Exploring Sound Design Tools: Contact Microphone

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This post belongs to a series where I´m using unconventional microphones to get interesting sounds.
Please have a look at the other posts from the series:

Hydrophone.
Coil Pickup.

I bought a JrF contact microphone a while ago to do some experimenting and see the potential these mics have for sound design. Here is what I've discovered.

As you may know, a contact microphone records sound from vibrating solid materials instead of the air. This gives these microphones some unique and interesting sonic qualities. Since we are not capturing the ambience around the recording, results usually feel isolated, without an acoustic context. This can be a blessing, no need to worry about reverb or background noise but also may result in dull boring sounds. I quickly discovered than experimenting and trying different props, microphone positions and methods of producing the sound is key to achieve interesting results.

On the technical side, contact microphones need to be connected to a high impedance input in order to have a good frequency response. If you want to get into more detail about this and contact microphone usage in general this is the place to go.

Now that you know the deal, here are some of my recordings. You can individually download every sound via freesound.org or download the whole package through this link.

Window Glass

I just attached the microphone to a large window and try different things.

The first three sounds were recorded with just damp hands, I was trying different movements and was surprised with some of the results, although most of it is just regular squeaks. 

As you can hear, something so simple creates a surprising amount of low end some times.

Next, I tried to try using a milk frother applied on the glass. These recordings exemplify very well the possibilities of these microphones. Usually, it would be impossible to avoid the sound of the machine itself but with a contact mic we are getting the sound of the glass reacting to the vibration without any of the motor. 

The first two examples show this. The other two are the result of applying the forther to the cable of the mic itself resulting in some weird and tonal sounds.

 

Metal Oven Tray

Next, I tried to record some impacts on a metal oven tray. No thing too remarkable on this one but I got nice clean metal resonances that are always good to have.

On the first recording, you will hear that the three small impacts sound kind of distorted. This happens when the microphone is loose so it vibrates against the surface of the object you are recording. This can be useful if you want to get a dirty sound.

Bicycle

I thought the the wheel spokes would be interesting to record and the sound was surprisingly heavy.

Despite having roughly the same length, different spokes produced very different metal overtones. 

I can see these being use with some dissonance in a horror soundscape.

Electric razor

This razor doesn't have different speeds but I discovered that I can use my finger to slow down the motor and create some interesting power on and power off.

There is a nice amount bass, this could be use as layers for sci-fi or fantasy, weird machines.

For the third sample below I tried to create some malfunctioning engine sounds.

Electric Toothbrush

This one is quite dull but could be used as a layer for a servo door or robot. Also, it has a weird chewbacca kind of tone.

Drying Rack

Nice metal impacts with a lot of resonance. Again, surprised with the amount of bass here.

As you can hear, some of the sound have that distorted quality coming from the microphone being a little loose.

The ratchet/castle door sound was done by just striking the different metal rods with a wooden spoon. Quite cool.

Printer

Lastly, I tried attaching the mic to my printer. The result is not very interesting but it could be nice as layers for a robot or some mechanical thing.

Conclusions

As you can see, metallic objects are probably the most interesting ones to record as they resonate more but I'm sure there are many other creative things to try with a contact microphone that I will explore in the future. Thanks for reading.

All you need to know about the decibel

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Here is an bird's eye view on the decibel and how understanding it can be useful if you work as a sound designer, sound mixer or even just anywhere in the media industry.

I've included numbered notes that you can open to get more information. So, enter, the decibel:

The Decibel is an odd unit. There are three main reasons for this: 

1: A Logarithmic Unit

Firstly, a decibel is a logarithmic unit1. Our brains don't usually enjoy the concept of logarithmic units since we are used to things like prices, distances or weights, which usually grow linearly in our every day lives. Nevertheless, logarithmic units are very useful when we want to represent a vast array of different of values.

Let's see an example: If we take a value of 10 and we make it 2, 3 or 5 times bigger, we'll see that the resulting value will get huge pretty fast on a logarithmic scale.2

  1. Note that I will use logarithmic units and logarithmic scales interchangeably. ↩

  2. I'm using a logarithm to base 10. Is the easiest to understand since we use the decimal system. ↩

 
How much bigger? Value on a linear scale Value on a logarithmic scale
1 Time 10 10
2 Times 20 100
3 Times 30 1000
4 Times 40 10000
5 Times 50 100000
 
The reason behind this difference is that, while the linear scale is based on multiplication, the logarithmic scale uses exponentiation.3 Here is the same table but with the math behind it, including the generic formula:

  1. And actually, the logarithm is just the inverse operation to exponentiation, that's why sometimes you will see exponential scales or units. They are basically the same as a logarithmic ones. ↩

 
How much bigger? Value on a linear scale Value on a logarithmic scale
1 Time 10 (10*1) 10 (101)
2 Times 20 (10*2) 100 (102)
3 Times 30 (10*3) 1000 (103)
4 Times 40 (10*4) 10000 (104)
5 Times 50 (10*5) 100000 (105)
X Times 10*X 10X
 

As you can see, with just a 5 times increment we get to a value of a hundred thousand. That can be very convenient when we want to visualise and work with values on a set of data ranging from dozens to millions. 

Some units work fine on a linear scale because we usually move within a small range of values. For example, let's imagine we want to measure distances between cities. As you can see, most values are between 3000 and 18000 km, so they fit nicely on an old fashioned linear scale. It's easy to see how the distances compare.

Now, let's imagine we are still measuring distances between cities, but we are an advanced civilization that has founded some cities throughout the galaxy. Let's have a look:

As you can see, the result is not very easy to read. Orion is so far away that all other distances are squashed on the chart. Of course, we could use light years instead of km and that would be much better for the cities on other stars but then we will have super low, hard to use numbers for the earth cities. Another solution would be measure earth cities in kllometres and galaxy cities in light years but then we wouldn't be able to easily compare the values between them. 

The logarithmic scale offers us a solution for this problem since it easily covers several orders of magnitude. Here is the same distance chart, but on a logarithmic scale, I just took the distances in kilometres and calculated their logarithms.

This is much more comfortable to use, we can get a better idea of the relationships between all these distances.

Like the city examples above, some natural phenomena that span through several orders of magnitude, are more comfortably measured with a logarithmic scale. Some examples are pH, earthquakes and... you guessed it, sound loudness. This is the case, because our ears are ready to process both very quiet and very loud sounds.4

  1. It seems like we animals experience much of the world in a logarithmic way. This also includes sound frequency and light brightness. Here is a cool paper about it. ↩

So the take away here is that we use a logarithmic scale for convenience and because it gives us a more accurate model of nature.

2: A Comparative Unit

Great, so we have now an easy to use scale to measure anything from a whisper to a jet engine, we just need to stick our sound level meter out of the window and check the number. Well, is not that simple. When we say something is 65dB, we are not just making a direct measurement, we are always comparing two values. This is the second reason why decibels are odd, let me elaborate:

Decibels are really the ratio between a certain measured value and a reference value. In other words, they are a comparative unit. Just saying 20dB is incomplete in the same way that just saying 20% is incomplete. We need to specify the reference value we are using. 20% percent of what? 20dB respect to what? So, what kind of reference value could we use? This brings me to the third reason:

3: A Versatile Unit

Although most people associate decibels with sound, they can be used to measure ratios of values of any physical property. These properties can be related to audio (like air pressure or voltage) or they may have little or nothing to do with audio (like light or reflectivity on a radar). Decibels are used in all sort of industries, not only audio. Some examples are electronics, video or optics.

OK, with those three properties in mind, let's sum up what a decibel is.

“A decibel is the logarithmically expressed ratio between two physical values ”

Let that sink in and make sure you really get those three core concepts.
Now, let's see how we can use them to measure sound loudness, that's why we were here if I remember correctly.

In space, nobody can hear you scream

14784812262_6f1534b0e2_b.jpg

As much as Star Wars is trying to convince us on the contrary, sound's energy needs a physical medium to travel through. When sound waves disturb such mediums, there is measurable pressure change as the atoms move back and forth. The louder the sound, the more intense this disturbance is.

Since air is the medium through which we usually experience sound, this gives us the most direct and obvious way of measuring loudness: we just need to register how pressure changes on a particular volume of air. Pressure is measured in Pascals, so we are good to go. But wait, if this is the most direct way of measuring loudness couldn't we just say that a pair of speakers are capable of disturbing the air with a pressure of 6.32 Pascals and forget about decibels?

Well, we could, but again, it wouldn't be very convenient. While the mentioned speakers can reach 6.32 Pascals and this seems like a comfortable number to manage, here are some other examples, from quiet to loud:

 
Source Sound Pressure in Pascals (Pa) Sound Pressure (mPa)
Microsoft's Anechoic Chamber 0.0000019 0.0019
Human Threshold of Hearing @ 1 KHz 0.00002 0.02
Quiet Room 0.0002 0.2
Normal Conversation 0.02 20
Speakers @ 1 meter 6.32 6320
Human Threshold of Pain 63.2 63200
Jet Engine @ 1 meter 650 650000
Rifle shot @ 1 meter 7265 7265000
 

Unless you love counting zeros, that doesn't look very convenient, does it? Note how using Pascals is not very confortable with quiet sounds while mPa (a thousandth of a Pascal) doesn't work very well with loud ones. If our goal is to create a system that measures sound loudness, one of the key things we need is that the unit we use can comfortably cover a large range of values. Several orders of magnitude, actually. To me, that sounds like a job for an logarithmic unit.

Moreover, maybe measuring just naked Pascals doesn't seem like a very useful thing to do when our goal is to just get an idea of how loud stuff is. A better way of doing this, could be to compare our measured value to a reference value and get the ratio between the two. This is starting to sound an awful lot like our previous definition of a decibel! We are getting somewhere.

So, what could we use as a reference level to measure the loudness of sound waves on the air? If you have a look at the table above, you'll notice a very good candidate: the human threshold of hearing. If we do this, 0dB would be the very minimal pressure our ears can detect and after that, the numbers would go up in a comfortable scale as we go up in intensity. Even better, if we measure sounds that are below our ear's threshold the resulting number will be negative, indicating not only that the sound would be imperceptible for us but also saying by how much. That's an elegant system right there. I'm starting to dig decibels.

Now, let's look at the previous Pascals table, but adding now the corresponding decibel values:

 
Source Sound Pressure in Pascals dBSPL
Microsoft's Anechoic Chamber 0.0000019 -20.53
Human Threshold of Hearing @ 1 KHz 0.00002 0
Quiet Room 0.0002 20
Normal Conversation 0.02 60
Speakers @ 1 meter 6.32 110
Human Threshold of Pain 63.2 130
Jet Engine @ 1 meter 650 150
Rifle shot @ 1 meter 7265 171
 

That looks like a much easier scale to use. Remember that dBs are used to measure both very quiet things like anechoic chambers and very loud stuff like space rockets. This scale does a better job for the whole range of human audition, it is fine tuned to those microphones we carry around and call ears.

Here is a nice infographic with some more examples so you get an idea of how some daily sources of sound fit in the decibel scale.

Decibel Flavours

Did you notice that on the table above there is a cute subindex after dB that reads SPL? What's up with that? That subindex stands for Sound Pressure Level and is a particular flavour of decibel. Since decibels can be based on any physical property and since they can use any reference value, we can have many different flavours of decibels depending of which measured property and reference value is more convenient to use in each case.

In the case of dBSPL, this type of decibel is telling us two things. Firstly, that the physical property we are using is pressure. Secondly, that our reference value is the threshold of human hearing. This is fine for measuring loudness on sound waves travelling through the air but, is audio information capable of travelling through other mediums?
AcousticSession.jpg

We have learned to transform the frequency and amplitude information contained in sound waves in the air into grooves in a record or streams of electrons in a cable. That's a pretty remarkable feat that deserves its own post but for now let's just consider that we are able to "code" audio information into flows of electrons that we can measure.

Since dBs can be used with any physical property, we can use units from the realm of electronics like watts or volts to measure loudness in a electrical audio signal. In this sense, both pascals and volts give us an idea of how intense a sound signal is, even though they refer to very different physical properties.

So, we need to establish which units and reference values will be useful to use to build new decibel flavours. We also need to label our particular flavour of dB somehow. This is usually done using a subindex (dBSPL) or a suffix (dBu).

Let's have a look at some of the most used decibel flavours:

dB Unit Property Measured (Unit) Reference Value Used on
dBSPL Pressure (Pascals) 2*10-5 Pascals
(Human Threshold of Hearing)		 Acoustics.
dBA, dBB, and dBC Pressure (Pascals) 2*10-5 Pascals
(Human Threshold of Hearing)		 Acoustics when accounting for
human sensitivity
to different frequencies.
dBV Electric potential (Volts) 1 Volt Consumer audio equipment.
dBu Electric potential (Volts) 0.7746 Volts Professional audio equipment.
dBm Electric Power (Watts) 1mW Radio, microwave and
fiber-optical communication networks.

As you can see, we can also use units from the electric realm to measure how loud an audio signal is. We will choose the most convenient unit depending on the context. Ideally, when using decibels, the type should be stated although sometimes it has to be inferred by the context.

If you read dB values on a mixer desk, for example, chances are they will be dBu, since this is the unit usually used in professional audio. When shopping for a pair of speakers or headphones, SPL values are usually given. Finally, when measuring things like an office space or a computer fan you will see dBA, dBB or dBC. These units are virtually the same as dBSPL but they apply different weighting filters that account for how we are more sensitive to certain frequencies than others in order to get a more accurate result.

And that's all folks. I left several things out of this post because I wanted to keep it focused on the basics. The decibel has some more mysteries to unravel but I'll leave that for a future post. In the meantime, here are some bullet points to refresh you on what you've learned:

Takeaways

The decibel:

  • Uses the logarithmic scale which works very well when displaying a wide range of values.

  • Is a comparative unit that always uses the ratio between a measured value and a reference value.

  • Can be used with any physical property, not only sound pressure.

  • Uses handy reference values so the numbers we manage are more meaningful.

  • Comes in many different flavours depending on the property measured and the reference value.

Exploring Sound Design Tools: Reformer Pro

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Kroto's Audio Reformer Pro is a unique tool for sound design. Here is a look at the software from a practical everyday perspective. I will focus on how it can improve your workflow and also on how it can spice things up on the creative side when doing sound design. I encourage you to grab the demo version and follow along.

Technology

Basically, Reformer takes an input (another recording or a live microphone signal) and uses its frequency and dynamics content to trigger samples from a certain library, creating a new hybrid output.

reformer diag.png

In other words, it allows you to "perform" audio libraries in real time like a foley artist performs props.

Versions

Reformer uses a freemium pricing model and comes in two flavours, vanilla and pro. The first one is completely free but only allows you to use certain official paid libraries. These can be purchased on the Krotos Audio Store where you'll find a huge selection of different libraries.

The pro version uses a paid subscription model and offers more advanced features. This is the one I'll be covering on this post. This version allows you to load up to four libraries at the same time and do a real time mix between them (the free version only allows to load one library per plugin). More importantly, it also gives you the power to create your own Reformer libraries using your sounds.

Interface

As you can see on the right hand side, Reformer Pro controls are quite simple and self-explanatory. Nevertheless, here are some features worth mentioning:

Since you can load four libraries at a time, the X/Y Pad on the left hand side of the plugin will allow you to mix and mute them independently.

The Response value (bottom left corner) changes how fast reformer is processing incoming audio. In general, faster responses work better with sudden transients and impacts while slower values will work better with longer sounds. If you notice undesired clicks or pops, this is the first thing you should try to tweak.

The Playback Speed functions as a sort of pitch control allowing you change the character and size of the resulting signal.

Reformer Workflows

As you can see, Reformer offers an imaginative way of manipulating sounds but how can this be helpful in the context of everyday sound design and mix work? Here are some ideas:

  • Sync: To quickly lay down effects in sync with the picture using your voice or some foley props. For example, covering a creature's vocalizations by hand is always very time consuming and on these kind of tasks is probably where Reformer shines the most.

  • Substitute: Imagine you have all the FX laid down for a certain object or character and now you have to change all of them to a different material or style. In this case, you could keep the original audio, since it has the correct timing and use it to drive a reformer library with the proper sounds.

  • Layer: Once you've stablished a first layer for a sound, you can use reformer to add more layers that will be perfectly in sync with no effort.

  • Make the most of a limited set of sounds: Sometimes you find the perfect sound to use for something but you don't have enough iterations to cover everything. You can create a reformer library with these few sounds and, playing with the playback speed, response time and wet/dry controls, get the most of them in terms of different articulations and variations.

Screen Shot 2017-12-10 at 15.44.15.png

Creating your own libraries

Reformer Pro includes an Analysis Tool that allows you to create custom libraries with your own audio content. I won't go into much detail about how to do this since the manual and this video covers the topic perfectly and the whole process is surprisingly fast and easy. I encourage you to try to create your very own library too.

Ideally, you should use several sounds that follow a sonic theme so you can have a cohesive library. At the same time, these sounds need to be varied enough in terms of frequency and volume content so you can cover as many articulations as possible.

From a technical standpoint, make sure your files are high resolution, clean, closed mic'd and normalized.

As an example, I created ghostly, sci-fi monster voice library using sounds I created with Paulstretch (see my tutorial for Paulstretch here).

You can hear below some of the original samples that I used to build the library. As you can hear, I tried to mix different vocalizations and frequencies:

And here is how the built library behaves and sounds when throwing different stuff at it. The first sounds are the result of monster like vocalizations and you can hear how the library responds with different combinations of timbres. The last sound on the clip is interesting because is the result of the library responding to a ratchet or clicking sound. As you can see is always worth trying to throw weird stuff at reformer to see how it responds.

You can find this library ready to use for reformer in the link below and give it a go:
Ghostly Monster Reformer Library.

Reformer as a creative tool for sound design

In my view, Reformer is not specifically designed for creative sound design as it lacks depth in terms of how well you can manipulate and control the final results. I miss having some control on how the algorithm creates the output signal in a similar way Zynpatiq's Morph plugin has it. But again, I understand sonic exploration is not the main aim of Reformer. Having said that, you can still achieve interesting designs mixing together elements from different kinds of sounds.

For example, we can use a recording with some interesting transients, like a rattling noise to drive some different libraries. Here is the result with a bell:

As you can hear, Reformer takes the volume information and applies it to the bell timbre. And here is a hum and the same rattle creating some sort of fluttering engine or mechanical insect sound. Just for fun, I also added a doppler effect to add movement:

Being able to control any sound with your own source of transients opens a huge window of possibilities. For example, you could use a bicycle wheel as an instrument to perform different movements and articulations. Pretty cool.

I'm just scratching the surface here. There are many more creative ideas that I would want to try. The demo version only runs for 10 days so make sure you can really go for it during those days.

Conclusions

Reformer is a very innovative tool that for sure makes you think in a different way about sound design. Being able to sync and swap sounds on the fly is probably where Reformer shines the most, allowing you to perform recorded libraries live as a foley artist would do. Definitely worth a try.

Shotgun Microphones Usage Indoors

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Note: This is an entry I recovered from the old version of this blog and although is around 5 years old (!), I still think the information can be relevant and interesting. So here is the original post with some grammar and punctuation fixes. Enter 2012 me:

So I have been researching an idea that I have been hearing for a while:

"It’s not a good idea to use a shotgun microphone indoors."

Shotgun microphones

The main goal of these devices is to enhance the on axis signals and attenuate the sound coming form the sides. In other words, make the microphone as directional as possible in order to avoid unwanted noise and ambience.

To achieve this, the system cancels unwanted side audio by delaying it. The operating principle is based on phase cancellation. At first, the system had a series of tubes with different sizes that allowed the on axis signals to arrive early but forces the off-axis signals to arrive delayed. This design, created by the prolific Harry Olson eventually evolved in the modern shotgun microphone design.

Indirect signals arrive delayed. Sketch by http://randycoppinger.com/

In Olson’s original design, in order to improve directivity you had to add more and more tubes, making the microphone too big and heavy to be practical. To solve this, the design evolved into a single tube with several slots that behaved in an equivalent manner to the old additional tubes. These slots made the off-axis sound waves hit the diaphragm later, so when they were combined with the direct sound signal, a noise cancellation occurred, boosting the on-axis signal.

This system has its limitations. The tube needs to be long if we want to cancel low enough frequencies. For example, a typical 30 cm (12″) microphone would start behaving like a cardioid (with a rear lobe) under 1,413 Hz. If we want to go lower, the microphone would need to become too big and heavy. Like this little fellow:

Electro Voice 643, a 2 meters beast that kept it directionality as low as 700 Hz. Call for a free home demostration!

On the other hand, making the microphone longer makes the on-axis angle narrower, so the more directive the microphone is, the more important is a correct axis alignment. The phase cancelation principle also brings consequences like comb filtering and undesirable coloration when we go off axis. This can work against us when is hard to keep the microphone in place, hence this is why these microphones are usually operated by hand or on cranes or boom poles.

In this Sennheiser 416 simplified polar pattern, we can appreciate the directional high frequencies (in red) curling on the sides. The mid frequencies (in blue) show a behaviour somewhere between the highs and a typical cardioid pattern (pictured in green) with a rear lobe.

mg19shotgunrotated.jpeg

This other pattern shows an overall shotgun microphone polar pattern. The side irregularities and the rear lobe are a consequence of the interference system.

Indoor usage

The multiple reflections in a reverberant space, specially the early reflections, will alter how the microphones interprets the signals that reach it. Ideally, the microphone, depending of the incidence angle, will determine if the sound is relevant (wanted signal) or just unwanted noise. When both the signal and noise get reflexed by nearby surfaces they enter the microphone in “unnatural” angles (If we consider natural the direct sound trajectory). The noise then is not properly cancelled since it does not get correctly identified as actual noise. Moreover, part of the useful signal will be cancelled, because it is identified as noise.

For that reason, shotgun microphones will work best outdoors or at least in spaces with good acoustic treatment.

Another aspect to have in mind is the rear lobe that these microphones have. Like we saw earlier this lobe captures specially low frequencies so, again, a bad sounding room that reinforces certain low frequencies is something we want to avoid when using a shotgun microphone. When we have a low ceiling, we are sometimes forced to keep the microphone very close to it so the rear lobe and the proximity effect combines and can make the microphone sound nasty. This is not a problem in a professional movie set where you have high ceilings and good acoustics. In fact, shotgun microphones are a popular choice in these places. 

Lastly, the shotgun size can be problematic to handle in small places, specially when we want precision to keep on axis. 

The alternative

So, for indoors, a better option would be a pencil hipercardioid microphone. They are quite smaller and easier to handle in tight spaces and more forgiving in the axis placement. Moreover, they don’t have an interference tube, so we won't get unwanted colorations from the room reflections.

Is worth noting that these microphones still have a rear lobe that will affect even the mid-high frequencies, but not as pronounced.

So hypercardioid pencil microphones are a great choice for indoors recording. When compared to shotguns, we are basically trading off directionality for a better frequency response and a smaller size.

Exploring Sound Design Tools: Paulstretch

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Have you heard this?

That video was, years ago, my introduction to "Paul's Extreme Sound Stretch" or just Paulstretch for short, a tool created by Paul Nasca that allows you to stretch audio to ridiculously cosmic lengths

Some years ago it was fashionable to grab almost anything, from pop music to the simpsons audio snippets, stretch them 800% and upload them to youtube. When the dust settled we were left with an amazing free tool that has been extensively used by musicians and sound designers. Let's see what it can do.

I encourage you to download Paulstretch and follow along:

Windows - (Source)
Mac - (Source)

The stretch engine

The user interface may seem a bit cryptic at first glance but is actually fairly simple to use. Instead of going through every section one by one, I will show how different settings affect your sounds with actual examples. For a more exhaustive view, you can read the official documentation and this tutorial before diving in.

As you can see above, there are four main tabs on the main window: Parameters, Process, Binaural beats and Write to file. I'm just going to focus on the most useful and interesting settings from the first two tabs.

Under Parameters, you can find the most basic tools to stretch your sounds. The screenshot shows the default parameters when you open the software and import some audio. 8x is the default stretch value, that may explain why so many of those youtube videos where using a 800% stretch.

The stretch value lets you set how much you want to stretch your sound. You have three modes here. Stretch and Hyperstretch will make sounds longer. Be careful with Hyperstretch because you can create crazily long files with it. There is also a Shorten mode that does the opposite, makes sounds shorter. If you want to make a sound infinite,  you can freeze the sound in place to create an infinite soundscape with the "freeze" button that is just to the right of the play button.

Below the stretch slider, you can see the window size in samples. This parameter can have quite a profound impact in the final result. Paulstretch breaks up the audio file in multiple slices and this parameter changes the size of those slices, affecting the character of the resulting sound as will hear below.

Let's explore how all these settings will affect different audio samples. First, here is a recording of my voice on the left and the stretched version with default values on the right hand side:

Cool. As you can see on the file name above, 8X is the stretch value while 7.324K is the window size in samples. Notice that the end of the file that Paulstretch created cuts abruptly. This can be fixed using lower values of window size to create a smoother fade out. This is the classic Paulstretch sound: kind of dreamy, clean and with no noticeable artefacts. You will also notice that, although the original is mono, the stretched version feels more open and stereo.

Just for fun, let's see how the Pro Tools and Izotope RX 6 pitch algorithms deal with a 8x time stretch:

This kind of "artefacty" sound is interesting, useful and even beautiful in its own way. But in terms of cleanly stretching a sound without massively changing its timbre, Paulstretch is clearly the way to go.

Let's play now with the window size value and see how this affects the result. Intermediate values seem to be the cleanest, we are just extending the sound in the most neutral way possible. Lower values (under 3K aprox) will have poor frequency resolution, introducing all sorts of artefacts and a flangerish kind of character. A couple of examples of applying low values to the same vocal sample:

Using a different recording we get a whole new assortment of artefacts. Below, you can see the original recoding on the left, the processed version with the default, dreamy settings on the centre and lastly, on the right, a version with a low window value that seems to summon beelzebub himself. Awesome.

On the other hand, Higher values (over 15K aprox) are better at frequency resolution but the the time resolution suffers. This means that, since the chunks are going to be bigger, frequency response is more accurate and faithful to the original sound, but in terms of time, everything is smeared together into a uniform texture with timbres and chracters from different sections of the original sample blending together. So, it doesn't really make sense to use high values with short, homogeneous sounds. Longer and more heterogeneous sounds will yield more interesting results as in this case different frequencies will be mixed together.

You can hear below an example with speech. Again, original on the left, dreamy default values on the centre and high values on the right. You can still understand syllables and words with a lower window value (centre sample) but with a 66K value the slices in this case are 2 seconds long, so different vocal sounds blend together in an unintelligible texture.

Basically, high window values are great for creating smearing textures from heterogeneous audio. Here is another example to help you visualize what the window size does.

On the left, you have a little piece of music with two very different sections: a music box and a drum and bass loop. Each of them is around 3-4 seconds long. If we use a moderate window size (centre sample below) we will hear a music box texture and then a drum texture. The different music notes are blended together but we can still have a sense of the overall harmony. On the third sample (right) we use a window size that yields a slice bigger than 4 seconds, resulting in a blended texture of both the music box and the drums.

Not only can you choose the window size, but also the type of window. Sort of the shape of the slices. Rectangular/Hamming deal better with frequency but they introduce more noise and distortion. Blackman types produce much less noise but they go nuts with the frequency response. See some examples below:

Adding flavour

Jumping now to the Process tab, here we have several very powerful settings to do sound design with.

Harmonics will remove all frequencies form the sample except for a fundamental frequency and a number of harmonics that you can set. You can also change the bandwidth of these harmonics. A lower number of harmonics and a lower bandwidth will yield more tonal results since a fundamental frequency will dominate the sound, while higher values will be closer to the original source having more frequency and nosie content.

See samples below, the first two are the original recording on the left and the stretched version with no harmonic processing on the right. I left the window size kind of low so we have some interesting frequency warping there.  Further below, you can hear several versions with harmonic processing applied and increasingly higher bandwidths. Hear how the first one is almost completely tone and then more and more harmonic and noise content creeps in. I's surprising how different they are from each other.

Definitely very interesting for creating drones and soundscapes, Paulstretch behaves here almost like a synthetizer, it seems like it creates frequencies that were not there before. For example:

Also worth mentioning are the pitch controls. Pitch shift will just tune the pitch as any other pitch shift plugin. Frequency shift creates a dissonant effect by shifting all frequencies by a certain amount. Very cool for scary and horror SFX.

The octave mixer creates copies of your sound and shifts them to certain octaves that you can blend in. Great for calming vibes. See examples below:

 

Lastly, the spread value is supposed to increase the bandwidth of each harmonic which results in a progressive blend of white noise in the signal as you push the setting further. The cool thing about this, is that the white noise will follow the envelope of your sound. This could be used to create ghostly/alien speech. Here are some examples with no spread on the left and spread applied on the right:

And that's it form me! I hope you now have a good idea of what Paulstretch can do. I see a lot of potential to create drones, ghostly horror soundscapes, sci-fi sounds and cool effects for the human voice. Oh, and also just stretching things up to 31 billion years is nice too.

Mini Library

Here is a mini library I've put together with some of the example sounds, some extended versions an a bunch of new ones. It includes creatures, drones, voices and alien winds. Feel free to use them on your projects.