Getting Started With Ableton Live 9

This set of videos, via Ableton, offers a basic introduction to working with Ableton Live 9. 

The series tutorials do not assume that you’ve worked with Live or desktop audio gear. They cover how to set up Live 9, how to record instruments, using MIDI controllers, how to make beats and melodies, exporting a mix and more.

Note: The YouTube embed automatically takes you to the next video in the series. You can also jump from one to another using the Playlist dropdown in the YouTube player.

9 thoughts on “Getting Started With Ableton Live 9

  1. it’s also kinda cool that you can use at most 512 samples in each ‘project’. all these modern dudes use thousands of samples, but maybe you can get better results by working within certain ableton constraints?!?!

    1. Sampling can be done for functions varying in space, time, or any other dimension, and similar results are obtained in two or more dimensions.

      For functions that vary with time, let s(t) be a continuous function (or “signal”) to be sampled, and let sampling be performed by measuring the value of the continuous function every T seconds, which is called the sampling interval. Thus, the sampled function is given by the sequence:

      s(nT), for integer values of n.

      The sampling frequency or sampling rate fs is defined as the number of samples obtained in one second (samples per second), thus fs = 1/T.

      Reconstructing a continuous function from samples is done by interpolation algorithms. The Whittaker–Shannon interpolation formula is mathematically equivalent to an ideal lowpass filter whose input is a sequence of Dirac delta functions that are modulated (multiplied) by the sample values. When the time interval between adjacent samples is a constant (T), the sequence of delta functions is called a Dirac comb. Mathematically, the modulated Dirac comb is equivalent to the product of the comb function with s(t). That purely mathematical function is often loosely referred to as the sampled signal.

      Most sampled signals are not simply stored and reconstructed. But the fidelity of a theoretical reconstruction is a customary measure of the effectiveness of sampling. That fidelity is reduced when s(t) contains frequency components higher than fs/2 Hz, which is known as the Nyquist frequency of the sampler. Therefore s(t) is usually the output of a lowpass filter, functionally known as an “anti-aliasing” filter. Without an anti-aliasing filter, frequencies higher than the Nyquist frequency will influence the samples in a way that is misinterpreted by the interpolation process.

  2. It’s a good start, but when it comes to “video promotion” then Ableton still has a long way to go in my opinion. Take these tutorials; they’re pretty decent and I’m quite sure there will be plenty of newbies who will find these helpful.

    But the thing is; they’re also not /that/ great. Like in one tutorial where its briefly explained how to use sound effects; without going into any detail the video shows how you can drop an effect onto a track, or drop it onto a return track while raising the “Sends” rotary. But by doing so makes it look as if both methods are different yet give you the same results.

    Personally I think Ableton should have used Huston Singletary a lot more in their videos, same goes for Dennis deSantis. Because not only do these guys know what they’re talking about, they can relay all of that in such a way which not only keeps your attention, but also manages to address a very broad scope…

    Example: Huston’s (IMO) awesome ‘Vocoder’ video tutorial not only explains the vocoder itself; but also addresses routing to some extend. While these modern tutorials seem to be more focused on “you select this option to make it work” without bothering to explain why it is so.

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