dannthr

Good! Yeah, it's a compression format, so it doesn't work quite the same as wav, which is uncompressed. It doesn't add the padding on the sides because it has a more elegant "parsing" system than your typical mp3 encoder when it encodes. But yeah, it performs nicely on computers and consoles and I believe it's natively or easily supported with OpenAL, which the PS3 is programmed using. You can get pretty clean compressions scales at as much as 12:1, which is great! But Flash doesn't support it, and iOS devices do not like it. So, you've got to pick the format right for the job--it's not always going to be the same one.

That's a waste of budget. Your intro segment should be not much longer than the reverb tail if it can be helped. Maybe just a couple bars at most. Here's an example of a long reverby (a reverb longer than one bar) 8 bar loop. [1] [2] |cut| [3] [4] [5] [6] [7] [8] [1] [2] (loop back to cut) In this case, you don't even need some fancy wrapping. Just render it out with the first two bars on the end and cut properly in a wave editor. (Cut on the 0 crossing on the same transient if you can help it) MP3: Yeah, mp3 is a challenge. Mp3 encoders pad the ends of the file with silence because of parsing, but there are ways around it. Avoid it if you can, though, it's a bit of a pain. You shouldn't have problems with your wav files though, so you might be doing something else wrong. Ogg is nice, nice compression rate, loops cleanly, but it kills iOS devices. Decoding it is a CPU hog and support will have to be built into the game.

Zebra 2 is the only synth you need.

Your mouth is probably the most advanced a cheapest device available to you for making pad sounds.

You can probably go lower than you think--dynamic range is what makes something feel big. It's an issue of contrast. If you have a picture with white and black, then the black will look blacker than if the picture had gray and black. Contrast. Quiet stuff surrounding big stuff is what makes the big stuff sound big.

Nice, sounds better. Try using DXF patches on the long sustains, so you can sculpt and shape it--think of the players arching their bows across the strings and the energy of that motion translating to dynamic changes. Let yourself detach the strings slightly between phrases (this means finding the beginning and ending of individual phrases, and treating those like a light arc, with a climax and everything).

Yeah, in my current orchestral templates, I have an instance of Kontakt for each String group (so 5 total), I have an instance for High Brass and Low Brass each, and so on and so forth.

Naw, you don't have to, I was just being lazy. You can use a Rack Instrument and route multiple MIDI channels to it for each articulation. PLAY supports 16 articulations in a single instance--more than enough for my example. EDIT: I put the same setup together at work, here's a screen shot of PLAY as a Multi-Timbral instrument placed on Cubase's Instrument Rack with 5 MIDI tracks routed to 5 of 16 MIDI channels available in PLAY, with all of the articulations on a different MIDI channel. http://www.dannthr.com/temp/ewqlsoplayss_multitimbral.png

Yes. And yes to idiomatic writing. And don't say I never did anything for ya! BAM!

I understand and appreciate your frustration. Please understand that while EWQLSO is a brilliantly engineered library (Dennis Sands' preliminary engineering on that prior to noise reduction post processing is brilliant), the library is very old and sampled with a philosophy that layering and stitching together different patches is what is required to get a really believable performance. This is what we call the Old Skool. Amazing things have been produced and created in the old skool, but it requires a great deal of effort from the programmer to get it right. Believe me when I say, my first gig in games was programming a famous composer's Symphonic Choirs composition. It's work, no one wants to do it, but it can be done. A good solo performance might require as many as 4 or 5 different patches, stitched together in sequence to serve a single performance.

You should be using dxf or xfd patches with your sustains and sculpting them by modulating dynamics using a midi continuous controller.

http://www.classicalguitars.ca/resonances.htm You'll find that an accoustic guitar is a fairly complex bit of engineering. The sounding board to which the strings are attached, resonates nicely at many frequencies (which is part of what makes it an effective musical instrument). The strings themselves will resonate based on their current tuning, and there are 6 (at least), providing a huge variety of fundamental and overtone based resonance opportunities. If you are really interested in your guitar, you'd have to remove the strings to get a more "honest" result to understanding the resonant frequencies of your sound board.

Interesting premise--I suppose this provides a reason to pick one key over another with purely electronic productions. But I pick my key based on the timbre of instrument ranges and let that selection be independent of tempo.

Probably. Put it this way, you have yet been able to record anything well with it. To explain why you're experiencing what you experience would require insight into your entire recording chain. The microphone alone is not the only thing through which passes your signal.

Mastering is simply the process of preparing a track for publishing to a specified media format. You shouldn't be designing a sound with it.

Resonance is on every filter--it's just a narrow boost at the cut off frequency--this is available on every filter because every filter has a cut off (on Band Pass Filters, this resonance boosts the center frequency). Controlling how much resonance there is on the filter is important--exploration and experimentation is important. The vowel sounds Ooooh have dominance in the fundamental, Aaaahs have a fairly balanced spread of overtones, while EEEEEs are bright with extra presence in the upper harmonics. This is why people can tell when you're smiling over the telephone--their voice sounds bright and forward because their vocal resonance is occuring in the front of their mouth, at the teeth, allowing higher overtones to be emphasized.

I've never used the word "Reese" to describe a bass before and I'm not going to say I'm a master of aggressive bass or anything like that, nor is my posting this video a comment on Zircon's recommendation--I made this video immediately after I posted and then fell asleep when it was encoding. So here, BlackPanther, I'm addressing your question in video form, but I do so as a sound designer familiar with synthesis, not as a beat master or some guy telling you how to make a sound. I do so as a way of telling you not how to create a sound, but as a way to watch someone try to create something from scratch, trying things out, sometimes failing, sometimes succeeding. You know those times when people ask a question on a forum and then someone posts a precious LET ME GOOGLE THAT FOR YOU post? Your question feels like that to a sound designer, that's not a bad thing necessarily, it just demonstrates a lack of personal engagement/exploration. Your question is taking the focus, but in earnest, I see a LOT of questions like yours on OCRemix, which only tells me that there are TONS of people who do not SEEM to make the effort to EXPLORE and TRY things out, one tiny step at a time. Music production and sound design REQUIRE personal exploration. With that, here you go:

An LFO modulating a band-pass filter--easy as pie. EDIT: Rendering/Uploading a video of me creating that sound (or what I think you're talking about)--should be up in a couple hours (HD and whatnot, takes a little while).

It's good to check your mix in a digital sum as well as in a natural mono situation (like check your mix by walking outside the studio and listening through the open door). With that said, there's some great music out there that is totally shit in mono.

You are reaping the rewards of being an early adopter. 1st: It sounds like your firewire hub is dropping off to power-save mode or some shit. Why don't you see about that, maybe you can fix it by turning that off. Old vs New: A lot of hardware manufacturers have had to convert their company goals to accomodate the new digital age--some have done this better than others. It's a lot of work and not everyone is up to task. I have a totally useless TASCAM US-428 sitting in my closet. I also have a totally functional M-Audio Audiophile 2496 PCI card--there will be a day soon when this will be useless to me, but as it is, I use it all the time. Internal Audio Card Recommendations: If I were putting together a new build, I would be paying a lot of attention to RME. Solid support, solid company, forward thinking, excellent DSP/Conversions, worth every penny.

If you save a MIDI as a Type 0 MIDI, this happens.

Don't touch Logic right now, wait until NAMM, see if there's any news. Logic hasn't been updated in a long while and considering what's happened to Final Cut Pro, it's hard to say how Apple feels about catering to the Pro marketplace. My two recommendations for DAWs right now: ProTools (Unbeatable for tracking and editing) Cubase (Unbeatable for VST programming/MIDI programming) Here's the thing about studio monitors. You're not going to be able to really get an appreciable experience out of your studio monitoring system with regards to critical frequency evaluation until you have a very controlled listening environment. What you can get is a detailed monitoring system that will help you evaluate timing and offer an alternative perspective (and less fatiguing listening tool). With that said, my recommendation for you, if you're dead set on getting some new speakers, assuming you listen in your bedroom/living room/home and you don't have a properly treated listening environment, is to get a small, 5" driver near-field. You don't need to pump out any serious wattage, listen at low volumes, you don't need a sub, because frankly speaking, you can't control sub frequencies in your tiny bedroom anyway. I have a pair of Equator D5s. Because they're small, and I have a good audio device that remains balanced at low volumes, I can listen at lower levels, keeping reflection energy to a minimum. The D5s are also Coaxial, which helps to prevent phase distortion between the tweeter and the woofer during cross-over, this is important when you're listening in a small space and the distance between the monitors and your head is fairly small. The Equator D5s are also really cheap because they sell a lot of units without going through a retailer--meaning they're available at about 40% of the price they would be available if they were sold through Guitar Center or something like that. There are also other selling points, I won't go into, just keep in mind that there's a chain of equipment/gear that needs to serve you in order to achieve a quality listening environment, the speakers are just the middle link. If this were my dilemma, I would go with a DAW or some virtual instruments.

Yeah, I was just thinking of it as a time/cost/result thing. Definitely keep up with 3D modeling and animation, if that's what you're doing, that stuff is great!

Hmmm, here's my feeling about the request, even though it's probably not what you want to hear, but maybe I can offer an insight since I consider myself someone capable of fulfilling this request. I often come to the Remix Request section of OCR to be inspired--if I feel like I've got writer's block or I'm bored or whatever. It seems fairly consistent that I do not find something that is in my vein or interest, but I like to look anyway. This is something that would be in my vein, that is something I would categorize as my kind of music. With that said, I feel like these two themes (and using the word theme is pretty loose referring to the Death Note cue which has minimal melodic material) are so similar in generic ways, that I feel the work and effort spent on transcribing both cues in order to elegantly combine them in a way that spoke to both of the sources or said something new would be a lot of work for a not very impressive resulting piece. That's my feeling, and that's what I think when I see a post like this. As someone capable of this, it doesn't sound like a fun idea to actually execute. Maybe other people will feel differently. Anyway, I hope that helps.

We have Auria at the studio--we used it on a location multi-track recording session sampling a moving car, worked pretty well! Plus FabFilter and PSP Audioware FX built in?! Yes!