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 ...


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

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.


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

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

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

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

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

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~), ...


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

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

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

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 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 ...


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

From what I've tried with Kontakt files: Comparing two files with a minor parameter change on a Hex editor. The format is proprietary and maybe obfuscated. You can always contact Native Instruments for more details. http://www.native-instruments.com/fr/support/contact-support/tech-support/ btw if you have any answer from them I would love to hear from you ...


2

I like to use frequencies higher than 48kHz for two reasons: It provides more scope for slowing a sample down and retaining some degree of 'interest' (previously inaudible elements become audible) There's the complication of the 'phase' aspect of waves, particularly when analysing spectra or passing through analogue electronics, which potentially is more ...


2

I am no expert, but could your guitar have been tuned to Just temperament, instead of Equal temperament? Correct me if I am wrong, but if one tunes the guitar by the ear, it does not necessarily sound completely in-tune from all chords? One does need to use tuning meters to check nice tuning against equal temperament instruments (like pianos or more ...


2

Like AJHenderson says, it's a bit unclear as to what exact advice you need. However I can give some general advice. It is tedious and a lot of work, especially with your first try. Things will get better after you've created a library (so start small and expand later). Think about what you really need (not what you want), this helps keeping things in a ...


2

(this was meant aas a comment to Marc W's answer, but ended up being to long for a comment...) To further elaborate on the first paragraph of Marc W's answer, capturing frequencies up to 98 KHz can be relevant in some extreme situations where effects processing can create new partials within the human sensitivity threshold. However the most important ...


1

2/ about the audio recorder limitations Audio recording devices are made to record signals which can be heard. Since our ears are limited to 20kHz, we just need a 40kHz sampling (the 20kHz is the half of 40kHz, named Nyquist frequency). If we just sample a signal at 40kHz, we would need sharp filters to avoid recording frequencies above 20kHz aliased to ...


1

For an application such as this, I think that measurement of the pulse width can be somewhat arbitrary as long as you are measuring all pulses in the same way. For instance, pulses of 200ms or longer should be measured using the same methodology as shorter pulses. This could be either to measure the start of the pulse being the first audio sample in the ...


1

I think the answers above whilst informative are over complicated for the OP question. Simple answer. One word: Interpolation. In post production we will record in the highest sample rates possible so we can slow down (time warp, time stretch) whatever you want to call it and retain sample quality without the need for the adc to have to recreate sample ...


1

Drum samples default is one shot. This is unlike, say, 8 seconds of a violins sample, which will use a release-bound envelope. There are a few situations where drums aren't one shot: Practical - Some drum samples are long and have long sustain (say a tom sample of 8 seconds). When you program your drums, your notes will be of a certain length and you may ...


1

Finding an instrumental and using the phase inversion trick is almost always the most reliable way to get usable vocals. Unfortunately if it's a pro-mastered instrumental, the waveforms won't be an exact match so there will be odd phasing quirks and bleed on the resulting a cappella. You could also hunt for Guitar Hero / Rock Band "mogg" files (multichannel ...


1

Not sure about the vocoder on that, maybe autotune, i think i hear autotune on the lower end of that voice shot. Try autotune and then add a vocoder in to see if it lines up. The keys right after the shot are perfectly aligned in terms of sound waves, whatever they did to that chord stab they did to the vocals.


1

Keymap pro. Discontinued but very valuable. Apple bought Redmatica and buried its products (for now). We hope for a resurrection or integration into Logic X ... who knows. See some Kontakt patches made with it.


1

Simply, your understanding is incorrect. MP3 does use a uniform sampling rate, it reduces resolution of the sound in areas where the resolution is unneeded so that it fits a simpler pattern that can be more efficiently compressed, but it decodes to a fixed sample rate. As EMV was kind enough to point out, you may be thinking of the bitrate. The bit rate ...


1

I've been an Acid Pro user for over 10 years. I've tried Ableton, Cubase, Logic Pro, and Reaper. Each have their pros and cons but all of them lack the easy learning curve and integrated features that would allow me to churn out a quality edit or remix in less than 5 minutes. Since it doesn't look like Acid Pro is still being developed and many windows ...


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