carrierandoperator

My first good P Bass cost $260. It was MIJ. I had it for about 15 years. It made it through many sessions in that time. I play a Custom Shop now which is very nice, but the MIJ paid itself off many times over. I don't regret buying that bass. Definitely check out the used Squier and Mexican basses in your area. Good luck!

A bass usable for demo purposes doesn't need to run you $800. You can grab a Squier Precision Bass for about $300 or a Standard Precision Bass for $500 -- those prices are new on sale approximations. Or look for local deals on used gear. Plenty of pros take these instruments to stage and studio. A real bass guitar will have better timing and tone than pitch-shifted audio. You can use a bass amp sim and go direct. As a guitarist, keyboardist, and bassist, a guitar would be the last instrument I would reach for when recording a bass line; unless I wanted something different. In the end, if it works for you, great. Learning bass is pretty fun if you can already play guitar.

If you open Scripter, click "Open Script in Editor", then browse Logic's own presets and Tutorial Scripts, you will find a good amount of the available syntax and supported functionalities documented by example. Of course, you may have already discovered that…

In the past, I approached this problem by simply using different Logic songs. I would load them in reverse-chronological order, based on the night's set list. At the end of the song, close the song (and the next song would be activated). Depending on the amount of memory you have and the complexity of your song, you may need to divide this into multiple "sets" or buy more memory. I then programmed automation to enable/disable plugs, fire off program changes to hardware synths/processors, etc. Yes, you can go much deeper with this approach in Logic but… Have you considered MainStage for this? I would try that first, as it is performance-oriented. As a direct answer to your question, I don't believe there is a way to automate selection of a Channel Strip Preset. If you are dead set on this, you might consider manually triggering next/previous channel strip preset via MIDI or Key Command. Manually triggering preset changes can result in significant audio dropouts or buffer underruns, which you probably don't want in your show.

But this doesn't happen with all synths on your system, correct? For example, it appears to work fine with EFM1. Logic applies the region delay correctly. Buffer size should not affect this process. My initial suspicion, which is probably wrong, is that the plugin you're using does not report its latency/tail correctly. That said, a file which begins and ends at zero-crossings will not necessarily produce a seamless loop in a musical context (ie once truncated to, say, 4 measures). Meaning, even if you locate the source of your issue, you'll typically still need to tweak/xfade that loop.

you're welcome. you should prefer offline bouncing, unless there's a specific reason to use realtime (such as interfacing with hardware). offline is faster, and less prone to error. a) if the first is missing, then start the bounce earlier in time, so the plugin can finish its reset cycle -- typically, this problem is the result of a plug that doesn't handle the spec quite right (even if it's dealing with its own constraints). c) you could have soloed only a portion of the midi regions, or disabled something in the signal path (e.g. sidechain of an insert)

and yes, it's worked for me

the manual details where you have to place them on your hdd for exs to locate and convert them. sfz is prolly the extension you want to use.

i bet those smileys 8) have something to do with it... ..your prokit is still out of date. fwiw, there are a few people posting similar reports.

if you're doing an offline (as opposed to realtime) bounce: a) no b) you're probably bouncing it to a 24 bit file. if so, then it will make *some* difference to optimize the level at this stage. obviously, do not clip. otherwise, don't bother modifying the signal if it's close to healthy, unless you're really pushing for the last bit. if 16 bit, then do adjust the gain (but why bounce at 16 at all?). what's 'pretty low', anyway? c) it shouldn't. if it does, then you may be disabling a dependency. if you're doing a realtime bounce: a) yes b) as above c) yes

i've been called worse =p but yes, some of them are a bit academic, difficult to learn, and/or under-documented compared to commercial options, but still worth mentioning (IMO). personally, i never delved very deep into csound either - that was a few years ago, i can't remember why i didn't spend more time with it at the time... *shrug* even so, commercial modular environments with hundreds or thousands of pages of documentation still get a lot of criticism that the docs are lacking... it's simply complex stuff. maybe that should be the subject of your next book??

and once you are comfortable with that, then i recommend using a modular system which has a large library (such as reaktor or max/msp -- FOSS options exist as well: notably csound, pd, and supercollider). then you can really dive in deep and learn how synthesizers work. you'll have a ton of sounds, presets, and synths to learn from.

assuming subtractive or sampler is your weapon of choice (because they are easy): start with the basics: generators (oscillators, samples) and nothing more. introduce amp envelopes introduce layering and panning introduce filters introduce simple global effects introduce modulation sources introduce more complex routing and more polyphonic processes add other synthesis types to your arsenal if you spend a day going through that list, then you won't understand it. truth is, this takes years, and not all people 'get it'. so take it slow and be sure you understand each component's role/operation (even in basic synthesizers). it's very rare for a global effect to define an excellent synth sound.

many people think pinning the output equals 'a big sound'... funny how much bigger a track with proper dynamic range sounds. translation: don't compress the life out of your mix. if the 'kick gets distorted in mastering', you're probably over-compressing/over-limiting (by a lot). if you think that's really incorrect, then your monitoring is probably deceiving you, or you don't know your monitors/room well enough (you can't expect to do this on headphones). so... make sure you have good monitoring/room, and take the time to learn how to get a mix to translate to other systems. that brings up another point: if the mastering process alters the sound of your track significantly, then you have things to fix in production (tracking/mixing).

i assume waldorf has a copy of logic for testing. just send them a session file which exhibits the problem, and the steps required to reproduce the problem using that session.