Have you ever watched a weather report?
When the program is over, you often see the signature of the speaker(s) displayed on the screen.
It is possible to reproduce this effect in TVPaint Animation, using the Paint > AutoPaint effect.
This is the perfect way to dynamically sign your future animated works of art!
go onto an empty animation layer, click on the Record a stroke button in the AutoPaint effect and then draw your signature on the screen.
* When this is done, click on the Stop button or press the [Escape] key.
You have now recorded the signature in the program memory. A long series of keys is visible in the timeline (see below).
* Delete the current image on the screen (i.e. the signature).
* Select the tool of your choice and adjust its parameters for the future stroke of the signature.
* Stretch your animation layer so that it covers all keys of the timeline.
* Select all its images and then apply the FX stack.
When you play the project you will note that the signature is "drawn" progressively on the screen.
* You may stretch or contract the set of keys right after recording the stroke as indicated above.
* You may create an layer with the number of images desired and select all images before recording the stroke.
* Recording the stroke generates a large number of key points on the timeline each of which stores the position of your stylus at a given moment. When the stroke is recorded, you may modify the position of the stylus on the screen for the key of your choice thanks to the numeric fields X and Y.
* For a given key, the Draw button indicates if the stylus was in contact with the tablet or not. If it is not enabled for a given key, there will be no stroke for this key (e.g. this is the case if you lifted the stylus during recording).
* Recording a stroke with the AutoPaint effect allows for other recordings, too : in particular the data relative to Pressure, Altitude, Azimuth and Finger wheel available on the tablet. It is also possible to choose whether this data will be restored or not when applying the FX stack by checking or unchecking the corresponding boxes.
* The section Stroke mode offers several choices : Normal, Size, Full and Last.
- The Normal mode progressively reproduces your stroke on the screen. This is the one we have been using since the beginning of this lesson.
- The Size modes requires adjustment of the parameter with the same name and will erase the stroke progressively in time.
- The Last mode is a kind of Size mode parameter set to value "1", which only draws one "point" of the stroke at a time.
- In Complete mode the complete stroke is drawn on the screen, independent of the circumstances.
* The last is the Noise parameter: it defines a possible deviation from the original path and is used to obtain less regular curves (see opposite, the TVPaint logo is slightly distorted, as if written with a trembling hand).
The AutoPaint effect is in a way similar to the Re-apply option as the result depends on the drawing tool selected just before application of the FX stack.
Opposite, after recording a simple stroke in the shape of a semi-circular arch, the AutoPaint effect was applied using the TextBrush tool in Letter mode with the rotation parameter connected to the stylus direction.
The text seems to follow the defined path in time, each letter having its own rotation depending on its position on the path.


Thaler.One announces plans to develop blockchain-based real estate marketplace and launch Thaler token backed by property assets.
Zurich, Switzerland – March 15, 2018 – Thaler.One, the blockchain-based investment fund, has announced plans to open the $220 trillion international property market to a wider range of private investors by introducing a decentralized platform, operating with both fiat and cryptocurrencies. Thaler.One is set on reaching the $5 billion of Assets Under Management (AUM) mark within the next three years. According to its founders, Thaler.One will set the new industry standard for next-generation digital transactions in the real estate market: the fund is professionally managed and advised by an international team of renowned real estate and investment banking professionals, setting it apart from numerous 'generic' blockchain-to-realty startups.
Thaler.One's strategic goal is to allow anyone to invest in real estate, historically a less volatile and less risky investment asset class than stocks or gold, a more predictable source of income. By design, the decentralized investment platform cuts out the 'middleman' and allows small investors to buy into real estate outside of their country of residence and manage their assets in a cost-effective and independent manner. The fund acts as an investment vehicle powered by Thaler tokens, blockchain-secured and backed by real estate assets, predominantly located in the European Union and the UK.
Thalers are security tokens issued in compliance with EU regulations and Regulation D in the US. The fund will open its first pre-sale of Thaler tokens in early April of 2018.
Thaler.One is a decentralized investment fund founded by industry veterans with more than 100 years of combined experience in investment banking, real estate, financial services and IT. Will Andrich, the fund's Co-founder and CEO, a professional investment banker, has held leadership positions at Morgan Stanley, LCF Rothschild and Astor Capital.
The Thaler.One Advisory Board includes Michael Lange (Managing Partner at ACG, Former Chairman of the Board at JLL), Mattia Rattaggi (Swiss Crypto Valley Association, former UBS Group executive), Saydam Salaheddin (Head of EMEA Real Estate, Credit Suisse), Stephen Inscoe (Founder of many real estate tech start-ups).


Prepare to be thrown way, way back on this Throwback Thursday. Because when I was on the red carpet at the CMT Music Awards last week, I was reflecting with the country artists about all the great videos we've seen over the past 31 years on CMT.
And we all have our favorites, right? But if you look at these artist favorites, it seems like Garth Brooks, Shania Twain and Reba are the clear winners, if we were picking winners. Which we're not, but still. Here are some thoughts from artists on the red carpet.
Jake Owen: Those early Shania (Twain) videos were pretty hot. But Garth's "Standing Outside the Fire" was so cool.
Lady Antebellum's Charles Kelley: "What Hurts the Most." When I first saw that, I cried like a baby. Also, "Does He Love You" with Reba and Linda Davis.
Justin Moore: That's a tough, tough, tough call. But I have to say "Brotherly Love" by Earl Thomas Conley and Keith Whitley. They were riding a bicycle as little boys, then it turns into their little boys. I guarantee no one else is gonna give you that answer.
Cassadee Pope: "Thunder Rolls" by Garth Brooks because I remember not really getting it when I was a kid, but I still loved all the drama and the rain. I knew something really bad was happening, but I didn't know what it all meant. I just knew it was captivating.
Cole Swindell: Tracy Lawrence had the best. I watched CMT so much, that when a video came on, I knew what it was even before the title and the artist name came up. When "Texas Tornado" came on, I'd call my buddy and go, "It's on" and hang up and that's all we'd have to say. He'd turn on CMT and we'd both get to watch it.
Josh Thompson: Hank Jr. and Waylon (Jennings) had a video called "If You Don't Like Hank Williams," and it's basically them just sitting at a table slamming shots of whiskey down on the table. That one for sure.


So you've got satisfied customers willing to go on the record to talk about how wonderful you are. Great! Now you have some choices to make on how best to communicate that message. You could do a case study, you could add a testimonial quote to your website, or you could create a testimonial video, in which your happy customer will spend a minute or so singing your praises.
Any of these would be a boost to your marketing effort, but the video may be your best option, combined with a case study.
Videos continue to increase in popularity online.
Why? There's something more substantial about watching someone recommend a company than just reading a quote – that in these cynical times a visitor may suspect was made up.
Should You Shoot Your Video Yourself, or Hire a Marketing Company?
360 BC Group has shot dozens of videos, including testimonials, for customers of all types and sizes. We have the equipment to create a professional looking finished product, and know how to work with our client's customers to put them at ease before the camera, and which questions to ask to elicit a rave review.
So if you hire a marketing company, all of this stuff will be handled properly. But if you choose to shoot the video yourself, here are some helpful guidelines to follow.
That means no quick cuts, no Michael Bay explosions, and no complicated concepts. Make sure information is presented in a clear and concise manner.
Unless your narrator is Ryan Reynolds or Scarlett Johansson, most people will tune out of a testimonial when it starts to become an hour-long tribute. There is no reason for your testimonial to run longer than 60-90 seconds.
That means the lighting, the sound, the camera focus and image quality, and anything else that will distract from the message if it is handled incorrectly. Sometimes companies bring a customer in to shoot a testimonial, and find out later that nothing they said sounds clear, or the image is so dim it looks like they're standing in a cave. When that happens, the customer may not be as enthusiastic about a repeat performance.
Most people do not speak in front of a camera on a regular basis. If your customer appears nervous or uncomfortable as he discusses your company, then what you'll get is not a testimonial, but a hostage video. Viewers may call offering to pay the ransom. You can avoid this by not simply pointing the camera at your customer and expecting them to deliver something clear and coherent. Talk to them first. Ask them specific questions – what was the problem facing your company? How did we help? What was the result? By doing so, you'll get a testimonial that is more detailed and valuable than a generic "I really like this company" sentiment. And if they freeze up, remind them of the questions by prompting them from an off-camera position. You can always fix it in editing.


Over the past few days it has snowed. It snowed a little on Monday and again on Tuesday, and while it looked pretty floating in large white balls of fluff, the air was warm enough that it all melted away moments after it hit the ground. Then, on Thursday, after a very cold, dry Wednesday, the weather reports were all over the board, as is usual when there is a potential for "winter weather" in Portland.
And then… at about ten in the morning… tiny, white flakes began falling from the sky.
The snow did not necessarily stick, but was blown around by incredible gusts of wind.
When I decided to leave work at twelve thirty in the early afternoon, snow was piling up, the wind was crazy, and the temperature was around 20 degrees (with a 'feels like' six degrees!). Needless to say, in a location where drivers are inexperienced with the white stuff falling from the sky, a half hour commute turned into nearly three hours, much of that spent waiting for a bus in 'blizzard' conditions. I rewarded myself with a hot chocolate at one of my favorite coffee shops close to home, and on my walk after the fortifying hot beverage, snow-covered the ground.
Snow was predicted for varying amounts of accumulation, and to my delight – there were measurable inches by the end of the day! Which of course meant that today I was probably going to skip the commute to the office and work remotely.
And I did – delighted to sit at home at my desk, and gaze out my single-pane window at the snow-covered city.
After a few hours of sitting, staring, and finishing some work, my legs could not sit still any longer, and I had to get out and run!
I layered up (three wool shirts!), headed out the door and quickly learned that running on lightly compacted, powdery snow is very much like running on dry sand. Challenging and so much fun; I found myself giggling every two minutes as I would slip or slide. And of course, I took pictures!
Where do you like to run when the weather is … compromising?
How do you stay warm? I thought about a scarf today to cover my nose, but decided against it as it could have been uncomfortable.
Keep warm, and have fun getting outside!
< Previous Sunday's Run: Cloudy with a Quartet of Birds – So Very Portland.
I can't believe how much it's still hitting us! I've been too nervous about running in it and have hit the dreaded treadmill at my apartment's community center. Though, I have enjoyed some walking in it and may try to sneak another snowy walk in later today.


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Timbre: Understanding and Manipulating Texture
By Martin Gerber Contributor
Published Aug 21, 2014 in Music 101
Read time: about 21 minutes
What is timbre? How does it work? And, more to the point, how can you use it to make beautiful music?
When we hear the same note played by different acoustic instruments, we can usually make a rough guess at what kind of instrument it is coming from. Our brains do a great job of distinguishing texture in the sounds we hear. We also can tell when sounds are synthesized or electronically processed, even while listening to them through our headphones or speakers. Even the sound of reverb in a cave has a certain texture to it, and that texture would be different if we covered all of the walls of the cave in acoustic foam.
What makes a sustained 440Hz note played on a trumpet sound different than the same pitch played on a violin? The difference comes from the vibrations at higher frequencies, which come into play and give the sound its texture. We call these additional frequencies overtones or harmonics.
Timbre (pronounced 'tamber') refers to the character, color, or texture of a sound. The melody tells us what pitches are to be played, and when, while the choice of instrument, acoustic environment, amplifiers and recording equipment determine the timbre.
One of the main differences between the sounds of instruments is their timbre – it is largely the color of the sound that tells us what instrument we are hearing. Timbre can be explained in terms of the higher frequency parts of sound. The pitch tells us the fundamental (lowest) frequency, and the timbre comes from the higher frequencies, which are called harmonics or overtones. Understanding the frequency dimension of sound gives a deeper understanding of processes like filtering, equalizing, and distortion, as well as how sounds can be synthesized.
The frequency dimension of sound.
Think of how sunlight is split into the colors of a rainbow when it passes through a prism. The colors we see come from how our brain registers the different frequencies (or wavelengths) of light. When we look at the rainbow coming out of a prism, we are seeing the frequency spectrum of light – i.e. the intensity of each frequency component of the light. The intensity of light at each of these frequencies is what determines its overall color.
The timbre of sound is analogous to color. It is determined by the intensity of sound at each frequency. So it does make sense that musicians often speak of the color of music when referring to timbre; it is a useful way to think about it. Just like splitting light with a prism, we can split sound into its frequency spectrum and look at the amplitude (volume) of the sound at each frequency. You can do this by sweeping a narrow bandpass filter (see the "Filters and Equalizers" section) across your sound, and metering the volume at each frequency. However, a faster, more reliable approach uses a device called a spectrum analyzer.
Being aware of the various dimensions of music helps us conceptualize it. It is a little more intuitive to think about dimensions such as volume, pitch, tempo, and groove. We are less accustomed to thinking about the frequency content (beyond pitch) as we don't have very many day-to-day experiences with it. If it helps, remember the prism and think of timbre like color. When you pay attention to the frequency dimension of sound, there is more there to play with.
Pitch, frequency, and texture.
Sound is a wave of compression and expansion in the air. We can generate these waves with instruments and speakers, and we can pick them up with our microphones and ears. In this section, we will draw some waveforms. You can think of these waveforms as existing in space (e.g. in the shape of the standing wave in a vibrating string or the air compression of a sound wave), or in time as the "signal" that we measure with our ears and microphones. Figure 1 shows an example of an arbitrary waveform.
Figure 1: An arbitrary wave. Three cycles (periods) of the wave are shown. The shape of the repeating unit defines the timbre.
Note that the same shape is repeated three times in the waveform in Figure 1. The frequency at which the waveform repeats itself defines the pitch of the sound, while the shape of that waveform defines the timbre.
It turns out that the shape of the waveform comes from the existence of harmonic vibrations or overtones, which are vibrations at exactly twice, three times, four times, etc. the pitch frequency (which is also called the fundamental frequency). Noisy and atonal instruments, like cymbals, have messier harmonics and often a poorly defined pitch, as a result of having continuous frequencies between the harmonics coming into play.
Waves can be added together. This is what is happening when a single speaker ends up sounding like an entire orchestra. The microphones recorded the sum of all the sound waves, and the speaker plays the sum back. I made the waveform in Figure 1 by arbitrarily adding together a few (sinusoidal) harmonics, as shown in Figure 2.
Figure 2: Building Timbre from Sinusoidal Waves
This building up of timbre in sounds by adding together waveforms is called synthesis. The same way we can synthesize waveforms with timbre by adding together harmonic waves, we can also analyze which harmonics make up a sound with timbre.
Noise and inharmonic sound.
Inharmonic sound departs from the ideas in the previous section, and the timbre contains frequencies away from the harmonic series (multiples of the fundamental or pitch frequency). Different sounds are inharmonic to different degrees. A bell or chime still conveys a certain pitch, however, when you sweep a filter across these instruments, you can hear that there are inharmonic frequencies participating in the sound.
Noise has a high degree of inharmonic content. White noise isn't harmonic at all. It has equal amplitude at all frequencies and so there is no harmonic series there to talk about. One of the great things about white noise, is that you can pick out any pitch that you want using a narrow bandpass filter (we'll get more into filters in the 'Filters and Equalizers' section). This is used to create nice sweeping sounds and sci-fi radio-transmission style glitches, when the frequency of the filter is swept across the noise. You can also get some good percussion samples using bandpass filtered noise as your input.
Amplification and Fidelity
When a signal passes through an amplifier, two things are added: noise and distortion. Keep in mind, the noise added by an amplifier won't be simple white noise (as discussed in the previous section). For example, there will often be a hum at 50Hz to 60Hz; this comes from the AC power outlet. The hum coming from the power line usually has it's own harmonics, and the spectrum of noise can be very far from being purely white.
Fidelity refers to the degree to which a signal is unaltered, or not distorted, as it passes through an amplifier. High fidelity (hi-fi) equipment does a good job of amplifying the signal without significant distortion. You can push both vacuum tube and solid-state (semiconductor) amplifiers into overdrive, resulting in a low fidelity (lo-fi) distorted (crunchy, fuzzy, static-sounding, harshly clipping, etc.) sound.
The change in texture that the lo-fi amplification imparts to your sound depends on the nature and quality of the amplification system. That wooly, warm distortion of a good tube amp is wonderful, and many other (sometimes inexpensive) lo-fi amps and effects can often be used to give an unusual character to sound in a recording or performance.
When distortion is introduced into a signal, additional harmonic content is gained. Overtones can be added to a sine wave by compressing or clipping it, and we will talk about the harmonic series of the square wave in the '' section (which is like extreme clipping on a sine wave). Noise adds frequencies in between the harmonics.
When all of this works together, it can sound great; sometimes, attempts at lo-fi approaches can be less pleasant on the ear. Lo-fi or hi-fi. Mildly distorted or overdriven. Use your ear and give the song what it asks for.
Analog to Digital conversion.
Analog to digital conversion is a complex subject. However, we should touch on it here since it can definitely play an audible role in audio signal processing. Analog to digital conversion is the last step in the digital recording chain, before the digitized data is stored to computer memory. If this process is done right—with good quality converters, and the input preamps properly adjusted—the sound can be almost lossless.
This wasn't always the case with early analog to digital converters, and a crummy analog to digital converter can really take away a lot of the hard work you put into finding that perfect timbre in the room.
When we digitize something, we are quantizing it (rounding it to discrete values) in both time and amplitude. Instead of a continuous analog waveform, now all we have is a bunch of digital data points of amplitude vs. time. So how can this actually get played back and sound the same as the analog (pre-digitized) signal? The short, somewhat hand-waving answer is that it is explained by Nyquist's theorem, where if the highest frequency is sampled at least twice each period, everything works out OK.
I can give a qualitative description of Nyquist's theorem here, hopefully enough to convey the ideas of good audio sampling, and the bad kind that leads to aliasing when high frequencies get 'folded' down and recorded as lower frequencies. When a signal is sampled at twice its frequency, it can be fully recovered during digital to analog conversion—if the amplitude resolution is good. This is shown in Figure 3.
Figure 3: A sampled sine wave at the Nyquist frequency (half of the sampling frequency). This particular signal is also synchronized perfectly and in phase with the sampling clock.
In reality, there will always be some shift in the waveform (called phase) between the sampling events and where the waveform hits zero, and Figure 4 includes an arbitrary phase shift to illustrate this.
Figure 4: A sampled sine wave at the Nyquist frequency. This signal is not in phase with the sampling clock.
The dotted green line shows the lowest frequency sine wave that can accommodate the sampled points, meaning that the proper waveform will be reproduced on replay. Notice that for a waveform where we are below the Nyquist frequency, the situation is even better. Again, the lowest frequency solution is the right one (as shown in Figure 5).
Figure 5: Signals with frequencies below the Nyquist frequency are sampled without loss.
When we try to sample a signal that exceeds the Nyquist frequency, as shown in Figure 6, you find that there is a lower frequency sine wave that satisfies the conditions of the digital data points. This means that that frequency will actually be reproduced at the lower frequency. This error in the frequency is called aliasing (or folding, since it folds down to lower frequencies).
Figure 6: When the signal frequency exceeds the Nyquist frequency, the wave is 'aliased' down to a lower frequency, and the signal is not accurately recorded.
If we ensure that the sampling frequency is at least twice our maximum frequency entering the system, then all frequencies in the signal should be properly resolved in the frequency dimension,as long as the signal is strong enough.
There is still the aspect of digitization that we haven't discussed, which is amplitude quantization. In the previous example, a perfect amplitude measurement was implied. In reality, an analog to digital converter rounds this amplitude to the closest binary value. In a 16-bit recorder, there are 216 (aka 65,536) binary values for amplitude. So this gives a pretty accurate representation if you fill the entire bank of binary values!
When you exceed the bit-depth (i.e. the 65,536 binary values) with a signal that is too strong, you get digital clipping. This can be awful sounding, especially in poorly designed analog to digital converters.
The trick lies in getting the preamp in the sweet spot so that you aren't clipping, but you are also filling your converter to get the most out of your resolution. The softer the signal is, the poorer your actual amplitude resolution gets. Compressing preamps, like tube amps, can let you clip the preamp to avoid clipping the analog to digital converter allowing you to really saturate that converter without fear of digital clipping.
Additive Synthesis
We already discussed how waves can be added together. This is the basis for additive synthesis. A square wave can be generated by adding together every odd harmonic (1st, 3rd, 5th, …) with the amplitude reduced by the harmonic number (i.e. 1,13,15,...), or mathematically:
But forget the mathematics, which really just describe the waveforms. You can see in Figure 7 that even with only 5 harmonics, we can get pretty close to a square waveform.
Figure 7: Adding together odd harmonics, each attenuated by their harmonic number (i.e. 1, 1/3, 1/5, etc ...), creates a square wave. The first five harmonics are shown here.
So to make an additive synthesizer, we need to be able to generate a whole bunch of harmonics and have some way of controlling their volume.
Additive synthesis is a very old technique, dating back to the use of pipe organs. Organ pipes do a pretty good job of giving relatively simple (almost sinusoidal) waveforms, and the drawbars determine how much air flows through each pipe giving the organist control over the timbre.
Additive synthesizers can produce really pleasing sounds, especially when the frequency and volume of the harmonics are modulated. Although the additive method gives great control over timbre, it is inherently bulky and expensive. However, with computational power continuing to rise, these limitations have been softened, and there are a lot of great software-based additive synthesizers on the market.
Filters and equalizers.
A filter changes the timbre of a sound by reducing or boosting the volume of each of its harmonics. Equalizers are made up of a variety of filters that can be used to manipulate timbre.
Two important filters are the high pass filter (HPF) and low pass filter (LPF). These filters attenuate everything above (for the LPF) and below (for the HPF) the cutoff frequency. These types of filters have control of resonance and roll-off. Roll-off controls are usually quantified by the degree of attenuation (in decibels) per octave (above/below the cutoff frequency). The resonance of the filter sets the amount of boost given to frequencies just below (for the LPF) and above (for the HPF) the cutoff frequency. Sweeping these types of filters across waveforms can give really nice timbres, and we will get into this more in later sections. High and low pass filters can be combined to give band pass filters that only let a defined band of sound through.
Equalizers allow you to amplify or attenuate the sounds at a variety of frequencies. Different equalizers have different capabilities, and can do a lot to modify timbres as well as making room for other instruments in a mix. You will find a parameter called Q, which is related to resonance and controls the breadth of the filter's effect. Equalizers are handy tools for reducing the amount of noise present in signals recorded from lines or mics.
You can always bring a hard roll-off high pass filter right up to the lowest played note on a track, and you will eliminate nothing from the sound except noise and background racket. Extending the cutoff frequency of your high pass beyond the lowest fundamental begins to introduce a sort of low-fidelity sound (in the sense of an old radio), and plays games with the mind through creating a missing fundamental.
Adding a soft roll-off low pass filter can be used to tame back strong overtones, distortion, or even a certain degree and type of clipping. And, stronger filters can be used to strongly isolate tones. Harder roll-offs on low-pass filters can sound better when dealing with harsh synthesized tones like square waves, and can really create strong variations in the timbre as the cutoff frequency sweeps over the overtones.
By cranking up the resonance, or Q, on a filter, you can quickly hear it's effect. You will find a sci-fi sounding bump in the sound. Try sweeping the cutoff frequency around, and you will hear this resonance picking out harmonics and noise in between.
Unless you are doing something experimental, you'll probably want to pull the resonance or gain at a particular frequency back to a nice subtle spot that makes the instrument fit into the mix. You can use manual automation of the resonance and cut-off frequencies to follow the dynamics of the song and get more experimental in particular spots of a song.
Subtractive Synthesis
An alternative to building up a complex timbre, one harmonic at a time, is to start out with a harmonically complex waveform and then use filters to achieve various timbres. This method is called subtractive since the filters remove harmonics from the signal in order to control timbre.
Figure 8 shows what the output of an ideal (i.e. very hard roll-off) low pass filter as it removes the last five harmonics from a square wave. Notice how the number of humps is equal to the number of harmonics present in the signal. You can imagine how the wave would become more square as you add more harmonics and get more humps.
Figure 8: Low pass filtering a square wave changes its timbre, making the tone more pure. The top waveform shows a square wave that has been filtered down so that only 5 harmonics are left, and each subsequent waveform has one more overtone removed.
Subtractive synthesis requires at minimum a signal generator and a filter. As a result, subtractive synthesizers are much less demanding than additive synthesizers in terms of the hardware required to build one. Subtractive synths are ubiquitous in the electronic music world as a result.
Acoustic sounds have timbres that changes over time. We can emulate this in synthesized sounds using modulation, which means we vary a particular attribute of the sound.
In an additive synth, we can have the powerful ability to modulate both the amplitude (volume) and frequency of each harmonic. Subtle modulation of the frequency of each overtone at different rates and amplitudes can have some really interesting effects.
With subtractive synthesis, we can only modulate the total volume and the attributes of any filters or effects that the input waveform is conditioned by. Even with these constraints, incredible sounds can be made with the use of envelopes, low frequency oscillators, and manual automation.
Envelopes output a control signal that changes over time and can be used to modulate the total amplitude, cutoff frequency, and any other attributes that the envelope is routed to control. Typical envelopes are of the Attack/Decay/Sustain/Release (ADSR) type, as illustrated in Figure 9. This is a powerful, albeit somewhat confusing, system that can take some getting used to because attack, decay and release define times, while sustain defines a level.
Figure 9: An ADSR (Attack, Sustain, Decay, Release) style envelope, which can be used to modulate all sorts of things.
Low Frequency Oscillator
Low frequency oscillators (LFOs) allow another waveform to modulate the attributes of a sound. The low frequency of an LFO means that the attribute being modulated changes very slowly relative to the audio frequency signal. This can create slowly or quickly evolving effects. LFOs can be routed to control parameters like amplitude (tremolo), frequency (vibrato), and filter cut-off (giving a sort of auto-wah effect). You may also find LFOs useful for creating stereo space as well by routing the LFO to the pan of a signal (auto-pan).
Vocoders
Vocoders are a type of modulator. In a typical vocoder, the "carrier signal" is split into a bunch of different bands (the number of bands changes the sound significantly). A "modulator signal" is then used to modulate the amplitude of each of the bands of the carrier wave. As a result, you can play a melody or chords on one instrument as the carrier, and whisper words into a microphone to modulate, and get that vocoder vocal effect.
You can get creative with what you use as your carrier and modulator signals. I have had a lot of fun routing my drum tracks, or silent drum tracks, to the modulator signal of a vocoder. This technique can be used to give a kind of atmosphere around your drum track, or can (by adding another silient drum track) be used to draw a more percussive timbre from a synthesizer waveform. You'll have much better luck with vocoding if you start with waveforms that are harmonically complex (square-waves, sawtooths, etc.).
When dealing with software synthesizers, and digital audio workstations, changes in parameters (e.g filter cutoff) can be programmed into the song. This process is called automation. One way to do this, is to pencil in the variations that you want to have in your song. For example, you could increase a low pass filter cutoff linearly throughout a build in a track to let those higher harmonics progressively leak through and buid energy by just drawing in two points on your automation curve.
A great, and more organic alternative to hard programming in your automations is to live record them. This means using whatever you have (the mod wheel on your synth, external control knobs, a tablet, or a computer mouse) to control your parameters while you record the changes that you make while the song is running through. This takes a little extra time, but it gives a great degree of humanity to music and I always appreciate when I hear that care in music.
As a side note, the idea of manual automation is also really useful in creating stereo space, as well as the harmonic space that we have been focusing on here. You can extend this to manually automating the dry and wet control on your reverb to convey forward and backward motion, and the pan knob to create left to right motion. Try controlling the two of these at the same time – something that could also be done using an XY pad.
Samplers can have a lot of the same filter features as a subtractive synthesizer; however, the input waveform is instead a sample. A sampler takes any recorded chunk of sound, and pitch shifts it across the keyboard so that it can be played like a synth. A little bit of care in the tuning, and some great sounds can come about.
You can get really creative here with sounds around the house, and adding a bit of portomento (amount of time spent sliding between notes) can really create some interesting effects. Some samplers allow for velocity crossfading, which can let you record two or more samples with different timbres so that the harder a key is hit, the more the other timbres ring through. These attributes of a sampler, mixed with amplitude and filter modulation really open up a world of possibilities for both live and recorded music.
Sterile and dynamic timbres.
We aren't used to hearing static tones (that have perfectly constant timbre) in nature. Acoustic sounds have timbre that is always evolving in a complex way over time. Synthesizers can produce sterile sounding tones if the timbre is not modulated, almost like someone speaking in monotone. Sometimes a sterile sound can work out great in a song, especially with some sympathetic effects or acoustic sound treatments, and there is always this expressive freedom to make the decision. Think of it like a spectrum.
When you record an acoustic sound, the timbre is inherently dynamic. To get expressive sounds that evolve in timbre out of a synthesizer can take quite a bit of time and effort, until you get over that learning curve. Once you become comfortable with various means of modulation, you will find that you can get expressive timbres out of synthesizers without too much bother.
With a subtractive synth, it really helps to have an input waveform that already has some timbral evolution. Samplers are great for this, as they have the filters, envelopes, and LFOs of a subtractive synthesizer, but can have really nice dynamic samples as the input waveform. With a static input waveform, the filter cutoff and volume can be modulated (manually, or via envelopes and LFOs) to achieve dynamic timbres.
If you start playing around with manual automation of parameters on your synth, or even do live recordings & performance with synths where you tweak these knobs on the fly, you will find that synthesizers and samplers can be used to achieve expressive sounds that can give your live performance or studio recordings a whole new dimension.
Generating nice dynamic timbres, or preserving those recorded from acoustic instruments is crucial to creating expressive music that engages the listener. These sounds are welcoming and keep drawing us into the music, especially when the modulations keep moving and don't get too loopy throughout the track. If you are wanting to use sterile tones and unchanging loops in your songs for artistic purposes, try weaving some modulations that evolve throughout the track and some instruments with dynamic timbres in with these to keep an overall sense of motion in your track. The spectrum between the two concepts always gives you room to find the balance that suits your vision.
Martin Gerber
Martin Gerber is working towards a PhD in Engineering Physics focused on solar energy research. He also skateboards, produces multi-genre music and plays in the band Stereography (vocals, guitar, and keys).
@martin_gerber
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