7

Once a signal is digitized it is treated as a number (as you quite rightly point out) and for 16bits the range of numbers are -32768 to +32767. The numbers are created by an analogue to digital converter. The analogue to digital converter (ADC) will have a maximum input range from -X volts to +X volts (i.e. real signals that you could measure inside your ...


7

For me I found that when I started using the DFH superior sounds I felt the "sound" aspect had been more or less taken care of, as the sounds in that library ( which is pretty old at this stage so there may be even better ones now ) are very hard to tell from the "real thing" ... as regards the feel, it depends on what genre you are recording and which ...


6

The only "natural" (i.e. "using only the data in the sample") way to change the pitch of a recorded sample is to change the speed of its playback. So a sample that sounds an octave lower should take twice as long to play back. But you have a tool that somehow manages to play that sample back octave lower, while somehow keeping the playback time the same. ...


5

The easiest way is obviously adding "swing" through the DAW or plugin. I find 1/16 helps my stuff "come to life". Also layering drum parts, shifting some slightly, can have a huge effect. Even a drum part doubled, with one slightly shifted, can make a huge difference. Remember to not be super rigid on your beats. Add closed or open high hats on off ...


5

You can use Cubase or Sonar (and probably other DAW's) to allow the beat to move with the groove. In Cubase, the Quantise function lets you go from being precisely on the beat to some way off :-) In Sonar it is "Groove Quantise" This will at least feel more realistic, but really what you want to do is try to play the beats you want to hear and feel where ...


4

I use Sox and CDP (Composers Desktop Project) for sound design work building my own libraries. These are command line tools that can be run from the windows shell or with another environment. I use Cygwin in conjunction with these tools and create scripts for different tasks that involve multiple repetitive steps. I then use exif tool for writing metadata ...


3

Apple's Logic software can accomplish this. From their manual: Automatic Tempo Matching The following audio files automatically match the project tempo and will follow any tempo changes made in the global Tempo track. Audio recordings made in Logic Pro (7.0 and later) For example, if you record a bass solo at 100 bpm, you can change the project tempo to ...


3

Be nice if you can post a sample or way we can hear this - can I run the Mathematica code easily?. 2000hz is really low for smooth audio, are you sure it's not just extra frequencies created by aliasing through the playback rate. In fact, I bet it's that. edit. I managed to recreate that in Max/msp using an object which reduces the sample rate (degrade~), ...


3

I would check out the glitchmachines and inear display plugins. They are amazing for being both weird and wonderful! Filtering is also a good way. For example the chaos created by Trent Reznor on the end section of his song "The Great Destroyer" was created largely in part with Metasonix gear. I won't get into this crazy gear...you have to check it out ...


3

The reason people use higher frequencies than the Nyquist Theorom suggests (which states the lowest sampling frequency for a given bandwidth that won't cause aliasing) is that during the editing process, the higher sample rate allows for more accurate editing and a better representation of any effects processing applied, even after sample rate conversion. In ...


3

A digitally sampled signal always is a representation of an analog signal. The proper process of going from analog to digital without oversampling involves applying a perfect lowpass filter cutting off at the Nyquist frequency before sampling. Conversely, reproduction involves outputting sampled pulses (rather than stairsteps) and then applying a perfect ...


3

How do I input two mics simultaneously on a Laptop? To be able to input two mics simultaneously on a laptop, you need a two channels audio interface with two mic inputs. Using one interface for both mics will guarantee that both signal are digitized by a common clock, which is a must for your application. There are numerous options for such an interface, ...


3

frequencies above what microphone allows Microphones have a frequency response curve. This is a random example: The exact shape of the diagram is different for each microphone, but they all have one thing in common: the response will be as close to flat as possible in the range we're usually interested in (20 Hz- 20 kHz for full-range audio microphones), ...


3

The most widely used standard for specifying sounds for a MIDI synthesizer is SoundFont, which contains all samples and control information in a single file. You need some SoundFont editing tool, and some software synthesizer that can use sound fonts.


2

There is no direct or absolute correspondence between 'original' amplitude and sample value. The concept of dB itself is one of relative amplitude -- unless you specify a reference value, as in dBa or dBm, all you know about a decibel is that it represents a (logarithmic) ratio. There is no 'zero dB' as such. In asking about original amplitude do you mean ...


2

The more important question here is why are you not doing this is what ever software you are using? But in response to the question at hand... As for tape decks the older Tascam 8 track and 2 track machines (at least the ones I have) do have a speed control but it is in the form of a pitch + or - selector with no units. There is no direct indication of ...


2

If I understand your question right then you are looking for samples to use in a commercial application. There is a big difference between purchasing the samples and creating them yourself. As far as I know if you create any sound using commercial software then the end result is your own. Eg if I use Cubase to record and release a song then Steinberg have ...


2

My 2 cents as Kontakt library deveper: As far as I can see from Evolve demos and videos, seems there's isn't much of "script magic" there. Seems to be a very basic script to control filter, EG parameters and FX triggering. I can guess all samples has been pre-processed and mapped to Kontakt for further mangling using the provided interface. About source ...


2

what i do if i need quick tune my sounds, i add standart kontakt piano, route it to the same midi channel play the note and tune my samples by ear, to match the piano pitch. btw...celemony melodyne can help you a lot in this situation, try the demo.


2

Sample rate or playback rate is just the speed at which your software or hardware reads through the audio buffer and plays back what it finds. Just adjusting the sample rate speeds up or slows down the rate at which the waveform is read, which effectively shortens or lengthens the waveform contained within. Consider a 440 Hz Sine wave. If you play that sine ...


2

In short: when you "crop your samples in a Drum Rack," you aren't editing the samples - you're editing the Drum Rack (or the devices inside it). Yes, this can be confusing at first. Live doesn't copy audio files, it references them In general there are two ways to load an audio file into Live and mess with it. Let's assume you're working with a .wav file: ...


2

I found this out today. Look at the knobs under the sampling window (root, tune start...etc) Turn the "Play mode" to FW-LOOP. Calibrate the "loop start/end" to your liking. i how that gave some insight.


2

It has been a long time since I've used Sony's Acid. From what I recall, Acid was particularly good for loop based music. Given that, Ableton Live is probably the closest thing to Acid, though the user interface is quite different. Traditional DAWs (Logic, Reaper, Cubase, ProTools, etc.) lay out music as a series of parallel horizontal rows, with time ...


2

Josh's recommendation about Apple's Logic is quite good since it is the only in-house pro audio software developed by Apple. Its essentially just an amped up version of GarageBand which comes with all new Macs. Since no one has listed any of the others I will list all of the notable ones by a few categories and will explain the common pros and cons ...


2

You can safely downsample and reduce the bitrate from a high quality recording, provided you use quality software to do it. Of course, the lower sample rates and bit depth will impose limitations on the sound quality, and you will either get aliasing artifacts or anti-aliasing noise, depending on the settings you choose. Below 44.1kHz and 16 bit the ...


2

I think it's ok for your question to be here. Not sure which examples you have looked at in-class, but here is some potential inspiration: Having said that, perhaps it's just a question of ...


2

I have quite a lot of experience with 'found' sounds, and as far as melodic sounds go: try a metal kettle or some saucepans (you might want to hit them with something soft, like a mallet, to avoid the loud 'clang') – a lot of them can produce a melodic note which you can then pitch-shift. I don't know if a glass/plastic bottle filled with water counts, but ...


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