts all frequencies in a selected range). Again, it helps me get my vocals to cut through a mix. This particular plugin is called a channel strip because on a traditional analog mixing board, each channel (including buses) usually includes the first 3 functions, and may use some type of tube amplification. The fact that it's called a channel strip shouldn't restrict you in any way about where to use your plugin. http://www.nomadfactory.com/products/bluetubes_v3/blue_analog_trackbox/index.html

I'm an SE user, so I don't have the VC-64, but I wouldn't expect there to be a big difference.However, it all depends on your priorities. Having this in your toolbox gives you yet another choice for dynamics and excitation...something to add a different flavor, give you some inspiration. I highly value having options when it comes to my FX. You might not care as much. Same thing with the graphical EQ. Some people might not care. I definitely agree with you that the visual rep is a nice feature, and I think having one would dramatically change the way I worked with the effect. But I do like the TrackBox's EQ sound, especially on my vocals, and so I may start using it to roll off highs and lows, while I might go back to the Sonitus for precision edits. I dunno, but it's fun to experiment

Nomad Factory's Blue Tubes Analog TrackBox channel strip. $15 (reg $189) XILS lab's 3LE synth. $20 (reg. $200) I downloaded the demo of TrackBox but I've had problems getting it to work in 64 bit. I was finally able to get the authentication software working using updated drivers from iLok, but SONAR x64 still won't properly scan and recognize the plugin. In SONAR x86 it works great. Gives my vocals a lot of warmth, and the EQ section is audibly smoother sounding (for lack of a better description) than the Sonitus EQ I normally use. So, I'll probably still buy it even if it only works in 32 bit. Still, I'd be interested to know if anyone gets it working on their 64 bit setup.

Print it. Saving your eyes is worth killing a tree.

Ah, you're right I didn't read carefully enough. So if that's the case, does that mean there's no channel input gain in FL Studio?

Interesting. Why not just use the built-in fader? The job of your master bus is to add up all of the channel signals being sent into it before sending them to your soundcard. Digital clipping occurs because the number that represents the amplitude of your signal is too large to be represented in whatever audio bit depth you're using (say, 16 bit audio). When you see the clipping indicator on the master bus, it's saying "I don't have enough bits to represent the sum of all of the audio you're sending me".So the question is, if the 16 bit master doesn't have enough bits to represent anything above 0dB, how are my individual channels able to go to +50dB, as in zircon's example, without clipping? The answer is that the math that's done on ALL of your channels (including the master) isn't done in whatever audio bit depth you have set; it's done using the mathematical capabilities of your computer which can use 32 bits (or higher). 32 bits can represent a much greater range of values than your 16 or 24 bit audio. So what the clipping indicator on the master bus is really saying is "Since I'm at 32 bits, I have enough bits to represent your audio, BUT if I send this to the 16 or 24 bit D/A converter in your soundcard, IT'S not going to have enough bits to represent this and will clip." A few things to note: 1) Technically yes, you actually can clip individual channel tracks but you'd have to go high enough that the value couldn't be represented in, say, 32 bits. That's +100's of dB...good luck trying that 2) It's probably still a bad idea to go too far above 0dB on any channel simply because you don't know how your plugins are doing their math. If they get overloaded, they'll clip your audio. 3) This doesn't apply to analog mixers. If your channel is +6dB and your master level is at -10dB, you may have a -4dB peak on the master but it will have already been clipped by the channel

Garret, you're most welcome for the info in the other thread. zircon's point here, in the context of limiters, is that you don't ever need to use a limiter to prevent clipping on individual source tracks - only the master, since the audio isn't sent to your speakers or mixed down before passing through it. I think most people prefer to leave to master level alone, just like you're doing, and zircon's -100 dB master level example was just an extreme case to get his point across.

yes, you can use that as your primary soundcard.