Absolute vs. Relative OSC messages

Traditionally, OSC messages require users to specify the desired value for a parameter. This approach is ideal for scenarios like setting precise trajectory values. For example, to set a parameter to a specific value, we historically used an Absolute OSC message:

/room/1/gain/ -10

In contrast, Relative OSC messages allow users to define an offset rather than a specific value. This means you can adjust parameters incrementally, such as increasing room gain by +3dB or decreasing source distance by -2m. This flexibility is amplified when you don’t have bidirectional integration, as the parameter current value is unknown, and can simplify the interaction with SPAT Revolution.

How Relative OSC messages work

Relative OSC messages follow the same structure as standard OSC messages, with the addition of a path keyword. For instance, to offset the room gain by +3db, you would use:

/room/1/relative/gain 3

This consistency ensures that the OSC path structure remains unchanged, allowing users to utilize wildcards to target multiple objects or add interpolation timing. For example, to offset the distance of all sources by 2 meters over 5 seconds, you would use:

/source/*/relative/dist 2.0 5.0

Getting started with Relative OSC messages

To assist you in leveraging this new feature, we provide a QLab5 template with examples of Relative OSC messages. This resource will guide you through your initial steps and help you explore the full potential of Relative OSC.

You can find all available controls in the updated OSC table, which includes all available OSC commands in SPAT Revolution.

QLab 5 Relative OSC Template

We encourage you to try out this new feature and share your template and use cases with us!

The post Introducing Relative OSC: Simplifying Audio Parameter Controls appeared first on FLUX:: Immersive.

Defining the reproduction strategy (Speaker system & panning technique)

While this article does not aim to delve deep into the subject of system design, it is worth discussing a significant challenge behind the ‘From Studio to Live’ journey. In essence, it’s about content creation, taking place in an ideal monitoring ‘sweet spot’, and live deployment, which invariably involves compromises and spread out audience coverage area. Notably, this discussion touches on the reproduction strategy to adopt, where the position-based family of panning techniques such as DBAP, KNN, and the more sophisticated WFS are our best friends for such live reality.

It is tempting to suggest, ‘Let’s surround our audience with a large number of speakers,’ and indeed, increasing the number of loudspeakers has the potential to enhance ‘resolution’, or, if you prefer, the accuracy of soundscape reproduction. However, for everyone to fully experience this heightened accuracy, we must ensure that each loudspeaker or loudspeaker array offers adequate coverage and sound pressure level (SPL). Ultimately, a reproduction strategy aligned with the artistic intent but conditional to a capable system stands as the key to achieving success.

The Role of Transcoders in SPAT Revolution:

Contemplating the use of transcoders, such as master transcodes, in SPAT Revolution for converting between speaker arrangements prompts inquiry. It’s crucial to note that these transcoders primarily facilitate the transcoding of ambisonic stream formats to channel-based speaker setups. Channel-based to channel-based transcoding serves as a matrix tool only, not a medium for down/up mixing from one speaker setup to another.

The construction of a SPAT Revolution soundscape

Understanding the components of the soundscapes in SPAT unveils essential elements to understand for a successful transition across systems:

Source Object (Position and parameters)

The foundation of the soundscape you are building is about manipulating all the source object properties. The efficacy of panning techniques and the reproduction system determines the experiential outcome. Adapting speaker arrangements and techniques in SPAT Revolution facilitates easy migration to diverse systems. This can be pretty much done ‘on the fly’.

Source Object Attenuation Model

The model, relying on source distance from the center reference point, influences amplitude and air absorption simulations. When employed, the protection zone establishes a pivotal threshold where our processing initiates the ‘simulated’ distance attenuation, influencing both amplitude and spectrum in the case of air absorption (minor high-frequency roll-off). This simulation aims to replicate our natural hearing experience in relation to distance. Overlooking this aspect while ‘scaling’ a mix can notably influence the resultant mix quality. By default, SPAT standard normalized arrangements set the distance threshold at 2 meters, aligned with the speaker line and typically extended to the farthest speaker in larger or non-normalized systems. Thus, this distance, spanning from the center reference point (0) to the protection zone’s boundary, defines the threshold. To illustrate, if this threshold is set at 10 meters and your source is 20 meters away, the distance has doubled, resulting in a loss of 6 dB per the physics principles. The drop factor ratio, a key source parameter, governs this 6 dB reduction. Notably, this concept adheres to a spherical nature and isn’t contingent on the speaker setup’s geometry. Consequently, appropriately scaling this factor in tandem with session setup alterations proves pivotal. This adjustment will impact the ‘direct/reverb omni ratio’ when utilizing SPAT’s room reverb engine.’

The Room’s Reverberation Model:

When applied to some or all audio source objects, the ‘room’ reverb settings play a central role in tailoring the mix to achieve the desired room ambiance. This adjustment typically occurs on-site, allowing you to mold the mix’s sonic character. However, challenges arise when transitioning from a rather acoustically inert or ‘dead’ room to a space with pronounced reverberation. In such instances, careful consideration is needed to either harmonize the reverb model with the new environment or judiciously omit or limit its use. A good starting point is typically changing the first reflections settings (Early and Cluster) and the overall reverb gain. Relocating from a studio setting to an open outdoor space presents a comparatively simpler scenario, as it offers a higher degree of control over acoustic characteristics.

Bringing It All Together

Now, having meticulously crafted your mix or artistic creation within the confines of the studio, the pivotal moment arrives: the transition to the live venue. Throughout the creative process, you’ve been fine-tuning your mix on a specific speaker monitoring setup, utilizing a set protection zone that emulates distance, and potentially crafting snapshots and automation using tools like DAWs or remote control applications such as QLab. As you establish a new speaker arrangement tailored to the venue, the transformation begins. The instant you actually apply this new arrangement into your mix session, SPAT Revolution springs into action, automatically initiating a scaling process. An alert marked ‘NEW SPEAKER ARRANGEMENT’ is announced, advising you that the selected new arrangement speaker distance differs from the previous one. Consider this: if your studio’s farthest speaker sat at a distance of 5 meters, and the new venue configuration extends it to 25 meters, SPAT Revolution will propose a scaling factor of 5. When accepted and applied, this factor cascades across multiple facets:

• Each current source’s distance parameter
• The protection zone, integral to the attenuation model
• A distance scaling factor within the room output section dynamically impacting recalled snapshots, plugin-based automation, and incoming OSC messages.

And there you have it – a seamless integration of the new arrangement. This methodology seamlessly synchronizes with direction-based techniques, as it fundamentally maintains the consistency of angles that these techniques rely upon. Additionally, the protection zone, enriching the mix with its attenuation model, undergoes appropriate scaling. Consequently, all automation, snapshots, and remote communications harmoniously adapt to the new system’s scale, thanks to the dynamic scaling fostered by the distance scaling factor within the room output section.

Navigating Challenges and Conclusion

While the automated scaling works seamlessly with direction-based techniques, such as vector-based methods, it might not serve as a panacea for position-based approaches. In other words, this automatic scaling may not wield an all-encompassing solution if you’re employing position-based techniques, such as DBAP or the Wave Field Synthesis reproduction approach. The intricacies arise due to the unique geometry of the speaker arrangement, rendering it more complex than a mere distance-based scale factor. The alteration in the spatial relationship between the speakers and sources, as dictated by the new setup, can yield substantially different outcomes. It’s essential to note that this doesn’t imply a breakdown but signifies a distinct rendition of the audio.

In such cases, it may be strategic to consider the pre-production environment, and the in-studio monitoring system to be more aligned in reproduction technique and speaker setup geometry to the future venue system, such as with a scaled-down version.

The post Navigating SPAT Revolution – From Studio Creation to Live Deployment appeared first on FLUX:: Immersive.

LFE vs Bass Management

Initially, it’s important to clarify that we shouldn’t confuse the primary role of the LFE (Low Frequency Effect) channel with the function of subwoofers. While the LFE bus is specifically used for low-frequency effects, subwoofers not only handle these effects but also assist with bass management to complement our main speakers, especially considering their limited low-frequency range. Whether it’s in a mixing/monitoring system, an in-venue system, or even certain home systems, there should be provisions to accommodate this. Additionally, some of the methods mentioned are tailored for unique subwoofer content, deviating from traditional approaches.

Ambisonic Generalities

Ambisonics is often described as a scene-based spatialization technique. Audio channels do not refer to a specific speaker in space, but rather to a part of the space. To make it work, ambisonics has to be decoded to a certain speaker layout. So if you are creating an ambisonic format mix (encoding) or simply want to audition an ambisonic source from a recording, it will need to be decoded to a layout.

Ambisonic panner rarely gives access to some kind of aux sends to feed a Low Frequency Extension bus. Equivalently ambisonic decoders rarely directly support subwoofers or bass management. This article will discuss three ways to handle such use cases in SPAT Revolution.

Reproduction Workflows

Reproduction workflows suppose that you have an ambisonic mix you want to decode on an existing speaker layout. We are then looking for a bass management solution over the ambisonic stream.

Omni Sub Send

The initial case we’ll address involves scenarios where spatialization in the subwoofer isn’t desired or necessary. In these situations, we can directly route the W channels from the ambisonic stream to the subwoofer channel.

Extracting the W channel to feed a subwoofer

By following the setup shown in the screenshot above, you can then change the send level to the subwoofers by using the gain of the ‘W to Sub’ master block.

You can download this template to test this approach.

Specific decoding for subwoofer

If you wish to have a sense of spatialization in the low-end, you could use a second ambisonic decoder that specifically targets the speaker layout formed by the subwoofers. Here, it is often preferred to use the ‘in-phase’ decoding strategy, as it will make sure that no out-of-phase signal will be generated. It comes at the price of a worse sound localization experience, which should not be too much of an issue in this spectrum region.

InPhase decoding mode for low-end content

See this template for further information.

Creation workflow

Per Sources LFE Send

It is possible to create an ‘LFE send’ kind of workflow that is independent for each source. The idea is to create a second room dedicated to the subwoofers. In a first approach, this room should be set to mono, and have the reverb muted. You can then use the room-specific as a ‘send to LFE’ control.

Example with a mono room acting as a LFE bus

To easily see the gain parameter related to the LFE room, you can use the search field in the parameters section. For example, strict: RoomGain2only displays the specific gain of the second room.

Example of control filters to only display the gain of the second room

Download this template to get more detail on this setup.

To extend this concept, you could use a second HOA room instead of a channel based mono room. Such an ambisonic room should be set in 2D mode, as a 3D subwoofer arrangement seems like a non-existent use case, and may also use a lower order than the main room. This will then give you two ambisonic streams, one dedicated to full range speakers, and one only to subwoofers.

Dedicated Ambisonic Room for LFE management. Note the use of dedicated HOA output for recording purpose.

See this template for further details.

It is also possible to combine both a dedicated room for LFE content and a bass management strategy to cover most of the playback situations you may encounter.

Conclusion

It is possible to handle subwoofers in Ambisonics reproduction by:

• Extracting the W channel and sending it to the subwoofers
• Using a second ambisonic decoder to target only the arrangement of subwoofers.

If you are at the content creation stage, and are already working with a HOA room, you could use a second HOA room to replicate a ‘send to LFE’ type of workflow.

The post Ambisonics: How to Handle Subwoofers and Low Frequency Effect appeared first on FLUX:: Immersive.

In this article we will talk about digital equalizers, how they work and how you can improve your mixing process with them.

Stating the obvious

Equalization is a tool that allows changing the timber of an audio signal. Basically, it is a sophisticated volume control that can target certain parts of the spectrum. So we can boost bass or trebles, cut mediums, etc. Basically EQs are built with a certain number of filters.

Equalizers have a long history that we are not going to cover in the article. We will focus on parametric EQ. Equalizers are classified as parametric when their target frequency and quality factor can be adjusted by the user. This is mostly the kind of EQ we are using when mixing or mastering.

These parametric EQs have three main controls :

• The frequency; which changes the target area of the audio spectrum
• The gain; defining how much we want to add or remove the frequency range we have selected
• The Q factor (or bandwidth); which sets how wide the processed band is. A high Q factor is a small bandwidth. A little Q factor is a wide bandwidth.

Sometimes you can also change the order of the filter. The higher the order, the steeper the filter’s curve is.

We also use four main filter shapes :

• The bell shape. It only boosts or cuts a certain part of the spectrum centered around the set frequency.
• The shelves. It boosts or cuts everything above (high shelves) or below (low shelves) the set frequency.
• The low pass. Let only pass frequencies below the set frequency.
• The high pass. Let only pass frequencies above the set frequency.

How digital EQ is built

Without entering into too much detail about digital signal processing, there are two main types of filters; filters which have finite impulse response, and those who have an infinite impulse response. Let’s clarify this.

You may have heard the words “impulse response” when talking about reverberation. In this context, an impulse response is the reaction of a specific room to an impulse. In simple words, if you clap your hands in a cathedral, and record the reaction of the space to the clap, you will get an impulse response, which is really, just an audio sample.

If we were able to clap our hands inside a digital filter, and record its output, we would also get an impulse response. The good news is that we actually have a way to do that, and it’s called a Dirac.

A Dirac is a very specific type of signal. Theoretically, it is an impulse as short as possible. In practice, you can imagine an audio file filled with zero, except for one sample that is at full scale.

So back to our filters. To simplify the question, let’s admit that filters that have infinite impulse response (IIR) have the property to be real time, they do not add any latency. Filters that have a finite impulse response have the property of having a linear phase shift, equivalent to a certain latency. Again, let’s clarify this point.

Phase and EQ

Phase in audio is a complex subject. If you listen to the world around you, you will not perceive the phase. We start to perceive it when it changes in time or when you listen to two correlated signals.

Correlated signals are simply signals that are close enough together. For example, two microphones that are recording the same instrument at the same time.

We often talk about the phase shift, and we need a reference point to measure it and to hear it. There is some very well-known sources of phase issues :

• Using multiple microphones to record the same instrument
• Filters

All analog and digital IIR filters generate some amount of phase shifting. This means that, when you are EQing the snare top microphone with an IIR equalizer, you are generating a phase shift which changes the sound of the top mic with the bottom one and even the overheads.

The phase distortion generated by equalizer is totally solved when using FIR filters, or phase linear EQ.

Here comes the linear phase EQ.

So now that we know that analog and digital IIR introduce phase shifting on the signal, you may think, “OK from now on, let’s only use linear phase EQ”. And I can understand the idea, but sadly, in the world of physics, everything comes at a certain price.

It’s true that linear phase EQ does not alter the phase relationship between microphones, but they have two major issues. The first one, they add tons of latency. It’s not always a big deal when mixing, but for a live mixing desk or for headphone cues, it is not acceptable. Linear phase EQ also has the very bad habit of creating a pre-ringing effect.

What is “pre-ringing” you may ask? Well, first of all, we will admit that all audio filters have a post-ringing effect. Remember that impulse response we’ve talked about? When we “clap in a filter”, using a Dirac, we get a response that has some length. The filter kind of “oscillates” if you want. Post-ringing is not an issue, it really is how filtering works. It is not really audible because it is masked by the input signal itself.

Pre-ringing on the other hand is much more audible, because it happens before the input signal onset. At this point, you may scratch your head thinking that filtering is about time traveling. Unfortunately, no it’s not. But the good news is that we can delay things. If you remember correctly, we said before that linear phase EQs add “tons of latency”. So because there is this delay, we can hear the consequence of the filter before the input signal. And thanks to the delay compensation of our DAWs, everything stays in sync.

Filters that generate content before the input signal have a mathematical existence, but not a physical one. They are said to be non-causal (because the effects precede the causes).

Pre-ringing generates an effect comparable to a “reverse” effect, and it smooths out the transients. If the filter works in low frequencies, the pre-ringing will increase. To reduce the pre-ringing we need to use a longer impulse response, and thus, increase the latency.

What EQ should I use ?

I admit that these previous parts may be very confusing. But in practice, it’s not very complicated.

Linear phase EQ has too many drawbacks to have a common practical usage. Also, even the linear phase advantage is quite marginal. Let’s go back to our snare top mic we want to EQ. The phase offset generated by the bell and shelves filter is marginal and its effect on timber is hidden in the filter itself.

The graphs above shows the magnitude and phase response of the same filter curve blend with the dry signal (50%). The first one is a minimal phase filter, the second one a phase linear filter. Notice that the main difference in the magnitude is the difference in the bandwidth of the filter. The phase response of the minimal phase EQ tends to make filter curves narrow. So if you can simply compensate for that effect by having a smaller Q factor.

When it comes to low pass or high-pass filters, the discussion is not the same. The phase offset generated by these filters is much more important, and thus, we have a severe effect on timber.

This is why I often advise people to only use IIR equalizer, and to use shelves filters instead of high pass to clean their track. If a high-pass filter is really required, I would use it on a bus, where all the correlated signals get summed into, to avoid any phase distortion.

We consider that correlated signals are signals from the same instrument recorded in the same room at the same time. Bleed between microphones can also cause some correlation.

Should I use a high-pass filter to remove sub-low ?

If you have read the previous part, you should have an idea of the answer. First, and this may be controversial, but there is a strange habit to high pass everything. It’s not that rare to see the idea that we should “clean” the low end of each and every track. Which is strange because there is rarely anything at all there. So why even bother ? Moreover, when we start to use many high-pass filters on many correlated signals, it can create some really funky phase interactions.

As previously said, a low shelf filter seems like a better alternative for this kind of application. I would also strongly recommend to only treat the low end (or anything really) if you hear something there.

I personally prefer to use a high-pass filter to improve the phase interaction between several microphones. Depending on the order of the high pass, the phase shift will increase or decrease.

• At first order (6 dB/oct), we register a phase shift of 90° at the cutoff frequency.
• At order 2 (12 dB/oct), we find a 180° phase shift at the cutoff frequency.
• At order 3 (18 dB/oct), we find a 270° phase shift.
• At order four (24 dB/oct), we come back at 360°

Be careful, the higher the order, the more chaotic the phase rotation becomes.

If you add the order of the high pass with the polarity switch of the mixing channel, you start to have a nice phase adjustment tool. It’s even better if you happen to use the Evo Channel, which features a high-pass filter with a selectable order, and a phase adjustment dial to go from anywhere from -180° to 180° phase shift.

Should I only use subtractive EQ ?

We often see this advice on the internet that we should prefer cutting things we don’t like instead of boosting frequency. Another approach is possible, though.

Critical listening is the best tool to sharpen when mixing, combining that with a bit of common sense we can simply say that :

• If we find that there is a disturbing frequency in the signal, we can remove it with an EQ cut.
• If we lack something in the signal, we can emphasize this area with an EQ boost.

As a general guideline, we tend to favor greater Q-factor for cuts and smaller ones for a boost.

In the extremes of the spectrum, I tend to avoid shelves for boosting, as it tends to push too many frequencies upfront. Bells offer more precision. On the other hand, a shelve filter is very efficient to smooth out too much high-end and to remove harshness.

The Sweeping of Hell

When we search for advice on the internet for learning how to EQ, we often find this method : take a narrow band bell EQ, give it a good amount of gain and sweep around the spectrum to find something that you don’t like. Once it’s founded, simply cut it.

To be honest, I think it is a terrible method. I don’t know if you experience the same thing as me, but when I’m listening to a bell filter, with a narrow band, boosted, nothing sounds right to me.

I would rather propose another method. Listen carefully to the signal you want to process. Listen to it in the mix (don’t use the solo button). Pay attention to interaction with other instruments if some sources are masking each other. Pay attention to things you’d like to improve in the timber. Eventually, you can take some notes. Then, for each problem you want to solve, you will make a bet. For example, if I found a resonating frequency on an electric guitar that is disturbing, I will say to myself, “I think it is at 300 Hz”. Then, I set my equalizer to 300 Hz, to give it a boost to emphasize this area. If it’s there : good job! You can now cut it. If it’s not there, go back to a flat response and repeat the whole process. Is the problem higher or lower from what we’ve just heard?

When we bet on a wrong frequency, it’s worth trying an octave lower or higher. We often target the wrong harmonic. Also, if you can directly cut instead of boosting, it’s a good improvement exercise too.

The advantage of this method is that you actually train your ear to detect certain frequencies. With repetition you will get the right frequency faster and faster.

Also, It is now very common to have a spectrum analysis tool directly embedded in the user interface. The Evo Channel and Evo EQ feature the same praised spectrum analyzer found in the FLUX:: Analyzer. On a single channel, a frequency analyzer can give some precious information. We can instantly see the harmonic content of a signal and target the right spot for our treatment. While it should not replace the ears, it can be a great tool for beginners.

TL;DR

Equalizer is one of the most used tools in audio production. The form we mostly use as audio professionals is the parametric EQ. Equalizer is built upon filters, whose numbers and shape depends on the usage. In the digital world, there are two main digital filter types : IIR (Infinite Impulse Response) and FIR (Finite Impulse Response). We tend to associate the first type with minimal phase EQ (similar to their analog counterpart) and the second type with linear phase EQ.

While linear phase EQ sounds like a major improvement, it actually has too many drawbacks to be a solid alternative for minimal phase EQ. Actually, filter results on correlated sources are pretty similar between minimal phase and phase linear EQ, as long as you use bells or shelf filters. For high pass and low pass filters, they should be used with caution due to their important phase shift.

Critical listening is the best approach to sound equalization. One should always interrogate himself about the nature of the problem he is listening to. The “sweep and boost” technique often found on the internet seems like a bad strategy as it does not help to identify the problem and it does not improve the ear of the mixer.

The post Tech Tips – How to use an EQ appeared first on FLUX:: Immersive.

In this article we will try to summarize the common practice of knowledge on compression and compressors.

Understanding the dynamic range

Before tackling down the usage for compression, we have to explain the dynamic range of a signal. It is the difference between the loudest peak and the noise floor. We usually like to have a good amount of dynamic to leave the musical intention intact and the noise inaudible.

It’s a fact that some signals have too much dynamic variation, very common when dealing with a voice. There’s large variations in levels when we talk or sing, and it is even worse at a few inches from a microphone. Historically, the only solution to smooth out the unwanted dynamic, is either to put the microphone further away, or to ride the fader up and down in sync with the performance.

Automating the Console’s Faders

The origin of compression came with the idea to automate the process of fader riding, basically, a compressor is a device that lowers the volume when it gets too loud. By doing so, it reduces the dynamic range of the signal, hopefully by removing only the unwanted variations. Devices of this kind started to appear for the first time sometime back in the 1940s.

It is important to note that a compressor does actually make the signal quieter, until you decide to compensate for the gain reduction. We can also safely conclude that a compressor reduces the loud part of the signal (and not the opposite).

Where to use a compressor ?

Compressors can be used at many points of the signal path, on single channels, on instrument buses, or even directly on the master bus.

At this point, having an effective routing solution can drastically simplify the question where. One simple solution is to sum everything that makes sense together. For example, on drums, if we have two mics on the kick, two mics on the snare and a pair of overhead, we could sum the two kick microphones in one bus, do the same for the two snare channels, and finally sum the kick bus, the snare bus, and the overhead, together inside a final drum bus.

Now, if I hear a dynamic issue on the snare, I can apply a compressor on the snare bus. If I hear another one on the microphone inside the kick, I can apply a compressor on the right channel, and so on.

Downward and Upward Compression

It is sometimes seen that a compressor is any kind of device that reduces the dynamic range of a signal. There are in fact two possibilities here; you can either lower its loudest part, or do the opposite and amplify the lowest part of a signal. This second method is much less common but can lead to very effective results. At FLUX:: we call compression everything that does downward compression, and we call it de-expansion with everything that does upward compression. You can achieve upward compression with Solera, Alchemist or Pure DExpander.

Why the need for compression?

The recording technique has a strong impact on the dynamic. The closer the microphone is to a sound source, the bigger the instantaneous dynamic of a source is. So, the heavy use of compressors in modern music is very much due to the generalization of close-miking technique. In more acoustic styles of music, like orchestral recordings, compressors are much less prominent because the recording technique deployed tends to favor microphones further away from the instruments.

Common settings on a compressor

The usual suspects found on a compressor, are these four parameters :

• The threshold (dB); when the signal is louder than the level of the threshold, the compressor starts to reduce the gain.
  
• The ratio; which determines the strength of the compressor. A 4:1 ratio means that when the signal exceeds the threshold by 4 dB, the output signal only sees an increase of 1 dB.
  
• The attack time, which determines how long the compressor takes to reach the set ratio once the signal exceeds the threshold.
  
• The release time, which determines how long the compressor takes to return to 0 dB of gain reduction once the signal goes under the threshold.

Faster time constant produces more harmonic distortion, while slower ones are more transparent but also less efficient.

There are more parameters that can be discussed about compressors, these will be tackled in another article.

Should a compressor be inaudible?

There is a common perception that a well-set compressor should be inaudible. While it certainly has some truth, it is also very misleading for newcomers.

How could we be satisfied with a mixing procedure if we can’t hear it? Of course we need to hear the effect of a compressor, otherwise, we should just remove it from the audio processing chain!

The rule of thumb is that, as long as we want a transparent dynamic management of an instrument, we should only hear the proper dynamic processing without the artifact of compression.

Fifty shades of Compressors

Despite the myriad of compressor models, we can easily distinguish four main usages for a compressor :

• Reduce the crest/peak of a signal (peak compression)
  
• Reduce the mean level of variation (RMS compression)
  
• Creating a “glue” effect on a bus (Bus compression)
  
• Using compression as an effect

For peak compression, we want to use compressors that have a fast behavior. In other words, we want them to be quite sensitive to transients, to be able to catch them.

⚠️ Peaks are amplified with closer microphone placement

For RMS compression, we are looking for the opposite behavior. We prefer compressors with a “slow” behavior. This type of compressors are not very sensitive to transients, so they are best suited to work on the global variation of the signal.

For bus compression, we usually prefer faster compressors too, but which are also capable of being gentle with the signal. As for special effects, there are really not many guidelines as long as the sonic result is enjoyable. Usually, the dirtier the compressor gets, the funnier it is.

Feedback VS feedforward

Before talking about feedback or feedforward compressors, we should look at a simple diagram of a compressor.

There are two main blocks in a compressor; a detection block, also called sidechain, and a processing block, also called gain reduction circuit.

If the sidechain is fed by the input signal, then the compressor is said to be in feedforward mode. If the sidechain is fed by the output signal, then the compressor is said to be in feedback mode.

⚠️ In feedback mode, the feedback loop usually starts after the make-up gain stage. So the output volume has a strong impact on the compression!

Analogue compressors are generally designed in feedback, because it is an easier way to build them. The feedback loop also introduces some kind of retroaction which tends to limit overcompression.

Feedforward is technically considered as better, because the quantity of gain reduction does not affect the gain reduction itself. They are more predictable and sometimes easier to handle, they can also easily over compress the signal.

Compressor Reactivity and RMS Window

We have used the words fast and slow to describe the behavior of a compressor above. These words do not really relate to the attack or release time, but much more to the RMS window (or RMS size).

The RMS size is how smoothed the signal is in the detection circuit of the compressor. A short RMS size (5-10 ms) will produce a compressor very sensitive to crest and transient. A long RMS size (40 ms and above) will produce a compressor less sensible to peak and more adequate to follow the global level variation of the signal.

This RMS size has a very strong impact on the sonic characteristic of a compressor.

Program dependent compression

We often encounter the term program dependent to qualify a compressor. This means that the nature of the input signal will alter some parameters of the compressor.

Pretty much all analogue compressors are program dependent. Their release time can vary a lot depending on the input signal, or, their ratio can be greater as the input signal voltage grows, etc. It’s really more a design constraint than a feature, but it happens to be quite pleasant, sonically speaking.

On the other hand, it is very easy to design a very predictable compressor in the digital domain. But sometimes, these designs can sound a bit dull compared to their analogue counterparts. This is why most modern digital compressors also implement many program dependent features.

Most FLUX:: compressors allow the user to set the knee size (the knee makes the ratio dependent on the input level) or to have an automatic release based on the input signal crest factor. Even the ratio can be made dependent on the crest factor! And if the FLUX:: compressor you own doesn’t let you set all of this manually, you will find a mode selector, like on Evo Channel or Evo Comp, that changes all these settings under the hood for you.

FET, Optical, VCA, Vari-Mu, what is it all about?

There are different ways to build an analogue compressor. Each of these buzzwords refer to particular technologies of gain reduction circuit.

The first technology known to build compressors is the so-called Vari-Mu. They involved a tube as a voltage-controlled gain reduction device. These kinds of compressors tend to have time constant on the slower side and a soft knee. So the hotter the input signal is, the harder the compression is.

The natural evolution of tube compressors was to replace the tube with a field-effect-transistor once they have become available. FET compressors are generally closer to a hard knee while still having a bit of a progression between no compression and full ratio. They also allow for a much quicker time constant. Their main drawback (or joy, depending on your goal) is their high harmonic distortion level. They tend to be more appropriate for peak compression.

The FET compressors were rapidly replaced by VCA. While all previous technologies mentioned are built around voltage controlled amplifiers, VCA are analogue processors specifically designed for signal amplification. They usually offer a versatile range of control at a low harmonic distortion cost. They also have the nice advantage of being very small and have allowed mixing desks of the 80s to have one compressor per channel.

Optical compressors appeared in the 50s. It’s based on an electronic optic cell. They have the reputation to be quite slow and clean compressors. The release time is very program dependent. This unit also has a rather soft knee and a frequency-dependent ratio. They are more suited for RMS compression.

One should be careful at characterizing the usage of one component to determine the sonic nature of the whole device. We should never forget that in such complex systems, the topology of the circuit is very important too.

Parallel processing

We can often find settings that allow blending the unaffected signal (dry) and the compressed signal (wet) together. It is called parallel compression. The idea is to set a very aggressive peak compression and to blend it back with the original signal to recover some of the natural dynamic.

This is a very colorful and effective processing, which tends to bring low-level information in front of the mix. If you struggle to find the right settings, try a compressor with a short RMS size, with a very short attack (< 3ms) and release (< 60ms) time. Then, lower the threshold so that a good amount of signal will be processed. Increase the ratio up to taste. It will sound like a lot, that’s the goal! We will recover some air by bringing the dry signal in.

Advantages of digital compressors

We have talked extensively about analogue compressors, but they aren’t that often seen nowadays. Like many other types of processing, many analogue processing has moved to digital counterparts, first for cost-effectiveness reasons, and also because of simpler workflows.

In digital sound processing, there are far fewer constraints on what we can do compared to the analogue world. It is then easy to build compressors with many settings (user accessible or not), like FLUX:: plug-ins.

This makes digital compressors way more versatile than their analogue counterparts. It sometimes makes them more difficult to learn, but realistically, you could replace a whole collection of compressors with just Evo Comp or Evo Channel, for example. In fact, it’s not very difficult to imitate famous analogue compressors.

There are also some things that can’t (or can very rarely) be done with analogue compressors. For example, a look ahead option allows delaying the signal in the processor section and can create a zero attack time while keeping a smooth envelope (and a low distortion). Such an option is common on digital compressors.

⚠️ Lookahead also add latency in the whole audio system. Be careful, depending on what you are doing!

TL;DR

Compressors are audio processors that are designed to reduce the dynamic of a signal. Most of the time they lower the signal when it goes above a certain threshold (downward compression). Sometimes, they can also amplify the signal when it goes below a certain threshold (upward compression).

There are basically two main usages for compressors, peak and RMS management. For peak compression, we prefer compressors with fast attack, fast release and short RMS windows. For RMS compression we prefer compressors with slower attack and release and longer RMS windows. The first categories often relate to FET or VCA compressors, while the second one relates more with Opto and Tube compressors.

Digital compressors have the advantage of being very flexible and can be adapted to pretty much any mixing situation. They can be both very clean and predictable, or very dirty and dependent on the input signal. One full featured one can replace a whole collection of compression devices. They also have the strong advantage of looking ahead, which allows for clean zero attack time.

The post Tech Tips – How to use a Compressor appeared first on FLUX:: Immersive.

If you are in a hurry, skip down to the TL;DR section of this article to get a summary, as well as a video explanation of the subject.

The very final step

You for sure got this by now; a limiter should be at the very last stage that the signal you are processing is passing through, the ideal place being at the very end of the master bus.

If you are not looking for a huge loudness in your mix, one single limiter will be more than enough. Otherwise, doing all the gain reduction necessary to get a very loud mix with only one limiter is often problematic. This is where many mastering engineers are using multiple limiters and process the peak reduction little by little.

There’s a similar concept to this in the Elixir called Stage, which is exactly like having several Elixir instances in a chain. For example, with Stage set to 4, it is like having four Elixirs chained in a row, and the gain reduction is then applied uniformly between all the stages based on the threshold value.

The best practice while processing a mix with a limiter, is to do it at a constant level. If your limiter automatically compensates for the volume loss of the crest being cut off, it will be more difficult to understand if the limiter is working too hard. So, while you are adjusting the limiter, always compensate with the output level to get the same loudness as the input signal. Once you are satisfied with the result, you can then remove the attenuation.

For example, using the Elixir limiter, if you have set an input gain of 5 dB, lower the threshold with 4 dB and activate the Make Up option, you then would have to lower the output gain to 9 dB to match the input level.

Loudness is better

If you want even more loudness from your mix, you may want to try the following tricks:

• Having the right balance between tracks helps tremendously. It’s a good thing to examine this first, it can eventually be solved by a stem mastering approach.
• You may need to reduce or remove the channel link of your limiter to avoid over limiting.
• Manage the low end of your mix. The more bass heavy the mix is, the harder it will be to get high loudness. Here, an EQ and multiband compressor will be your friends.
• Some parallel compression with a look ahead to avoid peak amplification can help you to increase the RMS level of the mix. We have two presets built on this idea in the Syrah compressor, called Parallel Enhancer Loud, and Parallel Enhancer Soft.
  

• You could also try to combine different types of limiters, for example, insert a multi-band limiter followed by a single band one. We also have a preset designed for this use case in the Alchemist plug-in called 5-bands: Limiter.
  

But be careful to not end up becoming a loudness war casualty, remember that loudness is not a necessity. Loudness may seem fun at first glance, but it will quickly damage everything that has been meticulously created at the recording and mixing stage.

Immersive audio and limiting

If you are doing immersive audio work, you may be wondering how to best apply limiting processing on content with more than two channels.

When working with classic surround formats like 5.1, 7.1, or even Dolby Atmos beds (5.1.2, 7.1.4, etc.), you will need a limiter that can handle as many channels as there are present in the surround bus. The Elixir plug-in is designed for this, and offers the possibility to process up to 16 channels simultaneously.

In this case, the channel link feature becomes useful. If you have spent time creating an immersive audio sound scene, you don’t want to arm it at the mastering stage. But engaging the channel link sometimes creates over compression. The Elixir features a dynamic mode that will process transients just like there is no channel link but the rest of the processing will be applied identically to all of them.

In case you are dealing with ambisonic streams, then you should always have all the channels linked together without any optimization of any kind (no dynamic mode for Elixir!). It is due to the fact that ambisonic channels are not mapped to any particular speaker, and making gain reduction on only some of them can really harm the sound stage after the decoding. Thanks to Elixir’s 16 channels, it can handle ambisonic streams up to the third order. But be careful after the decoding stage, it may generate audio peaks that cross the threshold.

When mixing in Dolby Atmos, you are dealing with an object-based mixing. Here the conversation becomes complex, because there is, to this day, no easy way to master an object-oriented mix. If you really want to have a final limiter, you will have to only mix with beds. The main issue is that it does limit you to 7.1.2 format. Otherwise, you can try to limit on the object directly, but it will be time consuming and heavy on the processor.

TL;DR

A limiter is used last in the effect chain. If another plug-in is inserted after, there’s a big risk that the level guarantee of the limiter is compromised.

Originally, only a limiter was used on the master, but sound engineers notice that chaining multiple limiters could lead to a more transparent result. With Elixir it is the equivalent of using the stage control.

Modern mastering techniques tend to favor stem processing. Multiple files are sent to mastering, each one of them corresponding to a main bus in the mixing session. Limiting can be applied directly on this bus.

For immersive content, a multi-channel limiter such as Elixir can be used on different kinds of bus size, from quadraphonic to 3rd order ambisonic. For channel-based bus (quadraphonic, 5.1, 7.1.2, etc.), channel-link control helps to find compromise between preservation of sound localization and over-processing. When dealing with ambisonic streams, the channel link should always be on (100% and no auto mode for Elixir).

When dealing with Dolby Atmos, limiting can be used on buses and on objects, but there is no easy way, for now, to link parameters to simplify the workflow.

The post How to Use a Limiter, Part 3 – Advanced processing and Dolby Atmos mastering appeared first on FLUX:: Immersive.

If you are in a hurry, skip down to the TL;DR section of this article to get a summary, as well as a video explanation of the subject.

Definition

A limiter is a very particular kind of device, bridging the gap between more conventional compression processing and saturation. There is a common definition stating that a limiter is a compressor with a ratio above 10:1. This means that if your input signal exceeds the threshold by 10 dB, there will only be 1 dB above the threshold at the output.

But this definition is not satisfactory for the kind of limiter we use at the end of the mastering chain, more commonly known as a True Peak Limiter.

True peak limiting means that at the very moment where a sample has a value superior to the threshold of a limiter, it will be caught by it. You can think of it as a compressor with an attack and an RMS window equal to zero.

Now, True Peak Limiters also have an infinite ratio, which is also a very important point; an audio sample cannot exceed the threshold. Think of it as a kind of warranty implied by a mastering limiter. If you have set the threshold to -1 dBTP, the audio signal will never go beyond this value. This is why a limiter could also be seen as a saturation processor, because it hard clips the input signal. Theoretically, you could replace a limiter by a clipper to get the same kind of warranty, but the sonic results will most probably be problematic and quite undesirable.

Family picture

Let’s look at what happens to very simple signals when we send them through a limiter. We will first look at the spectrum analysis of a sine wave at a frequency of 440 Hz, with the FLUX:: Elixir limiter on and off. The oscillator generates the tone at a -6 dBTP level, and Elixir has its threshold set at -9 dBTP. So, we should see a perfect -3 dB of gain reduction when Elixir is on.

Maybe you are wondering if there is any difference between the two previous pictures. And yes, there is a small 3 dB difference between the two peaks, which means that we did not add any saturation in the process.

Now, to make it a bit closer to sound we will have to handle with limiters, let’s modulate the amplitude of the sine wave. For this we have simply added a tremolo with a frequency of 4 Hz. It goes from unity gain to -inf dB.

Now, we can see that there are some additional frequencies here. These are added by the fact that the limiter is engaged and disengaged by the amplitude modulation, and its envelope adds some harmonic distortion. What should be kept in mind is that the harder the peak is, the more saturation will be added.

If you use them right, you could get a light version (none True Peak) of Elixir using FLUX:: Alchemist, Solera or Pure Compressor plugins. engage the infinite ratio option, set the delay to the same value as the attack, then play with the release and hold time to get the desired result.

Smoothing the clipping

To prevent and reduce any distortion added by a limiter, we use the envelope very much like a compressor.

Attack settings

Didn’t we say that a limiter has an attack of 0 ms? Well, not exactly. What we want to be sure of is that no sample can exceed the threshold. Using an attack of 0 ms is a solution but it also generates additional saturation that we want to avoid. So what could we do about it ? This is where lookahead comes in handy.

A lookahead, as the name implies, allows the algorithm to look ahead, before the signal. So, if we know in advance when the signal will pass the threshold, we could then manage to open the envelope before that happens.

Remember, because it is still, unfortunately, impossible to go back in time, lookahead will add latency to the signal.

Another way to understand it is to look at a block diagram of a limiter.

There is a detection circuit that will tell when the signal passes the threshold. In a traditional compressor this moment will trigger the envelope applied by the processing block. So, in this regard, the processing in a limiter is always kind of late. Now, the lookahead is a simple delay at the input of the processing stage.

This attack time allows for a softer clipping of the signal. It is not often seen as a parameter on the user parameter, but almost all modern True Peak limiters have this hidden under the hood. In Elixir, the attack time also depends on the input signal, to achieve a more musical result.

Release settings

The release time is more straightforward to understand than the attack time, as it is the same thing as in a compressor. The release time is the time for a limiter to completely stop processing the signal once the signal goes back below the threshold. It has a strong impact on the quality of a limiter.

• Set it fast to get a snappy result, with a more saturated character
• Set it slow to get a softer result, with a more compressed or pumping character

As for the attack time, the release in Elixir is dependent on the input signal.

Is True-Peak really True-Peak ?

A True-Peak limiter will always guarantee that you never exceed its threshold. At least, as long as you never do any kind of sample rate conversion after the processing!

Remember, in a digital audio workstation (DAW), we work with a digital representation of sound. To do so, we have sampled the audio signal at a certain sample rate (44.1 kHz, 48 kHz, 96 kHz, 192 kHz, etc.). So it is possible that the original signal had, between two samples, a value of higher value. After a resampling, this value may appear and generate a value above the threshold of the limiter. This phenomenon is known as intersample peak.

To prevent this effect, many limiters use oversampling. It is often hidden under the name intersample peak detection. Using oversampling will increase the resolution of the limiter and prevent intersample peaks from passing through. In Elixir, the oversampling only happens in the detection algorithm, while always being in sync with the processing algorithm.

TL;DR

A limiter is a dynamics processor. It bridges the gap between compression and saturation. A mastering grade limiter is characterized by an infinite ratio and a true-peak detection. This guarantees that a signal will never exceed the threshold of the limiter.

Because a limiter has a very strong behavior in regard to the input signal, it can generate distortion. To prevent it as much as possible, plug-in constructors use a complex envelope strategy involving looking ahead with attack time, and often give the user a way to control the release time.

Alas, oversampling is often used in limiters to prevent intersample peaks from passing through the limiter.

Next article in this series
Part 3 – Advanced processing and Dolby Atmos mastering

The post How to Use a Limiter, Part 2 – Limiter Theory – Knowing your tools appeared first on FLUX:: Immersive.

In this first article, in a series of three, we will have a very basic look at limiters to help the less experienced to sort out what’s going on. In the next article we will go much deeper, so stay tuned!

If you are in a hurry, skip down to the TL;DR section of this article to get a summary, as well as a video explanation of the subject.

What is a True-Peak limiter, and are there different types of limiters?

The most common definition of a limiter, is to consider a limiter being a compressor with a ratio superior to 10:1. During this series of articles, we will assess that what we call a limiter is a True-Peak limiter, which has more constraints than the previously mentioned type of limiter.

The limiters we are interested in here, the true-peak limiters, are dynamic tools specifically designed to reduce audio crest, very much like a safety guard preventing any clipping from the digital to analog stage. Thanks to the peak reduction it is possible to use such a limiter to increase the loudness of the input signal. 

Why the need for limiting?

The very last process of music production is to set the overall level, or loudness, of a mix. Limiting is the only safe way to amplify the loudness of a mix, but it comes at a cost.

A limiter will cut the crest of the signal to create some headroom to allow amplifying the rest of the signal. The cost of limiting is a loss of dynamic and an additional distortion.

The true-peak level is the actual level of the samples, or of the waveform if you prefer. The loudness is closer to our sound perception and smooth out quick sound variation because they do not matter that much in our perception of sound loudness.

Using a limiter also provides the guarantee to never clip the digital-to-analogue stage. This is a very important safeguard and explains why there is always a limiter at the end of the chain, even if it doesn’t do much.

Limiting will always come at the cost of more saturation added to the signal, but in a very much more transparent way than just cranking the output gain and clip the converters. Clipping the converter is considered as a technical error (even if some popular master does clips at the converting stage).

When is there a need for limiting ?

If you want to make a mix louder without clipping the digital-to-analogue converter way beyond the red, you will need a limiter. The limiter will reduce the peak and provide you with headroom to amplify the whole gain without clipping the output stage.

Limiting is also often needed to conform a mix to certain norms. For example, most music streaming platforms will refuse a mix with a true-peak level higher than -1 dBTP.

Is limiting mandatory ?

Limiting is mandatory in the sense that a mix should never exceed 0 dBFS. So using a limiter with a threshold at 0 dBFS will always prevent that from happening. Most of the time, the different target platforms (streaming, broadcast, etc.) ask for mixes that do not exceed -1 dBFS.

Increasing the loudness of a mix is never mandatory. Maybe we will create a debate here, but loudness in music production is very much an aesthetic decision. So, to continue with these controversial topics; a louder mix does not sound better than a quiet one. Actually, it sounds less dynamic and more distorted. Also, there are no norms in music diffusion. Each and every platform has its way to handle the loudness of submitted audio files. 

For example, we often encounter the idea that a good deliverable should have a loudness of -14 LUFS-I with a true peak never exceeding -1 dBTP . This value comes mainly from the Spotify guidelines. But, it is not entirely exact, as Spotify offers different loudness targets for their customers. There is a loud (-11 LUFS-I), normal (-14 LUFS-I) and quiet mode (-19 LUFS-I). Apple has recently moved from -16 LUFS-I to -18 LUFS-I and before 2022, YouTube normalized loudness at -12 LUFS-I (now -14 LUFS-I). So which one should we choose? The common consensus is around -14 LUFS-I because it covers the biggest user base.

Then, what happens with a file that is above the target? If a file is submitted with a loudness target above the recommendation of the platform, the file volume will be simply dropped by the number of dB necessary to match the recommendation. So the process is transparent to what you’ve mixed and mastered.

If a file is lower than the target, most of the streaming services do nothing about it. The file will simply be quieter than the other one. YouTube used to be an exception before 2022, but Spotify in loud mode will limit the content to match the -11 LUFS-I target.

So how do we handle this mess? It seems there are three possible solutions.

• Follow the most common recommendation, aka Spotify (-14 LUFS at -1 dBTP)
• Align on the loudest one to make sure that no processing will affect your work (at the detriment of the dynamic range)
• … Don’t care about it?

Actually, the last point is the one defended by the author. Loudness and more importantly dynamic range is not only a technical thing, it is also an aesthetic choice. Some genres of music have built their aesthetic on very compressed and saturated sound, where others want to have all the accessible dynamics.

As a general guide, we will simply assume that it is a best practice to never exceed -1 dBTP. Also, it is preferred to have the loudest peak of a program, or a song, to always hit this -1 dBTP target.

TL;DR

Mastering limiters are designed to reduce the crest of a mix and allow it to increase its loudness. Their true-peak characteristic allows them to never let a sample cross the threshold.

Limiting should always be used to prevent a mix from clipping the digital-to-analogue converter. However, due to the many different loudness targets found in the streaming services, it is difficult, if not impossible, to “go-to” recommendation as for the loudness of a track. It seems to be more an aesthetic choice than a technical one, at least, in the music industry.

Appendix

dB ? dBFS ? LUFS ? What is it all about ?

There are quite a few acronyms and concepts to explain around sound pressure level and how it is measured. Because sound is a mechanical wave, the primary way to measure the sound pressure level is to measure how the pressure evolves in a space.

First, the relation between sound pressure, and how we experience the sound level, is not linear. For example, when the sound pressure is doubled, we do not perceive a sound twice as loud. In fact, to have a sense for a sound being twice as loud, we need to multiply the pressure by ten. This is why we express the sound pressure level in decibels, which is a logarithmic scale that is much closer to our perception. When the sound pressure is doubled, there is a gain of +3 dB. When the pressure is multiplied by ten, there is a gain of +10 dB.

Depending on the field of interest, there are many different units built around the decibel scale. The one that is used in digital sound is the dBFS, or decibel full scale. In the digital domain, sound is represented by samples, whose amplitude can take absolute values between 0 and 1. The number of actual values that a sample can take between 0 and 1 is defined by the quantification (16 bits, 24 bits, etc.). But this is a linear scale, and thus, it does not correspond to our perception of sound. The dBFS solves this problem. A value of 1 in linear corresponds to 0 dBFS, a value of 0 in linear corresponds to -inf in dBFS (-96 dB at 16 bits, -144 at 24 bits, etc.).

Now that we have a scale that behaves closely to our perception, we need to find a way to measure sound loudness. Here, a peak measurement (the value of each actual sample in digital sound) is not a good candidate, because fast sound variation in volume does not matter that much in how we perceive loudness. Also, the frequency has a strong impact on how loud a sound seems. This is why engineers have proposed the loudness unit.

There are different time windows for the loudness measurement : momentary, short-term, long and integrated, which correspond to the following citation from the EBU Tech 3341:

1. The momentary loudness uses a sliding rectangular time window of length 0.4 s. The measurement is not gated.

2. The short-term loudness uses a sliding rectangular time window of length 3 s. The measurement is not gated. The update rate for ‘live meters’ shall be at least 10 Hz.

3. The integrated loudness uses gating as described in ITU-R BS.1770-4. The update rate for ‘live meters’ shall be at least 1 Hz.

In the music industry, it is the integrated value that is used as a reference for streaming services.

Next article in this series
Part 2 – Limiter Theory – Knowing your tools

The post How to Use a Limiter, Part 1 – True Peak limiting and Loudness processing appeared first on FLUX:: Immersive.

As mentioned in different articles, when using audio devices to route to/from SPAT Revolution, Pro Tools is handling the needed delay compensation based on your routing / plugin usage. That being said, when you are extracting the audio from the SPAT plugin Local Audio Path (LAP), the delay compensation is not taken into consideration in regards to all the other objects you have in the session. In the case of Pro Tools, the delay compensation happens down the line between these tracks and the final bus they feed.

The problem is simple. ​You are inserting a SPAT send plugin in LAP mode on a strip after other plugins that may introduce latency. To avoid this, the SPAT Revolution 22.9 update includes a solution to allow the user to report the latency of the strip (in the plugin) and then have SPAT Revolution do the required latency compensation.

Time for a little operation – Delay compensation mechanism in SPAT Revolution!

In the latest plugin interface, a new delay Input delay field is available in the SPAT Send plugin. It provides the ability to report the latency of this audio object in samples. Once declared, when these audio sources will be connected in SPAT Revolution,  a delay compensation mechanism will apply the delay needed to each of the object input in SPAT Revolution to ensure that they are being aligned as they would be within a typical DAW Routing.

Although this operation is manual, it simply means reporting the delay information of the track itself to the field dedicated to it in the plugin. Below are three (3) SPAT object aux tracks,, one with a delay occurred because a plugin causing latency. You can simply open the user interface and report the track delay, 

Need more information on Pro Tools integration? It can be found in the Pro Tools section of the  SPAT Revolution User Guide.

The post Reporting latency for delay compensation in SPAT Revolution appeared first on FLUX:: Immersive.

In a Previous Tech Article, we’ve covered the basics of integrating SPAT Revolution into the Pro Tools environment. At the base was the use of the SPAT Revolution send plugin and the Local Audio Path (LAP) mode to route your audio to your SPAT Revolution rendering engine. 

As the session and routing requirements grow, a few things must be kept in mind. Keeping things organized in the session and adopting an object-based workflow becomes ideal. 

Many advantages come by adopting an object track-based workflow and taking advantage of the routing folder (Pro Tools 2020.3 and above)  and the newest routing possibilities if you have the later Pro Tools 2022.09 (New AUX I/O). An article on Pro Tools 2022.9 AUX I/O covers this part of the subject.

Below are two examples of routing, the first one using the Local Audio Path mechanism, the second using actual audio devices.

Using routing folders (aux tracks) as your SPAT send objects

• Allows to sum multiple audio tracks to the same audio object
• Those auxes are becoming your SPAT Revolution objects
• Using the routing folders (auxes) simplify routing, organization and keep things clean.
• Pro tools being pre-fader insert only, using the SPAT send plugin (LAP Mode) to extract audio directly on audio tracks lose the ability to automate the fader functions.

The newest templates in the Pro Tools section of the SPAT Revolution User Guide are adopting exactly this workflow. Both using external audio bridging solutions or the Local Audio Path (LAP) feature of the SPAT Send plugin.

As mentioned above, the routing folder track is at the base an aux track. They come with the advantage that they are a folder so they keep things organized. When creating a routing folder, it does a few things for you. It creates the aux patched to a specific bus for input. In the above, this is the « SPAT Mono Object 1 » audio bus that you can route any audio track to. The key takeaway is that once done, the only thing left is to route this audio object routing folder to the desired audio output or use the SPAT send plugin with Local Audio Path on it. 

Moving your audio tracks to the routing folders

If you have audio tracks that you want to send to a routing folder, you can simply right-click on that actual track and choose the Move to function.

You can then move to any previously created routing folders or create one on the fly.

Moving an audio track to an existing routing folder or creating a new one

This is of course possible with a group of audio tracks as well, that you for example would want to “declare” as a single audio object to SPAT Revolution. The last step to assign it to the SPAT Send plugin, and either activate the local audio path (LAP), or patch the output to the desired audio bridge output.

For example,on the above animation , you have  4 audio tracks to declare as an object. Select them, right-click and choose “move to new folder” (or simply press the  Shift+Command+Option+N on a Mac or Shift+Control+Alt+N on Windows) with routing. Give it a folder name (such as your object name) and you are done.

Instantiating the SPAT Send plugin on the Routing folder 

The last step is to insert the SPAT Send plugin into the actual routing folder(s). This will first allow you to automate all the SPAT Revolution source parameters to Pro Tools for writing your automation. You can simply enable the mode in the plugin interface if you want to use the Local Audio Path (LAP) option rather then actual audio I/O devices to route to SPAT Revolution. You can always hold the CTRL key before to enable it to do it to all instances of the plugin into your session

Routing using Local Audio Path (LAP)

If you plan to use the Local Audio Path (LAP) function to route the audio to SPAT Revolution, you have to make sure to follow the proper routing of those tracks to maintain good sync.

This routing is to have all objects routed to a SPATSync bus (namely a dummy bus).  This is explained in the Pro Tools intregration to SPAT Revolution article and throughout the Pro Tools section of the SPAT Revolution User Guide. 

Ideally, you use the provided templates as a start point to understand this important routing well. It involves routing all your SPAT Objects to a single SPATSync bus and making sure the SPAT Revolution Renders return track(s) in Pro Tools, using the SPAT Return plugin, are patched to this bus as an input. With that, a good sync is well kept.

The post Using Pro Tools routing folders with SPAT Revolution appeared first on FLUX:: Immersive.