New track, Precipitous Sublimation, on Soundcloud
New track of cinematic music – Precipitous Sublimation – now on Soundcloud.
Fletcher named UCLA Extension Outstanding Instructor
Fletcher has been named one of 2015’s Outstanding Instructors in UCLA Extension’s Department of the Arts.
New Plugins and Sample Libraries of Note
Hive by u-heU-he is a small developer from Berlin run by Urs Heckmann that makes software synths and effects processors. Among the software synths that u-he has developed are Zebra2, a wireless modular synth capable of an impressive array of sounds and enhanced by the Dark Zebra upgrade which features patches used in Han Zimmer’s Batman scores, Bazille, a modular system with patch cables that allows for extremely flexible patching, and Diva, a virtual analog synth with a high degree of sonic authenticity.
U-he’s latest release is a software synth called Hive. Hive is built around a relatively simple concept – a dual oscillator engine that allows the user to layer two voices in the tradition of synthesizers like the Yamaha CS80. But don’t let the sound of that fool you. Hive is capable of a wide variety of sounds ranging from lush pads to fat basses to electronic rhythmic sequences. In fact, the simplicity may be welcome, as Hive is significantly easier to program than Zebra2 or Bazille and has a smaller CPU footprint. Nevertheless, the sound quality is great and the synth comes with many presets that cover a wide range of synthetic sounds.
Media composers will probably find Zebra2 and Bazille the most intriguing of u-he’s synths due to their flexibility in programming and wide sonic range, however you really you can’t go wrong with any plug-in in the u-he product line as they all sound phenomenal. Hive is great choice for someone without much experience in synth programming who wants to get good sounding results quickly. For those with a bit more background in synthesis, Zebra2 and Bazille offer endless possibilities for new sounds.
Signal by Output
LA based Output has recently released a unique sample library in Kontakt format called Signal. Signal was conceived with the idea that rhythmic pulses are an important facet in music making and a library focusing on pulsing sounds was lacking in the world of sample libraries.
Signal comes over 700 presets ranging from aggressive distorted sounds to ambient ethereal ones. Patch selection can be filtered by keywords so if you need an epic organic sound with a triplet feel you can select on the appropriate adjective to select patches matching those descriptions, a great time saver for those working under fast deadlines. The sounds tend to be electronic in timbre, though there are some acoustic sources such as felt piano, harp and marimba, and if you have find a sound you like the feel of it is easy to change the sound source.
Each preset features four macro sliders to quickly change the basic characteristics of the patch and the sliders can be easily assigned to MIDI faders for real time morphing. For those looking for more control, all parameters and pulses can be adjusted by clicking on the “pulse engines” tab. This brings up an intuitively laid out interface that is easy to navigate, which is a good thing as Output doesn’t come with any documentation. The lack of documentation can make some more advanced editing a bit frustrating.
Signal is a great library for any media composer who has a need for pulses, arpeggiators and morphable rhythmic sounds in their music, which probably covers most of us. There is no danger in trying it out and risking disappointment since Ouput offers a 14-day money back guarantee for anyone who is unhappy with Signal.
Omnisphere 2 by Spectrasonics
If I had a desert island synth plugin, it would be Omnisphere. Omnisphere has everything you could ask for in a synthesizer: a huge library with over 8000 presets covering a wide variety of genres, incredible programming power, and a beautifully laid user interface the makes programming and editing a joy. Now they have made it even better.
Omnisphere 2 adds 4500 new patches and soundsources, over 400 new DSP waveforms, new arpeggiator features, an enhanced interface and the ability to use your own audio files as a soundsource, to name but a few. The new patches and soundsources alone would be worth it, but with all the other features this upgrade is a must.
For those who don’t own the original Omnisphere, this plugin is an essential part of a media composer’s arsenal. The sounds are stellar and cover a huge range of styles from hardcore electronic to very organic. The user interface is easy to navigate and it is simple to make the presets unique with programming features such as the “orb”, which allows a user with no programming background to drag a circle around to change the sound in interesting ways. For those who want to dive in deeper, the synth engine in Omnisphere is extremely powerful and customizable. Omnisphere 2 just makes one of the best synths on the market that much better.
Nugen MasterCheck and ISL2
I don’t really like to master my own music given the choice, but sometimes it is necessary to do some DIY mastering when sending tracks out for demos, submitting to music libraries or releasing music online. For a long time, the general mastering principle has been to make a track sound as loud as possible through compression because listeners will perceive a louder track as sounding better when compared to other pieces within the same genre. That paradigm appears to be changing with the advent of streaming music services.
Streaming services such as Spotify, iTunes Radio and YouTube are all using loudness normalization to match volume levels between different tracks in their catalogs. The basic idea is that their software analyzes each track and matches the volume level so that if a listener is playing a Beethoven symphony and then switches to a dance track, they won’t have to adjust the volume control on their playback device, the software will do it for them. This is a big deal because it means there is no longer any value in overcompressing tracks to make them sound louder. In fact, overcompressed tracks will sound lifeless in comparison to tracks with a wider dynamic range. I suspect this approach will be adopted across all music streaming services in the not too distant future.
Nugen Audio’s MasterCheck is a plugin you insert on the master buss of your DAW to monitor your mix levels and allows you to determine how your mix will sound when played back through various streaming formats. MasterCheck has several displays which provide useful information for mixing for targeted delivery platforms. The LKFS (Loudness, K-weighted, relative to full scale) meter displays the peak level of the track allowing one to easily monitor mix levels, while the PLR (Peak to Loudness ratio) meter monitors the peak level of a track relative to normalization which helps to give a sense of the overall dynamic content of the mix.
MasterCheck includes a number of presets so you can hear what your mix will sound like played back on various streaming services. By clicking the “offset to match” button, you can hear the results of Spotify, iTunes Radio and others’ loudness normalization on your mix in real time. You can even compare your mix to other tracks using the external reference feature.
For those in need of a high quality meter to assist with the mastering process, it is hard to beat MasterCheck.
Originally published in The Score, Volume XXX
Practical Approaches to Making Sounds That Evolve
Sustained sounds need subtle changes and movement to remain interesting to our ear. When an instrument like a violin sustains a note, there are subtle pitch and timbral variations imparted by the player that keep the note from remaining static. Electronic sound sources, on the other hand, don’t change or evolve without some help from parameters within the synth. Textures that are static sound lifeless and flat, particularly when mixed against dialogue and sound effects and played back on a medium like television under less than ideal listening conditions. Giving your synth sounds movement and evolving timbres will make them sound richer and add more depth to your mixes.
Let’s take a look at some techniques for making a sound evolve.
1. Use multiple LFOs set at different rates to modulate instrument parameters.
Low frequency oscillators are key to creating timbral movement because they can provide cyclical change to a parameter. LFOs are often used to simulate vibrato and tremolo by modulating pitch and volume but can modulate virtually any parameter in many plugins. A classic way to provide sonic evolution is to route the LFO to the filter cutoff, set the LFO shape to a sine wave for a smooth transition and the LFO frequency to a low number (1 Hz or lower) so the change happens slowly. This makes the sound get brighter and duller over time as the LFO moves through its cycle.
To make complex evolving sounds you need multiple LFOs to modulate discrete parameters at different rates. Many synths only have a single LFO, but instruments such as Omnisphere, Massive, Zebra2 and Logic’s Alchemy (brand new in version 10.2) feature up to six LFOs that can modulate almost any parameter on the instrument. By setting different rates for each LFO, you create a sound that is constantly changing and non repetitive since each parameter cycles through its changes at a different speed. It isn’t necessary to provide a lot of modulation (usually represented as depth or amount) as subtle changes are often most effective on pads and sustaining tones such as drones. Good candidates for modulation are filter cutoff, resonance, panning, timbre shift, and volume.
2. Use envelopes in addition to LFOs.
Envelopes modify the attack, decay, sustain and release of a parameter. The limitation of an envelope is that, unlike an LFO, it doesn’t repeat, so when an envelope reaches its sustain level, it no longer modifies the parameter until the note ends and the envelope goes through its release portion.
Synth envelopes have traditionally been used to modify the amplitude or filter. In many plugins, envelopes, like LFOs, can be routed to modulate virtually any parameter. For pad sounds, envelopes are useful for movement as their shapes are more complex than the cyclical LFO. Omnisphere and Alchemy are examples of synths that feature very long envelope times. These plugins can have attacks, decays and releases of up 20 seconds, which translates to 60 seconds for an envelope to go through its entire cycle. For a drone that needs to sustain for minutes on end this may not be enough for a constantly evolving sound, but for many sustained tones those lengths are more than sufficient.
3. Automate parameters to precisely control the way the sound changes over time.
A plugin’s parameters can be automated with MIDI continuous controllers and via track automation. Most plugins allow you to assign a MIDI continuous controller by right clicking on the parameter you wish to automate, clicking learn MIDI CC and moving a knob or fader that sends that MIDI CC on your MIDI controller. You can then record or draw in any fader moves you wish to make. Most plugins’ parameters can also be automated using your DAW’s track automation if you simply want to draw the automation in with your mouse. Automation is a great choice for sonic variation if your instrument doesn’t provide many modulation options or if you want very precise control over the way the parameters change over the course of the cue.
4. Use more than one sound source to create evolving textures.
Synths that use more than one sound source are great for evolving textures. By changing their relative levels over time, you can create an ever-shifting soundscape. A cool effect is to apply independent LFOs to pan the sound sources at different speeds. I generally like to use subtle panning, as a small amount gives gentle movement without calling too much attention to itself.
Orbit is an example of a great sounding Kontakt instrument built around the idea of using multiple sound sources to create shifting textures. Orbit uses four sound sources that it cycles between. Each source can be filtered, panned and tuned independently and the movement between sources creates a constantly evolving timbre.
Alchemy presents another method for switching between its sound sources and parameters. In Alchemy’s performance section there are eight boxes representing different parameter settings for a given patch. The boxes are bounded by a blue rectangle that can be dragged smoothly from one box to another, incrementally changing preset parameters and morphing the sound. You can add these movements to a recorded MIDI track by setting your sequencer to overdub MIDI and recording the mouse movements.
5. Use sampled sound sources rather than electronic oscillators.
Synths that can play samples are useful for creating rich textures since samples of acoustic sounds are more timbrally complex than electronic oscillators. Omnisphere, Alchemy and Izotope’s Iris 2 are examples of plugins that can load samples as sources. You can quickly make a sound your own by finding a preset you like and swapping out the sound source. I often use this method in Omnisphere when I want to create a unique sustain sound. I create a basic sustaining sound, use LFOs and other modifiers to morph the sound as described above, then replace the sound sources to create a number of variations of my original sound.
6. Use insert effects to create variation.
If you have a patch you like on a synth that doesn’t feature many modulation options, insert effects can be used to create variation. I often will give my pads a subtle pulsing effect by using tremolo. My favorite plugin for this is Tremolator by SoundToys. Tremolator has a number of shapes you can use and it syncs to tempo. I add a small amount of depth to give the sound some movement. A similar effect can be achieved with Logic’s Tremolo plugin, which can create either mono or stereo tremolo effects depending where you set the phase settings.
Autofilter plugins can be used to filter a sound with an envelope or LFO that syncs to tempo. Your DAW probably has one as part of its stock arsenal of effects. I like to use FilterFreak2 by SoundToys. FilterFreak2 has two filters that can run independently of one another. The filters sound amazing and can be used to create long slow changes by unsyncing the plugin tempo from the DAW’s tempo, setting it to its lowest setting of 30 bpm and letting it cycle through 16 bars. At this setting it takes the filter over two minutes to complete its cycle.
As media composers, our work is deadline driven. While I love getting lost in the sonic possibilities available to me through plugins, I rarely have the time to program a unique sound from scratch in the middle of a project. A lot of presets sound great, but may not evolve in the way that you need them to in the context of your track. These techniques will help give your sounds movement and keep them from sounding dull and lifeless. Small changes can go a long way to keeping a sound interesting over time and add richness and depth to the electronic textures you use in your music.
Originally published in THE SCORE magazine, Volume XXX Number FOUR.
Immersive Reverb for Orchestral Samples
Reverb is a beautiful thing. It creates a spatial context for sound and serves as a unifying force for tracks recorded in disparate spaces. While pop mixing allows for great variety and choice in reverb settings, orchestral musicians play together in a large concert hall and this can present some challenges when mimicking this musical setting with samples. When used well, the reverb acts as a unifier for getting all the samples to sit together in their own virtual space, even when using libraries created by different sample manufacturers.
Like many composers, I own a number of orchestral sample libraries.
I have my favorites for different sections of the orchestra and I find that some libraries are more suited for certain styles of music and orchestration. The libraries are most flexible when they are recorded with a minimal or adjustable amount of reverb. When I apply my favorite reverb setting, it can serve as a sonic “glue” for my virtual orchestra and make it sound as if all the instruments are playing in the same hall together.
The prototypical way of using reverb in a digital audio workstation is to create an auxiliary track, insert a reverb plugin and use post-fader sending to send level from the track that you want reverberated. Post-fader means that the signal gets sent to the bus after it hits the fader. If you lower the volume of the track, the level of reverb diminishes by the same amount. To adjust the amount of reverb you want on your track, you change the send level that is bussed to the reverb. Post-fader sending works great in musical situations where we want to hear a dry, unaffected signal mixed with the reverb. In most mix scenarios, we hear more of the dry than the reverberated signal.
Post-fader sending is how we hear reverb applied in most pop situations. The instruments are generally recorded with a microphone close to the source and reverb is added to give a sense of spatial depth using the technique described above. If one moves a microphone further from a source, more of the reverberation of the space that the source is recorded in will be captured on the recording. Close miking gives an engineer a lot of mixing options because we can add reverb to a sound to give it depth but we can’t remove it if it is found on the recording.
This approach of adding reverb to a dry recording works quite well in mixes where we are used to hearing a lot close miked instruments. Guitars, drums, keyboards and vocals are typically miked close to the source. But this approach doesn’t sound natural on orchestral samples. The reason is that, as listeners, we don’t hear an orchestral section from a close vantage point. We listen from a distance and, as a result, the sound of the room plays a big role in the sound of the sections. Simply applying more send level using post-fader sending doesn’t really work because we still hear too much dry, close miked sound.
A workaround for this is to use a type of sending that is generally reserved for headphone mixes. Pre-fader sending sends the signal down the bus to the reverb before it hits volume fader. This allows you to create a track where the reverb is more prevalent then the dry signal, because the fader now controls the level of the dry signal. As you lower the fader, you decrease the level of the dry signal but leave the reverberated signal intact since the bus level is unaffected by the change to the fader. This, in effect, reverses the function of the send and the fader from the post-fader scenario. The send now sets the level of reverberated signal and the fader sets the ratio of dry/wet signal.
This approach works well for orchestral instruments because if an orchestral sample is recorded with minimal room sound I can essentially put it in the hall by removing a greater amount of dry signal than would be possible with post-fader sending. If I want one section to sound further away from the listener than another, I just lower the fader for the more distant section. The result is that the tracks sound as though they are placed in the hall rather than the hall being added to the sound.
The downside to pre-fader sending is that since the fader no longer controls volume you cannot use the automation lane of the track to automate volume. With virtual instruments (VIs) there is a simple workaround – you simply use a MIDI continuous controller (either cc7 or cc11) to change the volume level of the VI. Audio tracks don’t respond to MIDI controllers, but you can to insert a plugin that can change the gain of a track as a substitute for your missing volume control. Most DAWs have a built in plugin designed for this purpose. Logic’s “Gain” and Pro Tools’ “Trim” are examples of plugins that perform this function.
I find that the best reverb plugins to use for orchestral samples are convolution reverbs. Convolution reverbs use impulse responses (IRs) to “sample” acoustic spaces and, as such, are more natural sounding than algorithmic reverbs. My favorite convolution reverb is Altiverb by Audioease, but there are convolution reverbs made by other manufacturers and the principle is the same. Many DAWs include their own convolution reverbs such as “Space Designer” which is found in Logic. These plugins come with IRs of different spaces and give you the ability to load impulse responses of your own.
An interesting result of using a convolution reverb with pre-fader sending is that it has a noticeable effect on the timbral quality of a sampled instrument. Pre-fader sending makes the choice of reverb instrumental to the sound of the VI itself. I’ve been amazed at how running string samples through a good sounding IR can warm up the sound and take off the harsh edges that I often hear in string libraries. It can improve the sound of the samples without the need for corrective EQ.
Many of the newer orchestral libraries allow one to adjust the level of recorded reverb through the control of different microphone positions. The library manufacturers provide samples recorded with microphones at different distances from the section. These are often named something like close, stage and room and you can mix in different amounts of the mic position within the sampler’s interface. If you have a library that is recorded in this manner (Spitfire Audio and Cinesamples are sample manufacturers that take this approach) you don’t need to apply additional reverb as the VI instrument has all the reverb control than one might need. However, each mic position requires additional samples so loading all mic positions for a preset can take a considerable amount of RAM. For this reason, I usually just load the stage mic setting and add additional reverb using pre or post-fader sending, depending on the sound I am trying to get.
If you are mixing libraries that have reverb in their recording with libraries that are recorded dry, it is important that the your reverb matches the one on the recording so that the instruments sound like they are in same space. Choosing a similar type room and matching the reverb decay works quite effectively in most cases. You can figure out the decay time by listening to a percussive sound and timing how long it takes for the reverb tail to die away.
Film music mixers often add some high end digital reverb when mixing orchestras for film. Lexicons or Bricastis might be used to add some additional depth and sheen to the sound. This can simulated with great effect on an orchestral mockup by adding a convolution reverb with an IR (IRs can be made from audio equipment like reverbs and delays in addition to acoustic spaces) of one these units to the master bus. IRs of high end reverbs can be found for purchase on the internet if your convolution reverb doesn’t have a preset for the unit you like. You can then add a little Lexicon shimmer to your mix (I find a 10% mix ratio is a good starting point) to get that film score sound.
Getting samples to sit together in a virtual acoustic space is a key to creating convincing sounding orchestral music electronically. Pre-fader sending can help to create more depth for the orchestral samples in your mix and integrate them convincingly into your reverb’s hall. Used in conjunction with a good convolution reverb, I have found this approach to yield more convincing results with my mockups than conventional post-fader sending.
Originally published in THE SCORE magazine, Volume XXX Number ONE.
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