Hi there!
I’ve written a guest post on asoundeffect.com about audio library management and metadata tips and tricks.
You can have a look here: https://www.asoundeffect.com/sound-effects-collection/
When working with video, an audio pull up or pull down is needed when there´s been a change in the picture´s frame rate and you need to tweak the audio to make sure it stays in sync.
This subject is somehow always surrounded by a layer of mysticism and confusion so this is my attempt of going through the basics and hopefuly get some clarity.
Source: https://bit.ly/2V1CfwX
First, we need to understand some basic digital audio concepts. Feel free to skip this if you have it fresh.
Whenever we are converting an audio signal from analogue to digital, all we are doing is checking where the waveform is at certain “points” in its oscilation. These “points” are usually called samples.
In order to get a faithful signal, we need to sample our waveforms many times. The number of times we do this per second is what determines sampling rate and is measured in Hertzs.
Keep in mind that if our sampling rate is not fast enough, we won´t be able to “capture” the higher frequencies since these would fluctuate faster than we can measure. So how fast do we need to be for accurate results?
The Nyquist-Shannon sampling theorem gives us a very good estimation. It basically says that we need about twice the sampling rate of the highest frequency we want to capture. Since the highest frequency humans can hear is around 20Khz, a sampling rate of 40Khz should suffice. Once we know this, let´s see the most comonly used sampling rates:
| Sampling Rate | Use |
|---|---|
| 8 KhZ | Telephones, Walkie-Talkies |
| 22 Khz | Low quality digital audio |
| 44.1 Khz | CD quality, the music standard. |
| 48 KHz | The standard for professional video. |
| 96 Khz | DVD & Blu-ray audio |
| 192 Khz | DVD & Blu-ray audio. This is usually the highest sampling rate for professional use. |
As you can see, most professional formats use a sampling rate higher than 40 Khz to guarantee that we capture the full frequency spectrum. Something that is important to remember and that will become relevant later on is that a piece of audio is always going to be the same lenght as long as it is played at the same sample rate that it was recorded.
For the sake of completion, I just want to mention audo resolution (or bit depth) briefly. This is the other parameter that we need to take into consideration when converting to digial audio. It measures hoy many bits we use to encode the information of each of our samples. Higher values will give us more dynamic range, since a bigger range of intensity values will be captured. This doesn´t really affect the pull up/down process.
Let´s now jump to the realm of video. There´s a lot to be said on the subject of frame rate but I will just keep it short. This value is simply how many pictures per second are put together to create our film or video. 24 frames per second (or just fps) is the standard for cinema, while TV uses 25 fps in europe (PAL) and 29.97 fps in the US (NTSC).
Keep in mind that these frame rates are different not only on a technical level but also on a stylistic level. 24 fps “feels” cinematic and “premium” while sometimes the higher frame rates used in TV feel “cheap”. This is probably a cultural perception and is definitely changing. Videogames, which many times use high frame rates like 60 fps and beyond, are partially responsible for this taste shift. The amount of motion is also very important, higher fps will be the best at showing fast motions.
But how can these different frame rates affect audio sync? The problem usually starts when a project is filmed at a certain rate and then converted to a different one for distribution. This would happen if, for example, a movie (24 fps) is brought into european TV (25 fps) or an american TV programme (29.97 fps) is brought into India, which uses PAL (25 fps).
Let´s see how this kind of conversion is done.
Some people think that audio can be set to be recorded at a certain frame rate the same way it can be set to be recorded at a certain sampling frequency. This is not true. Audio doesn´t intrinsically have a frame rate value the same way it has a bit depth and sampling rate.
If I give you an audio file and nothing else, you could easily figure out the bit depth and sampling rate but you would have no idea about the frame rate used on the associated video. Now, and here comes the nuanced but important point, any audio recorded at the same time with video will sync with the specific frame rate used when recording that video. They will sync because they were reocrded together. They will sync because what the camera registered as a second of video was also a second of audio in the sound recorder. Of course, machines are not perfect and their clocks may measure a second slightly different and that’s why we connect them via timecode but that’s another story.
This session is set at 24 fps, so each second is divided into 24 frames.
Maybe this confussion comes from the fact that when you create a new session or project in your DAW, you basically set three things: sampling rate, bit depth and frame rate. So it feels like the audio that is going to be inside is going to have those three intrinsic values. But that is not the case with frame rate. In the context of the session, frame rate is only telling your DAW how to divide a second. Into 24 slices? That would be 24 fps. Into 60 slices? That´s 60 fps.
In this manner, when you bring your video into your DAW, the video´s burnt in timecode and your DAW’s timecode will be perfectly in sync but all of this will change nothing about the duration or quality of the audio within the session.
So, in summary, an audio file only has an associated frame rate in the context of the video it was recorded with or to but this is not an intrinsic charactheristic of this audio file and cannot be determined without the corresponding video.
A frame rate change is usually needed when the medium (cinema, TV, digital…) or the region changes. There are two basic ways of doing this. One of them is able to do it without changing the final duration of the film, usually by re-distributing, duplicating or deleting frames to accomodate the new frame rate. I won’t go into details on these methods partly because they are quite complex but mostly because if the lenght of the final picture is not changed, we don´t need to do anything to the audio. It will be in sync anyway.
Think about this for a second. We have changed the frame rate of the video but, as long as the final leght is the same, our audio is still in sync which kind of shows you that audio has no intrinsic frame rate value. Disclaimer: This will be true as long as the audio and film are kept separated. If audio and picture are on the same celluloid and then you start moving frames around, obviously you are going to mess up the audio but in our current digital age we don’t need to worry about this.
The second method is the one that concern us. This is, when the lenght of the picture is actually changed. This happens because this is the easiest way to fix the frame rate difference, specially if it is not very big.
Let´s use the Telecine case as an example. Telecine is the process of transfering a old fashion analogue film into video. This is not always the case but this usually also implies a change in frame rate. As we saw earlier, films are traditionally shot at 24 fps. If we want to broadcast this film in european television, which uses the PAL system at 25 fps, we would need to go from 24 to 25 fps.
The easiest way to do this is just play the original film 4% faster. The pictures will look faster and the movie will finish earlier but the difference would be tolerable. Also, if you can show the same movie in less time in TV that gives you more time for commercials, so win, win.
What are the drawbacks? First, showing the pictures a 4% faster may be tolerable but is not ideal and can be noticeable in quick action sequences. Second and more importantly, now our audio will be out of sync. We can always fix this by also playing the audio a 4% faster (and this would traditionally be the case since audio and picture were embed in the same film) but in this case, the pitch will be increased by 0.68 semitones.
In the digital realm, we can achieve this by simply playing the audio at a different rate that was recorded. This would be the digital equivalent to just cranking the projector faster. Remember before when I said that an audio file will always be the same leght if it is played at the same saple rate as recorded? This is when this becomes relevant. As you can see below, if we play a 48 KHz file at 50 KHz, we would get the same speed up effect that a change from 24 to 25 fps provides.
This would solve our sync problems, but as we were saying, it would increase the final pitch of the audio by about 0.68 semitones.
That increase in pitch may sound small but can be quite noticeable, specially in dialogue musical sections. So how do we solve this? For many years the simple answer was nothing. Just leave it as it is. But nowadays we are able to re-pitch the resulting audio so it matches its original sound or, alternativaly, we can directly change the lenght of the audio file without affecting the pitch. More on tese methods later but first let’s see what happens if, instead of doing a reasonable jump from film to PAL, we need to go from film to NTSC.
If a jump from 24 to 25 is a 4% change, a jump between 24 to 29.976 would be a whooping 24.9%. That´s way too much and it would be very noticeable. Let´s not even think about the audio, everybody would sound as a chipmunk. So how is this accomplished? The method used is what is called a “2:3 pulldown”.
Now, this method is quite involved so I’m not going to explain the whole thing here but let’s see the basics and how it will affect our audio. First let´s start with 30 fps as this was the original frame rate for TV in NTSC. This makes sense because the electrical grid works at 60 Hz in the states. But as people who, for some reason, are happy living this way, things were bound to get messy and after color TV was introduced and for reasons you can see here, the frame had to be dropped by a 1/1000th to 29.976.
A 2:3 pulldown uses the proportion of frames and the interlaced nature of the resulting video to make 4 frames fit into 5. This is because a 24/30 proportion would be equal to a 4/5 proportion. Again, this is complex and goes beyond the scope of this article but if you want more details this video can help.
But wait, we don’t want to end up with 30 frames, we need 29.97 and this is why the first step we do is slow down the film from 24 fps to 23.976. This difference is impossible to detect but crucial to make our calculations work. Once this is done, we can do the actual pulldown which doesn´t change further the lenght of the film, it only re-arranges the frames.
What does this all mean for us, audio people? It means that we only need to worry about that initial change from 24 to 23.976 which would just be a 0.1 % change. That’s small but it will still throw your audio out of sync during the lenght of a movie. So we just need to adjust the speed in the same way we do for the 4% change. If you look again at the picture above, you’ll see that that 0.1% is the change we need to use to go from film to NTSC.
As for the change in pitch, it will be very small but we can still correct it if we need with the methods I show you below. But before that, here is a table for your convenience with all the usual frame changes and the associated audio change that would be needed.
| Frame Rate Change | Audio Speed Change | Pitch Correction (If needed) |
|---|---|---|
| Film to PAL | 4% Up | 4% Down // 96% // -0.71 Semitones |
| Film to NTSC | 0.1% Down | 0.1% Up // 100.1% // + 0.02 Semitones |
| PAL to Film | 4% Down | 4% Up // 104% // +0.68 Semitones |
| PAL to NTSC | 4.1% Down | 4.1% Up //104.1% // +0.68 Semitones |
| NTSC to Film | 0.1% Up | 0.1% Down // 99.9% // -0.02 Semitones |
| NTSC to PAL | 4.1% Up | 4.1% Down // 95.9% // -0.89 Semitones |
There are two basic methods to do a pull up or pull down. The first involves two steps: first changing the duration of the file while affecting its pitch (using a different sample rate as explained before) and secondly applying pitch correction to match the original’s tone. The way to actually do the first step depends on your DAW but in Pro Tools, for example, you’ll see that when importing audio you have the option to apply SRC (Sample Rate Conversion) to the file as pictured above.
The second method is simply doing all at once with a plugin capable of changing the lenght of an audio file without affecting its pitch.
Also, keep in mind that these techniques can be applied to not only the stereo or the surround final mix file but also the whole session itself, which would give you much more flexibility to adjust your mix on this new version. This makes sense because a 4% change in speed could be enough to put two short sounds too close together and/or the feel of the mix could be a bit different. Personally, I have only used this “whole session” technique with shorter material like commercials. Here is a nice blog post that goes into detail about how to accomplish this.
As for changing a mixed file as a whole, wether you use a one step or two steps method, you will probably find that is easy to introduce glitches, clicks and pops in the mix. Sometimes you get dialogue that sounds metallic. Phase is also an issue, since the time/pitch is not always consistent between channels.
The thing is, time/pitch shift is not a easy thing to accomplish. Some plugins offer different algorithms to choose from depending on the type of material you have. These are designed with music in mind, not dialogue, so “Polyphonic” is the one that is usually the best option for whole mixes. Another trick you can use is to bounce your mix into stems: music, dialogue, FX, ambiences, etc and then apply the shift to each of them indepentdently, applying the best plugin and algorithm to each. This can be very time consuming but will probably give you the best results.
As you can see, this whole process is kind of tricky, particularly the pitch shift step and this is why in some occassions the audio is corrected for sync but left at the wrong pitch. Nevertheless, nowadays we have better shifting plugins to do the job. Here are some of the most commonly used, although remember that non of these works perfect in every ocassion:
-Zplane Elastique: This is in my opinion the best plugin and the one I personally use. It produces the least artefacts, keeps phase coherent and works great on whole mixes, even with single step processing.
-Pro Tools Pitch Shift: This is the stock time/pitch plugin that comes with Pro Tools. It is quite fast but is prone to create artifacts.
-Pro Tools X-Form: This one is more advanced (comes blunded with Pro Tools Ultimate) but it still suffers from some issues like giving dialogue a metallic tone or mesing the phase on stereo and surround. Also, it is slow. Veeeery slow.
-Serato Pitch n Time: I haven’t tried this either but I had to mention it since it is very commonly used and people swear by it.
-Izotope Time & Pitch: It can work well sometimes and offers many customizable settings that you can adjust to avoid artefacts.
-Waves Sound Shifter: Haven´t used it but it’s another option that seems to work well for some applications.
Which one should you choose? There is no clear answer, you will need to experiment with some of them to see what works for each project. Here is a good article and video comparing some of them.
I hope you now have somehow a better understanding on this messy subject. It is tricky from both a theoretical and practical level but I believe is worth figuring out where things come from instead of just doing what others do without really knowing why. Here are some takeaways:
Sampling rate and bit depth are intrinsic to an audio file.
At the same time, an audio file can be associated to a certain video frame rate when they are both in sync.
The frame rate change process is different depending on the magnitud of the change.
An audio pull up or pull down is needed when there is a frame rate chenge on the picture that affects its lenght.
The pull up/down can be done in two steps: lenght change first, then pitch correction or ir can be done in a single step.
Time/Pitch Shift is a complicated process that can produce artefacts, metallic timbres and phase issues.
Mixes can be processed by stems or even as whole sessions for more flexibility.
Try different plugins and algorithms to improve results.
Thanks for reading!
Igniter is Krotos’ new engine sound design plugin. They have been kind enough to sent me a license to have a look and see what it can offer. Igniter allows you to virtualize vehicle engines (real, sci-fi or fantastic) combining a granular section, a set of synthetizers and two sample managers. It includes performance controls so you can automate the vehicle RPM, engine load and many FX (including doppler) in order to get a realistic sounding engine. It comes with a big amount of presets including sport and utilitarian cars, planes, helicopters, trucks, motorbikes and sci-fi vehicles.
So here is my in-depth look at the plugin features with some examples here and there. I encourage you to follow along in your own DAW, you can find a full featured demo here.
The interface is clean and easy to read and you can resize the window which is very nice. A main section (left side) occupies most of the screen and includes all the audio sources we can use. These sources are divided into four different tabs: Granular, Synth, One Shot and Loop and also includes a file browser.
On the right hand side we find the engine master on/off switch and the main revs knob in the middle. This revs knob acts as a gas pedal for the whole plugin. At the top, we find the Mod system, where Igniter’s true power resides, since it allows you to dynamically link any parameter within the plugin to the revs knob using envelopes and LFOs. Lastly, at the bottom right side, we find the FX and mixer sections. Let’s see all these in more detail.
If you need more info, most features are well covered in the manual and on Kroto´s videos. What follows is my own take on the plugin capabilities, plus some wish list features that I would love to see in the future.
This is probably the most complex and important generator. It combines granular synthesis with real recordings to re-create a virtual engine with a revolutions or RPM knob that you can “drive”. Each vehicle includes two mic perspectives: engine and exhaust and we can easily mix between both with an slider.
When I saw this, it occurred to me that it would have been nice to also include an interior perspective as this would be very useful for vehicle scenes like chases. After some looking around, I discovered that all vehicles have a “In-car” preset which solves the problem. But this is not a true recording of the car interior, but a recreation of it using EQ and convolution reverb. Would this be too different or unauthentic compared to a true inside recording? To be honest, I don’t know since I don’t have a huge amount of experience doing car sound design but I suspect these presets will suffice pretty well for most applications and, of course, you can always tweak them to suit your needs or even do your own "in-car” processing outside of Igniter.
In terms of how the granular engine actually works, we can’t see what’s going on under the hood (see what I did there?) but I suppose the plugin is using recordings at different steady RPMs and blending them together as you act on the engine. This is similar to the approach used in middleware like Fmod for use in video games. The result is pretty natural and smooth and driving the RPM feels responsive and clean.
For now, you can’t add your own sounds to the granular section, as they would probably need to be edited in a very specific way for them to work here. On release day, Krotos offers 13 different vehicles that use this granular option but I’m sure more will be added with time or maybe made available for individual purchase in the future.
As you can see on the interface above, there are basically two ways to control the engine simulation: manual and auto. At the same time, every car comes with a set of three presets, two of them are manual and a third uses auto. So, using the example pictured on the right hand side with the Dacia 1310:
-Dacia 1310: “Free” mode that uses manual driving.
-Dacia 1310 Manual Gears: Uses manual driving but with pre-determined gear shifts on the Revs progression.
-Dacia 1310 Auto Gearbox: Uses the auto mode.
Let’s see what’s the difference between these three:
In general, manual mode allows you to freely change the engine’s RPM an also gives you a “load” knob. This parameter simulates if you are putting pressure on the engine or, in other words, if you are applying pressure on the gas pedal or not and allows us to create more realistic sounding gear shifts and decelerations.
The difference between both manual presets is that on the “free” mode, the relationship between the granular RPM and the master rev knob is completely linear by default, so you have to play with the RPM value yourself to imitate the act of shifting gears. Here is a video of me just doing that with a Revs pass first, followed by a Load pass. As you can see, to achieve a natural result, you need to drive the parameters in a realistic way. It would require a bit of practice to follow onscreen action like this but it feels very easy and responsive.
On the other hand, the other preset type, “Manual Gears”, has the gear shifting already soft coded into the mod section, including on load and off load changes. Of course, you can tweak this as you please but the preset gives you a nice starting point. As you can see, in this mode you don’t need to imitate the engine revs with your automation and you can just use curves to describe how hard you want to accelerate or decelerate.
For the most part, this works quite well when going up on the revs but going down forces you to go through the whole set of gears which doesn’t always feel natural, although sometimes you may want this (Formula 1 cars kind of do this sometimes). I tried different ways to avoid this, like staying within the boundaries of the same gear or jumping fast from a higher to a lower point on the envelope, although this needs to be carefully drawn as automation. A potential solution to this issue would be that the RPM ramps don’t occur when we decelerate, only on our way up on the revs knob.
You can also notice how the load drops are already coded into the revs progression, which is pretty handy and also shows that I was too subtle with it on my free test.
The third preset, Auto Gearbox, uses the auto option which doesn’t allow you to directly control the granular RPM or load and simply gives you an slider called “Power” that we can use to accelerate or brake while the gears shifting is hard coded and can’t be tweaked. This would be similar to driving an automatic car.
Here is an example of me using this mode. Compared to the others, it feels a bit unresponsive at the start but once you get speed it works well, although the gear shifting doesn’t always feel “in the right place”. As long as you don’t need very precise and fast changes in RPM, this mode can be useful to get natural results quickly.
By the way, you may hear some clicks and pops on my examples above. I am not sure a 100% if this is coming from Igniter’s or was a internal audio recording problem but definitely the Audi R8 seems to be a bit more “clicky” on the exhaust than other cars I tried later.
Lastly, the granular section also includes some other advanced controls:
-Shuffle Depth controls how thin or wide is the slice that the granular engine uses to select the samples. Higher values can help make the sound more natural and varied. Using the mods, you can, for example, make this value go up as the RPM goes up.
-RPM Smoothing: It slows down the response time to the changes in RPM. You can try increasing this if the engine feels too wild or decreasing it for a more fast response, which could be useful on auto mode.
-Idle Fade: Use this to adjust the fade between the engine on idle and low revs.
-Crossfade: It controls the blending between different grains or audio slices making it more abrupt or smooth.
-Lim Threshold & Kick: The documentation doesn’t cover these two but I suppose they are related to an internal limiter.
It includes 5 oscillators with two different waveforms each that you can blend together. You can also control the frequency and gain of each of the oscillators. There is frequency and amplitude modulation available for each oscillator plus a vibrato option.
And that’s pretty much it. Sounds basic but it is indeed powerful as you are able to link any of these parameters to the master rev knob creating dynamic designs that will grow in intensity and speed as the revs go up. You can also combine synth layers with real engines to create hybrid engines that combine real recordings and synths.
Here are some examples of sci-fi designs I did from scratch. Something that I missed is more options for the noise generator. it would be great to have more noise colours to create textures with or maybe a filter to shape it. The ability to apply separate FX to different oscillators would also be amazing.
This tab allows you to trigger certain individual sounds on specific moments on the rev progression curve. Maybe the most obvious use for this section is to trigger tire skids when we go up on the revs or screeching breaking sounds when we go down. In any case, this section is great to add sweeteners and flavour to the design.
There are four slots where you can drag and drop sounds. Unlike the granular engine, you can use your own sounds here and drag and drop them from finder. Each slot can be monitored independently and there are individual knobs to control volume and pitch. Both of these can also be controlled with an envelope instead of a knob, which offers interesting possibilities.
On top of the sample area there are four “timelines” each of them corresponding to one of the slots. Here is where you can choose when do you want the samples to be triggered but the horizontal axis doesn’t represent time but rev progression. In other words, you get to decide where in the acceleration curve you want some samples to be triggered.
Directionality is also accounted for. You can trigger samples as the revs go up or as the revs go down depending of where the triangle is looking. You can also have a sample that will be triggered both ways (diamond shape) and stop currently playing samples on the slot (square shape).
In general, the system is clever and nice to use but I feel that you’d really need some playlist and randomisation controls to make it really powerful. My idea would be to basically turn each of the slots into something like an fmod event. This way, you could add a playlist of sounds and control how to cycle through them or randomly jump between them.
This will give you a much richer system, where you can use sets of skids, terrain or engine pop sounds to choose from each time the event is triggered. For this to work well, you should be able to choose how deterministic the system is, in case you need predictability. Being able to tweak or re-shuffle the samples that were triggered after a pass would be also a good approach. I know Krotos is working on a run-time, middleware version of Igniter, so maybe something like this is already in mind.
Although the one shot section includes an option to loop its samples, this tab gives us much more power and control of sounds that need to be looped. It can be used in conjunction with the granular system or just by itself to create a completely new vehicle system.
This is pretty powerful. It allows you to have your own responsive car design, provided that you have recordings of steady RPMs to use. You can also use the loop section it to add texture or detail to the granular generator. You can add things like gravel, dirt, snow, clattering, squeaking or engine pops and link their intensity to the master revs knob.
You have four slots for loops and you can control their volume and pitch. The interesting and very handy thing is the section on the upper side. It allows you to customise how you want to blend your four loops together giving you the tools to smooth out both the crossfades and pitch changes between the transitions.
To obtain a good result, you need to make sure you have audio clips that loop cleanly. The Amp section helps when determining the boundaries between the clips but I miss more control on the actual volume of each of the sounds when I need to balance them out. I’ve noticed that actually some of the factory presets use the mod section to control this by using the general gain of the whole looping section but this strikes me as bit left field. Shouldn’t I be able to control the gain of each sample with the amp section? A gain parameter independent of the crossfades is needed here, I think.
On the other hand, the Pitch section is very nice to have and it works well. It would be amazing to actually being able to analyse the pitch of each of the samples and get a “suggested pitch curve”. This could be just an starting point so you can then tweak them by ear later.
The workflow in the loop section is a bit odd since you can’t hear anything unless the main engine switch is on but then if you switch it on, the first loop triggers so you can’t hear what you want to hear in isolation unless you manually mute the first slot. It feels kind of odd. Additionally, when building the loop progression, sometimes a slot doesn’t emit sound and you need to manually hit its play button. Kind of annoying.
So here is an example where I’ve built a Peugeot 307 engine from library recordings. For sure, the result is not as smooth as the granular presets and it sounds a bit “processed”, it’s like you can hear the artificial pitch bending too much. There are also many dropouts in the audio level and I don’t know if this is my fault or if there is a way to remedy that. The factory presets that use the loop system are cleaner than this but I can still hear some dropouts on those so maybe this is a bug?
As for the sound in general, it depends on how you drive the RPM and I assume creating a robust and good sounding vehicle system takes more sample preparation and tinkering than my quick test took. I was also thinking that maybe I chose the incorrect range of RPM loops and I missed having more slots so I can use more RPM states and make the progression smoother.
It is used to choose and monitor samples for the granular, one shot and looping sections. The tagging system is very nice. Igniter includes a nice selection of different engines and sweeteners, many cars also include recordings of doors, horns or wipers ready to use. I’ve noticed that you can’t drag and drop these sounds from Igniter to Pro Tools, which will probably be my first instinct if I just want a car door sound on the DAW’s timeline. The alternative would be to have the sound on the One shot section and either trigger it via Pro Tools automation or via the timeline system.
Other than the factory sounds, you can also use the “Files” tab to browse around your own computer files, including external drives, which is very nice.
Something I’ve noticed and is a bit counter-intuitive, is that in order to preview a sound on the browser, the engine button needs to be on, maybe that’s the case because that button just mutes the whole plugin internally but it took me a minute to figure it out.
As I have mentioned before, this is a very powerful and important section of Igniter and probably the one that I liked the most. It reminds me of Propellerhead’s Reason where you can flip the rack and apply envelopes and LFOs to any parameter in the system.
Basically the mod section allows you to link any parameter within Igniter to the master revs knob. You just need to drag the name of the desired parameter and drop it on the mod area. Then, you can edit the envelope that will govern this behaviour and also use an LFO to add some randomness or movement to any of these relations. The range or scale of the change can be adjusted with the sliders that appear to the right of each parameter. There are 8 mod slots so you can create very different envelopes and very complex systems.
By default, the RPM within the Granular section is inked linearly to the master revs and from there, you can link all sorts of other stuff, including FX, to make the engine more dynamic and responsive. Have a look at the presets to get some ideas of what you can do with this, it really allows you to get creative.
I was also thinking that it would be very nice to be able to use the mod section on other things than the RPM. As an experiment, I tried to turn the master revs knob into a distance knob, decoupling it from the granular RPM and linking it in several ways to volume, reverb and EQ.
Why would I want to do this? Because controlling the distance and perspective between shots is probably one of the most time consuming things to do in a vehicle scene. My experiment kind of works although when you do this, you loose the ability to link other stuff to the vehicle RPM. So, for a really powerful, all in one, vehicle design tool, I would love to have 3 master parameters: Revs, Distance and a maybe a third custom one. This is maybe outside of the scope or workflow that Krotos had in mind but that is at least how I would try to design it. Of course, you can create a similar effect just on your DAW but using this method you are able to link many things at once to the “distance knob” like engine/exhaust mix, granular FX, reverb sends, etc, speeding up workflow massively.
This section is pretty straight forward, nice to use and clean. You can control the level for each of your audio generators plus you have an FX send and Pan pot. While the sends and FX are pre-fader, the Pan is post fader. Each section has a rack with 5 slots where you can hook up FX. The FX that we can use are:
-EQ: Very nice parametric EQ with everything you need. Works great.
-Compressor: Very good too, with a gain reduction meter and a limiter mode.
-Limiter: Simple and clean dedicated limiter, useful to make sure you don’t saturate the output at high RPMs.
-Saturation: Good for adding some extra nastiness to an engine with extensive controls and colour presets.
-Transient Shaper: An unusual addition to a plugin like this since engine sounds don’t have many transients but it could be cool to use to add or remove dynamics to the granular section or on sweeteners.
-Flanger: Nice for sci-fi designs.
-Noise Gate: I suppose it could be useful if you have a noisy recording on your one-shot section.
-Ring Mod: Pretty cool and alien sounding and a nice addition for creating sci-fi stuff.
-Convolution reverb: Very good to have to recreate distance or an “in-car” sound. The controls are quite simple but you probably don’t need much more. I miss more outdoors IR in the factory library.
-Doppler: Very nice if you need to quickly cover passbys. You can control it independently or attach it to the main Revs knob. Passby presets are already created for each vehicle which is very handy.
In terms of workflow, Igniter allows you to create the engine RPM movements in a very quick and flexible way and of course, you can always come back and tweak the automation to make it work better. Additional passes controlling other parameters (like load) can add extra realism and detail.
The loops are nice to have since you can, for example, make any car go on gravel or dirt, for example, with just adding a loop layer to the granular. The one shots are not that useful, in my opinion, since you can only have five individual sounds and you can’t assign probability or playlists to the triggers, so every time you pass through them on the RPM curve, you would hear the same exact sound. The way it works right now, I think you would be better of just editing sweeteners like skids manually on your DAW the old-fashioned way and use Igniter for the engine itself but I’m open to be wrong about this.
You would probably need two instances of Igniter, one for exterior shots and one for interiors, unless you want to do the interior treatment outside. Once you have the basic RPM behaviour down, you would then need to mix it into the scene with fader work, pan and distance attenuation. That’s why I was thinking that it would be cool to have a dedicated master distance knob so you can tweak this in one go once you find a reverb that works with the scene. With these system I’m imagining, you would do an RPM pass, a distance pass, some tweaks here and there and you would be done for that car. Rinse and repeat.
Lastly, it’s also important to mention that Igniter offers a multi output so you can get an individual signal from each layer and mix them in any way you want in your DAW. This is very much appreciated.
Krotos offers an expanded version called “Igniter Full Tank” which includes all the unprocessed and processed recordings used to build all the presets. You get a lot of coverage for every vehicle in Igniter plus loads foley and sweeteners. The recordings are a great library just by themselves (75 GB of additional audio) and in combination with Igniter will allow you to cover every single detail and sound you may need. To clarify, these extra sounds come as separate audio that you can then browse within Igniter, but they don’t include new presets or vehicles.
I hope both you and me now have a good understanding of how Igniter works and what it can offer. I had a lot of fun testing the plugin, Krotos keeps giving us innovative tools to create custom, unique soundscapes and I feel that with them we can offer much more value to our clients because the result is unique and personal.
Above all, the granular system sounds great and I know how hard is to make interactive engines sound good. I’m sure more content will come for the plugin in the future and maybe some workflow quirks will be fixed with time. As for the features I’ve been suggesting, they are just my own take on how I would improve the software’s workflow and capabilities and since I’m sure some concepts and perspectives have escaped me, I will remain open to new and better ways of using Igniter as it spreads across studios worldwide.
Thanks for reading!
The fascinating thing about Impressionism is that it assumes that a painting is never going to able to recreate reality as accurately as a photograph. Once you leave behind the burden of precision, the artist is free to do what art does best: expressing a feeling, a mood, a state of mind. Impressionism relays more on movement and light than shape and form. The composition is open and the boundary between foreground and background is blurred.
An impressionistic painting doesn’t look like a real place but a distant memory, the impression a place leaves deep in your mind. It looks like the blurry pictures from a dream that linger in your mind just before you forget them.
That’s pretty much how far my artistic knowledge goes but I hope you get an idea. I was thinking that it would be cool to try to translate that approach into sound design by creating soundscapes to go along with some impressionist paintings. But before we do that, we can’t forget that, in a way, this already happened among a very specific sub-section of sound designers. The ones that limit themselves to a narrow amount of defined pitches and timbres: music composers.
I was introduced to Impressionist music by the amazing series Young People's Concerts by Leonard Bernstein which I really can’t recommend enough. If this is the first time you hear about it, just go watch it. There is a whole show about Impressionism.
He does a better job than me in explaining it but, basically, when impressionism is translated into music we are trying to express the feeling, the essence of something in a subtle and seductive way. We are not explaining, we are suggesting. This often results in dreamy melodies (whole tone scales are a staple) and the use of exotic and unresolved harmonies. For the most part, composers limit themselves to traditional instruments but they try to get the most from them in terms of timbre. Piano is probably the instrument of choice for Impressionism, using its large range, dynamics and polyphony (pedals are heavily used).
As an example, here is what musically happened when Manual de Falla, who was born in Cádiz like myself, moved to Paris and met the Impressionists. Maybe is not its most known side, but sometimes flamenco has a dreamy, exotic quality that I think is perfect for this style of music.
And here is a maybe a more canonical example by Debussy. Notice how the melody is usually unresolved. Like in a dream, you don’t really know how you got there and there is no clear conclusion. This music maybe doesn’t sound that different or special to you, as these traits have been assimilated into mainstream music (think jazz) but keep in mind that in that time it was quite a contrast to the musical establishment.
If Impressionism doesn’t want to be constrained by shapes, colours or composition, maybe the most logical way to translate this idea into sound would be to forgo concepts like harmony, melody or rhythm. When you do this, only timbre is left and since shaping timbre is kind of my job, it sounds like a perfect fit.
My first approach to a Impressionistic soundscape is simple: just create an auditory complement to the visuals, extending the world within the painting to a new sense. Let’s lay down sounds that could exist in the scene and that go well with the feeling it transmits.
I’m using first the one that gave the style its name (and it was meant as an insult), “Impression, soleil levant” by Claude Monet:
Here is a second one, using “Woman in the Bath” by Edgar Degas:
At first, I thought I would use reverb to blurry sounds together in an analogy of how painters mix colours. But I soon discovered that doesn’t work very well. For the bath painting, I wanted to express a feel of intimacy, a sense of “costumbrismo” which actually was one of the other features of Impressionism: to portray everyday life.
Reverb doesn’t help with this because it creates an unnatural space that doesn’t complement the painting but opposes it. Monet’s Sunrise scene uses more reverb but only enough to match the environment that we are being presented.
One more thing was apparent: it helps to have elements in the scene that suggest motion, since most things that make a sound are moving in some way.
Here is “Effect of Snow on Petit Mountrouge” by Édouard Manet.
Since this painting was created during the franco-prusian war, I decided it could be cool to also tell a little story within the soundscape. I wanted to capture the peaceful calm of a winter snowy day somewhere in Paris. The calm is then broken when distant cannons are heard and the french soldier who is contemplating the scene has to go back to his post.
Finally, here is “Gare Saint Lazare” by Monet again:
I chose this one because I liked the painting from an aesthetics point of view, it has movement and life. And of course trains are a nice sound design opportunity.
After working on these four soundscapes, l realized I was mostly describing the scene and maybe transmitting some of its essence by choosing certain sounds but not being technically impressionistic. I was basically adding a soundtrack to the painting.
Their relaxing, atmospheric quality goes well with audio that borders on being ASMR. It’s somewhat ironic that the best complement to an impressionist painting is a soundscape that does the opposite: being descriptive, detailed and realistic. Maybe it makes sense in a way. These paintings suggest instead of being explicit so there is room for audio to add to the experience.
Of course this got me thinking about how it would be to create soundscapes for other art styles. Probably the ones that distort reality in different ways like expressionism or cubism could be good candidates. Maybe something worth exploring in the future.
But can we use audio in a way that gets to the core idea of Impressionism? To do this, we would need to go more experimental and abstract. We would need to stop using descriptive sound, forget about what you can see and focus on the feeling the painting creates.
I thought about using Paulstretch since if you play with the window size, you can blurry and smear sounds together, like painters mix colours. This worked nicely as Paulstretch tends to sound very dreamy. The following soundscape was created from only one audio sample, this recording of some wind chimes:
I created different layers in Paulstretch playing with the window size, pitch shifting and adding harmonics. I refrained form using any “real” audio. Here is “The Cliff at Étretat after the Storm” by Monet.
As you can hear I’m getting somewhere interesting. I tried to evoke a warm summer feeling although I’m sometimes dangerously close to the line between being dreamy and being unsettling. My first instinct to solve this was to use music tricks, like pitching layers a fifth away from each other but I didn’t want to relay on musicality too much.
Here is another darker example using a fantastic painting, “Winter, Midnight“ by Childe Hassam:
This one was created from a music stinger. If you hear both closely, you can tell it’s the same base sound but in a drone, dream-like state. It works well because the musical impacts are stretched creating some movement in the soundscape and some changes in tone.
And finally the last one turned out quite creepy, maybe too much for the painting but I like the result nevertheless. I used a combination of layers from Paulstretch, using the tonal / atonal slider to remove most of the “musicality” from the sounds (which were kind of musical). Here is “Moonlight, Isle of Shoals” by Childe Hassam.
If this got you interested in learning Paulstretch, I have a blog post about it that goes deep into how it works.
It’s cool to work with the concept of “pure sound design” without the burden of mere description but at times it seems to feel too close to atonal music. That last soundscape got me thinking about Ligeti and Penderecki. But is this something bad? Maybe is atonal music which is too close to “pure sound design”. Maybe they are the same thing but looked from different perspectives.
In any case, both approaches to the creation of a painting soundscape are valid and worth pursuing, I think. Just the idea of using visual art to inspire audio work is a good way to get your creative juices flowing and tackle things in a different way.
Other than that, I was also reminded that sound is not only simple description, it also conveys feelings and can somehow capture the very essence of a place, an action or a character. That’s something to always keep in mind.
Hello! Here are some ideas and tips that I think could help you make better decisions while buying audio equipment.
I like to see any piece of gear as an investment so I try to choose products that are known for being robust and durable. There are always cheaper options out there but I don’t mind paying a higher price if I have a better guarantee that the equipment is going to last for longer and be more reliable.
In order to determine durability, a good hint could be that the manufacturer offers a longer guarantee period than legally required and/or a good reputation among veteran users (some detective work in audio forums is a must). It is also a good sign when a product is manufactured in Europe or the US, although this is not very frequent and doesn’t guarantee a higher quality necessarily.
Buying higher end gear is particularly relevant for audio since electronic components are quite important in determining quality and life expectancy. The use of cheap plastic instead of more durable components like metal is also commonplace and something to avoid, specially in field equipment.
Something else to think about is that durable gear is usually well known in the industry and may give clients some extra confidence to hire you before others.
On the flip side, you can’t always afford to buy higher quality equipment and sometimes you may need to opt for entry level gear. This can also happen when you need an specific thing for a gig and don’t have time or money to find the best possible option. In those cases, well, you probably need to bite the bullet but in general my advice would be to wait if you can. Flip more burgers and sweep more floors. Once you have enough to at least access the mid tier, go for it. In my experience, those investments will pay off. Ten fold. You need to spend money to earn money.
I bought a Tascam HD-P2 in late 2011. I chose this model because of its reputation and quality. To this day, I still use it as my main recorder for sound effects. It has also accompanied me through features films and documentaries, on snowy cold exterior days and crazy hot Seville summers. It has never failed or died during a take.
I am not saying the HD-P2 is perfect. It only offers two microphone inputs, the pre-amps are not ultra clean (but they quite good for their price range) and the powering options are limited. Nevertheless, it served me well throughout my first years working in audio, it gave me confidence and allowed me to get a huge return on my investment.
The mighty HDP2. Respect.
I think this is key. Don’t get dazzled with fancy stuff that you are never going to use. It is important that you think about the features that you actually need and then look for the best option the market has to offer.
Hopefully, I will have the chance to record more frequently now.
Of course, in order to do that, you need to know what your needs really are, which is the tricky part. Do you prefer more channels or a higher resolution? Bigger memory or longer battery life? If you know what kind of specific work you are going to do, this is going to be easier to decide. Try to narrow your needs and priorities.
I recently bought a Sony PCM D100 because I wanted to have something portable to record on the go. This recorder is quite expensive (for a handheld device) and doesn’t have XLR inputs which for me is a big issue. But the thing is my goal is to have something really portable so I can record in situations when a big rig would be cumbersome.
So I am losing the XLR feature in exchange for great quality of audio, battery life, internal memory and construction. All of them features that are essential if I’m going to use this on the go.
Sound is something that can be objectively measured but, nevertheless, the way we experience it is quite subjective. People apply all sorts of descriptions to audio like “silky”, “airy” or “muddy”. I’m not saying these are not useful or that these don’t describe real properties but sometimes I think we get caught up in these terms too much.
This problem is twofold. On one hand, sometimes people are so ready to justify their purchase that they start to hear mystical properties in a piece of gear. On the other hand, sometimes we can actually really tell the difference (in terms of clarity or timbre profile) between two pieces of gear but it is so small that it’s only noticeable while soloing and/or A-B testing. If the final consumer is probably not going to tell the difference, is it really that important?
Don’t get me wrong, I still think that audio quality should be a priority but usually when investing in equipment the very expensive stuff gives you diminishing returns. You need to really expend a lot of cash to get from the professional to the “elite” level. Maybe you don’t need to.
So yeah, choose quality but don’t get crazy. Beware of mystical claims and 20K€ cables. I honestly think that if we forced people to take blind A-B tests comparing decent gear with very high end equivalents they would be amazed with how close they can be.
Before buying a new fancy microphone, maybe stop for a second and think about the small stuff. There is always something outdated or in a bad condition. Maybe it would sensible to improve on those weak areas first.
Sure, you don’t need fancy solid gold cables but get yourself some decent ones. Another good example of this could be battery management. If your gear uses batteries of any kind, invest in good chargers. I recommend you get familiarized with the stuff that video and photography folks use. Smart chargers are a great option since they have independent charging cells and programs to keep batteries healthier.
Audio cases (I like Portabrace) are also a great option to make sure your equipment is safe while traveling or on location. I bought my Tascam HDP2 with a Portabrace case and it’s really a worthy investment. The velcros work like the first day eight years later.
This powerex charger is a very nice option if you need an army of batteries for your recorder and/or wireless kits.
Some people are more risk averse than others and this is something you need to take into account. In my case, I don’t feel confortable rushing things or spending large sums of money so I try to avoid doing those two things at once. If you are similar to me, remember that at some point you have to take the leap and is going to feel uncomfortable. But that’s good. That’s what they mean when they say “Is good to step out of your confort zone”.
When I bought the Rode Blimp v1, I could not afford anything better. It’s an OK starting point, but I would not recommend it for a long term investment. Not very durable.
If, on the other hand, you tend to rush things, well, take it easy. It may help to give yourself some time to make sure to make the right decision. Sharing your situation with friends or colleagues may help too, you’d be surprised by how much better you can see things when you articulate them out loud and get feedback.
Personally, I don’t like to buy second-hand stuff because I feel like I’m taking a big risk but if you are confortable with that, it’s definitely an option. It helps if you can check the condition in person and knowing the seller is ideal. If you are buying online, using sites with a reputation system is a must. Other than that, second-hand is a risk that may pay off or end up in disaster. So ask yourself: how much more money am I willing to pay to get peace of mind instead?
Any piece of equipment that is reasonably popular is going to have some scary reviews. That’s the nature of the polarized online world: people only bother giving 1 or 5 stars, so there isn’t much nuance. Having said that, reviews are still a valuable resource when used with caution.
My approach is to focus on quality rather than quantity. Sure, you can found many reviews in Amazon nowadays but I would prefer to check audio forums or specialized stores first. You can also check reviews for a product on online stores that you are not planning to use. If you are in Europe, B&H and Sweetwater are great. If you are in the US, Thomann is a fantastic source.
Other than that, your best bet is to join and participate forums like Gearlutz. With time, you’ll get to know people there whose opinion would probably be more valuable than a random Amazon user.
The Sennheiser MKH 416 was my first mic and almost the only one for some time, forcing me to use it in many different ways (on location, for foley, for SFX, for VO…)
Scarcity may sound like a bad thing but I think you can learn a lot from it. Limiting yourself to a small number of tools forces you to be creative, try new things and of course you will master them. Is hard to do that if you have too much stuff so my advice would be to really make the most of what you have before buying something new.
For me, a good example of this is audio libraries. If you already have a decent amount of sounds, there is probably a lot you can do with them. Doing sci-fi or fantasy sounds, for example, will force you to experiment with what you have around in terms of recording gear and plugins and you will learn far more than if you just buy yet another library.
