Which version of VSCO 2 CE is right for me?

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VSCO 2 CE is a powerful free chamber orchestra library, but it comes in nearly a dozen different formats, most of which are made by a variety of third parties.

Each version has strengths and weaknesses, and may provide compatibility with different software or work better in a certain workflow.

Summary

  • If you’re just getting started, either use Bigcat’s VSTi/AU Version or Orchestools 306. Both are easy and work in most DAWs (Orchestools 306 also runs on Windows as a standalone application).
  • If you have Kontakt, use Bigcat’s Sketching Chamber Orchestra or The Alpine Project. Both are very good; Alpine also uses other sources too.
  • If you like using SFZ or have ARIA or Sforzando (free), use either the Vanilla SFZ Version or the Virtual Playing Orchestra (VPO). Vanilla is unprocessed, VPO is processed and with extra samples. SFZ format is open source (literally just text files!) and documented here, if you like ‘hacking’ patches to make cool things!
  • If you have Renoise, use Oliver’s XRNI conversion.
  • If you have Sampletank 3 (compatible with free Sampletank Custom Shop) or want something that goes beyond just orchestra and into more effects, use Orchestools One.
  • If you have a Synthstrom Deluge, use the Synthstrom Deluge version.
  • If you like to work with raw .wav sample files or use a sampler not listed above, try out the Original .WAV Format, or the 256-sample pack or 50-sample pack (in the ‘Notes’ section of the former).

Natural vs. Controlled Behavior

In sampling, we conceive of the behavior of a library on the spectrum of ‘natural’ to ‘controlled’. Samples cut in a ‘natural’ way have slow, sometimes slightly irregular attack behavior, slower response, and more variable dynamics and mapping. The result of this is a very natural sound, but only when used in the way the samples “want” to be used (e.g. slow, ambient-ish music).

On the other hand, samples cut in a ‘controlled’ way are rigorously flattened out, made to behave in identical ways, so that the user can modify the response via ADSR or other controls more easily. Such samples do not sound good initially, but with some careful modulations can be almost as realistic as ‘natural’ cut samples while providing you more control in the end, allowing you to shape the sound to your needs rather than shaping your writing style to suit the samples.

This divergence may be noticed most between the ‘Vanilla SFZ Version’ and ‘Bigcat’s VSTi/AU Version’. The SFZ version has no adjustment to the samples and thus is much looser, and arguably more realistic when individual notes are played. However, Bigcat’s version has been normalized and processed to behave more consistently. This makes it sound sometimes less realistic (e.g. a finger cymbal as loud as a bass drum), but if you pick velocities carefully and use volume to modulate the sound, you can get a very good sound out of it.

More naturalistic versions are: Vanilla SFZ, Original .WAV, Deluge, XRNI
More controlled versions are: Bigcat VST, Bigcat Kontakt, VPO, Alpine, Orchestools

Contents

The only version required to have all of the samples is the Original .WAV Format. All other versions are free to incorporate parts of the library, or to include parts of other libraries.

Includes full sample set: Original .WAV Format, Vanilla SFZ, XRNI, Deluge
Includes partial sample set: Orchestools One & 306, Bigcat VST, Bigcat Kontakt
Includes 3rd party samples: VPO, Alpine

Operating System

Note: Versions may not be compatible with the latest nor future operating systems. Please visit the version page for more information.

For Windows only: Orchestools 306
Windows + MacOS: Vanilla SFZ, Bigcat VSTi/AU, Bigcat Kontakt, VPO, Alpine, XRNI, Orchestools One
Not compatible with either: Deluge
Platform Agnostic: Original .WAV

Software Requirements

Requires no software: Orchestools 306, Original .WAV
Requires only a DAW (except Protools): Bigcat VSTi/AU
Requires DAW + Sforzando (free) or other SFZ player: Vanilla SFZ, VPO
Requires DAW + Kontakt (full, not free Player!): Bigcat Kontakt, Alpine
Requires DAW + Sampletank Custom Shop (free): Orchestools One
Requires Renoise: XRNI

Requires physical Synthstorm Deluge: Deluge

The Maxims of Guerrilla Sampling

Here is a list of lessons learned through sampling on the go that may help others avoid stumbling blocks. Note that these are not as applicable to professional sampling, but rather apply to the “run-and-gun” Guerrilla sampling discussed in my other blog posts.

Equipment and Gear

  • Don’t be afraid of budget gear, but be sure you get to know it before relying on it for anything of value. If a $200 mic can do the job just as well as a $1000 one while being cheap enough you won’t care of it is dropped or wears out, use it.
  • The nature of diminishing returns means that for gear over $1000, your gains will become much smaller. It is best to keep your entire rig under $4-5K if you want to be flexible but still provide quality recordings.
  • Develop your gear with an objective in mind- low-noise, pleasant frequency response, light weight, high durability, and high flexibility. Seek unusual gear for added value to products (e.g. a cheap vintage dictation mic on Ebay means you can make a “vintage” mic position on your products).
  • The second-best is often the best because it sounds great, but costs less and will set you apart sonically.
  • Develop a setup plan and rehearse setting up your gear. If it takes you more than 10-15 minutes to set up 2-3 mics and your signal flow, you have a grave problem. 5-10 minutes is optimal. Double those numbers for 4-6 mics.
  • However, safety of gear always trumps speed. Screw mics into their baskets/mounts after screwing the mounts onto stands (optimally leave mounts screwed onto stands or use quick releases). Be cautious with quick releases- they may save time, but badly manufactured ones or careless usage of them may result on a tumbling mic (invest extra- I like the Gator quick releases at this time).
  • Speed comes from experience, not knowledge. As much as theory matters, doing things a lot matters twice as much. Rehearse your setup and takedown regularly.
  • Set up chairs/tables, interface/hardware/power, stands, mics, then cable runs in that order. This allows your hardware time to warm up/boot up, particularly important with tube preamps.
  • Avoid using the gear of others, including studios, whenever possible, unless you are familiar with the specific gear or it is well maintained. Using stands is okay, but cables is iffy (a broken cable can cause painful delays).
  • If a cable or mic stand is dead/broken, either fix it properly or throw it out. Do not keep faulty, damaged, or broken equipment, it will cause trouble, mistakes, and slow you down.

Location, Location, Location

  • Always go for a space you know well over a possibly superior space you know very little.
  • Always go for a larger/wetter space over a drier/smaller space when in doubt. It’s easy to isolate wetness, it’s hard to fix liveliness. Never pay extra for dryness.
  • Learn how to use mic patterns, dead spots, moveable barriers/gobos, etc. to isolate sounds.
  • When in doubt, ask about a sound. If someone lives there, they may know about it (e.g. refrigerators, mouse deterrents, computers, power supplies).
  • Use and be aware of your environment at all times to your advantage- a stage may have curtains, a room may have dividers, a home may have folding tables that can be upended all to resolve or reduce noise issues.

Mic Placement

  • Build your mics around your “primary array”, or main pair of mics.
  • The primary array should not typically be wider than your ‘wingspan’ if using AB.
  • The primary array should not be Blumlein or XY; near-coincident arrays are okay, as is a well-executed Mid-side if isolation is not an issue. Spaced pair and Decca tree are standard.
  • The primary array should be no closer than 6 feet and no farther than 15 feet from the sound source.
  • The primary array should be a balance of wet and dry leaning between 60-80% towards the dry.
  • If recording an ensemble, never place the primary array mic closest to your loudest instrument(s)- seek an ensemble blend from the mic position to begin with.
  • Following the primary array, design the rest of the mic placement to either (A) accent (offering the user additional colors) or (B) contrast the primary array (offering the user different directions), depending on your overall design goal.
  • Don’t put a mono mic more than 6 feet from the sound source.
  • Don’t put a close mic on something if it needs space to develop- instead place it more as a ‘spot’ or ‘accent’ mic.
  • Don’t put a mic where you wouldn’t want to sit/stand/float.
  • Never put a mic directly behind a horn unless you want it to sound like the 1970’s.
  • If you’re recording more than one player or an instrument with a large sounding area, matched pairs matter a lot less. Sometimes it may even be advantageous to mic one side of an ensemble with one mic and the other with a different one.
  • Just because something sounds good in person doesn’t mean it will sound good in the recording. Always do a test recording before proceeding.
  • Most mic placements were invented by very creative people living in archaic times with incredibly archaic technology at their disposal (a.k.a. Alan Blumlein in the 1930’s, you know, when people used 78’s and wire recorders, the condenser was 15 years old and the ribbon was 10 years old, and stereo recording was an “experimental science”). Don’t be afraid to break the mold and try something different.

Workflow

  • Just because x worked for someone else, that doesn’t mean it’s right for you. Always experiment and find the best way to do something.
  • Keep to as few programs as possible- keeping everything in Reaper, for example, is in your favor, as it seems to be the most favored platform for cutting/editing samples.
  • If the above isn’t possible, use a dedicated waveform editor such as Adobe Audition to keep the BS features low and allow for speedy and easy exporting of stems.
  • Keep your signal path as bullet-proof and simple as possible. Avoid any kind of multi-device routing, mixing boards, complicated runs, existing infrastructure on location, and borrowed equipment if able.

Studios

  • Avoid studios unless you know the owner and/or they will let you use your own gear entirely (hopefully for a significantly reduced price). The space is what matters most.
  • Don’t pay for an engineer to sit there and do nothing for hours while you record- that’s just a waste of your money and their time, not to mention, they’ll probably get bored and start working on something, which, in a poorly designed studio, could cause background noise. You honestly shouldn’t even be using their computers anyway which is more time wasted by setting up a session there, having to set up a talkback system, running back and forth, etc.
  • Avoid studio consoles, patch bays, etc. However attractive the siren’s song, they will only slow you down and provide ample fuel for Unknowns to throw a wrench in your project, not to mention lengthen and complicate the signal path.
  • Avoid studio personnel and their ‘advice’ unless you really don’t know what you are doing. They will also slow you down.

Working with Others

  • Always use working musicians/engineers over trained musicians/engineers.
  • Seek individuals who are first and foremost highly competent on their instrument, but also creative and flexible. Sometimes a less competent musician who is more creative/flexible is the better choice than a less flexible but most competent one.
  • Keep people who do good work happy and they will do better work. Find the best way to encourage each person you work with to do their best, be it humor, timeliness, or simply friendly companionship.
  • Always coach a musician from the same room. Never sit in the control room or in a corner when you should be out on the floor coaching. Sit behind the main array with direct eye contact with the musician for directions, preferably also with a direct line of sight to the interface/laptop to see check levels.
  • Never rely on other people to complete a task until they have shown their ‘true colors’. Plan for failure and delays so releases will come in a timely fashion and you will be continually surprised and pleased by expedient work.
  • Never give anyone else creative control over the project direction except the main musician (if a ‘creature feature’ library) and the UX designer.

Personal Progress

  • If you have to study something to learn it, you probably don’t know enough foundations to truly learn it properly yet. Gain more experience with the foundations first.
  • If making a choice between the conventional and unconventional of near-equal gains, always go for the unconventional. It will almost always be more rewarding.
  • Focus on developing a system rather than goals/milestones. Although it may be helpful to keep goals in mind, they tend to turn into either insurmountable defeatism or appear to others as compulsive lies when broken.
  • Delegate the tasks which you are least qualified to do, but never pass on those tasks upon which the fate of your business lies unless you are absolutely sure of the ability of the delegated.
  • Keep an eye on your competition. You may decide to do what they are doing, or go against what they are doing, and perhaps both at the same time.

The Art of Guerrilla Sampling

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Sampling Session for VSCO 2, Summer 2015

Sampling is the process of converting an analog instrument into a digital emulation using recorded “one-shots” of the real instrument in action. Guerrilla sampling is doing all that with minimum costs and maximum efficiency by using existing infrastructure and extreme mobility.

In this post, I will discuss the principles of creating samples, the economics involved, and general tips and tricks to yield the best results when working with time, space, or financial restrictions.

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VSVI Dev. Blog 7: Digital Audio Myths

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Artificial harmonic glissando

In digital audio, we think about fidelity in four domains:

  • Bit Depth (bits)
  • Sample Rate (Hz)
  • Bitrate (kbps or kb/s)
  • Channels

It’s important to understand what each of the terms means when shopping for samples and sample libraries, as some advertised features will do very little more than exponentially increase the size of the library, making it seem more valuable than it is!

Bit Depth describes just one single element of the digital audio equation: noise floor. It is the measure of bits in each sample taken. It does not have ANY role in the sound-quality of your audio, only where the noise-floor is located. With noise-shaped Dither, 16-bit audio can easily cover beyond the theoretical range of human hearing (-96 dB, or as far down as -120 dB with shaped dither). We always record and process in 24-bit for improved filter performance, but provide our instruments in 16-bit.

Why? The preexisting noise-floor, even with careful recording and even denoising is always considerably higher than the theoretical -96 dB (possibly extendable down to -120 with correct dither usage) noise-floor found in all recordings done with microphones. In fact, any sample library developer that tries to sell you samples with more than 16-bit audio that were not recorded in a fully isolated and insulated anechoic chamber is wasting their bandwidth and storage space, and your hard-drive space and time.

The best way to understand bit depth is to imagine we have a set of sine waves we’ve cut into 44100 columns (samples). At each slice (in 16-bit PCM), we pick a number between −32,768 and +32,767 (with 0 representing the middle line of the waveform) that most closely resembles the point we see on our analog arc and place our sample there. It must be an integer (0,1,2,3, etc.). If we get something very small (i.e. quiet) and boost it a bunch digitally, then we will encounter artifacts from our earlier quantization (don’t worry, you would have to be recording something at close to -40 or lower dB for this to happen). For 24 bit, we get to pick from 16,777,216 possible integer points. Therefore, smaller waveforms are possible to represent quieter waveforms and boost them digitally without encountering quantization distortion. 32 bit float is another form, using a float rather than an integer, so it can provide a decimal value. Because it is so resource expensive and the dynamic range it provides is essentially completely unnecessary (extending exponentially beyond the range of human hearing), 32-bit float is not used except for the recording of highly unpredictable sources and ultra-critical processing, and requires ultra-high-end equipment and recording conditions to generate any necessary need (most mic and preamp self-noise is far too high for 32-bit noise-floor), i.e. industrial/scientific uses. It can be useful for extremely heavy effects processing on a single signal, where repetitive quantization could result in a noise increase, but the amount of usage would require a very, very powerful computer just to function.

Sample Rate is a function of the total frequency range that can be represented in the digital audio. It is a measure of the number of times the audio signal is sampled every second. Under the Nyquist Theorem, if we accept the hearing of a young female toddler may be, at its very greatest, 20,000 Hz, a sample rate that would fully include all frequencies in this range would be 40,000 Hz (40 kHz). Add a little buffer and do a little manipulation to make synchronization with video recording easier, and voila, 44.1 kHz! In Europe, they decided to add a bit more of a buffer (no pun intended), and went with 48 kHz. We use 44.1 kHz at all point in the sampling and distribution process.

Why? A sample rate of 44.1 kHz extends beyond the maximum range of human hear (if you’re male and/or over 20, your hearing likely drops off around 17-18 kHz). Recording too much higher results in distortion on equipment (amplifiers, speakers, etc.) not designed to handle those rates, which could result in issues with our customers, aside from using enormous amounts of space. Any developer who sells samples more than 44.1 kHz that are not intended for very extensive resampling/manipulation is possibly multiplying the size of their library by 2, 3, or even 4x for NO perceivable improvement. Beware!

If we take our bit depth example from before, imagine we had 16 (or 24) rows and wanted to cut our sine waves in a different number of columns. Increasing the number of samples would mean increasing the fidelity to each sine wave. Remember, a lower sample rate means any sound greater than 1/2 the frequency of the sample rate will be lost (this is why applying an 8kHz sample rate results in a sound not dissimilar to the fidelity of 78 rpm records, at which time, recordings could only reach around 4 kHz total frequency range).

Bitrate measures the number of total bits stored every second. In lossless audio, this is Bit Depth * Sample Rate (or for 44.1/16 mono, 705.6 kbps (stereo would be 1411.2 kbps)). Bitrate only changes from the lossless measurements if a form of lossy compression is applied, such as the .ogg vorbis or .mp3 lame codecs. Lossy compression, for obvious reasons, degrades the sound quality of the audio, no matter how little you use. For .mp3, anywhere down to 320 kbps is more or less indistinguishable from uncompressed signals for most music (particularly signals without strong transients) for consumers. We do not use any lossy compression in any stage of our development process.

Why? Compression compromises the signal much more than other formats. Chances are, many customers will want a higher fidelity sample than compressed audio is capable of.

Channels describes the number of different audio streams used. Most modern audio work is recorded in stereo (2-channel), and occasionally in mono (1-channel), although recent advances in technology have led to the development of affordable ambisonic microphone arrays, capable of recording a 360-degree signal, and, with the help of a decoder, reduce it to a single 2-channel, 4-channel, 7-channel, or so on experience. We record all instruments in stereo whenever possible, and if multi-mic recording is done, occasionally used arrays of mono or stereo design to capture different angles.

How does this fit in with other digital formats, such as video?

In digital video, we think of a number of frames per second, and an amount of data per frame (for example, 30 frames per second of 720p footage (that’s 921,600 pixels per frame) is 27.65 million pixels every second. Typically the color of each pixel is expressed in 8-bit, so we would end up with  221,184,000 bits (about 27.65 MB every second, or 221,200 kbps, compared to a mere 1,411.2 kbps for 44.1/16 audio). Of course, in this example, we assume zero compression and also leave out other information that might be included in the specific codec, but it is a good way to get a feeling for the size of audio data.

How do all of the above elements fit together?

In digital audio, we comprise our recording of a series of samples (sample rate), each containing a certain number of bits (bit depth), with a certain number of channels. Multiplying these three values will give us an understanding for the total amount of data being transferred in bits (make sure you convert bits to bytes if you are concerned with storage space).

IotW 6: Shure Hercules (?)

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Although the exact model of this mic is a bit of a mystery, it sounds pretty good!

Something often neglected in sampling and recording is the less expensive side of microphones. For example, when wanting to produce a track with an early jazz sound matching the era in which it was written a bit closer, I turned to this interesting mic I picked up at a tag sale years ago, which appears to be a late model or descendant of the Shure Hercules.

Check out the results (unaltered) and a parallel stereo recording with the XY capsule of a H6. That trumpet solo at 50″ sounds straight off a 78, minus the distortion and cracks. It also worked well on another period piece.

VSVI Dev. Blog 6: Mic Usage in Sampling

“Ur Doin’ It Wrong”; Image by Geoff Kaiser

There is a fascination, in the last few years, with primarily three features of orchestral sample libraries: If the number of multisamples/RR is in two digits, if it has sampled/”live legato, and how many mixable mic positions are available. Today, I’m going to talk a bit about the latter, “multiple mixable mic positions”, as well as using microphones effectively to create an effective experience for the end user of the samples, to the point where they really do have control over the tone of the instrument.

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IotW 5: Performing for Samples

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I’m not always the one behind the mics, such as this (admittedly shaky) shot from last summer’s sampling bonanza.

If you’ve seen trailers for virtual instruments with real footage of the musicians performing, you probably see a 10-second or less clip of some cool note or just some silent close-ups while some dramatic music created using the plugin months after the original session.

In reality, sampling sessions are long, slow, borderline Zen marathons of endurance, especially when alone, as is often the case in such Guerrilla-style sampling sessions as those I often run. In which case, I either see something like the above or like the image below-

Any pianist will tell you sitting at a piano for two hours is a long time... sitting at a piano for two hours playing one note at a time waiting for each note to decay is an eternity.

Any pianist will tell you sitting at a piano for two hours is a long time… sitting at a piano for two hours playing one note at a time waiting for each note to decay is an eternity.

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IotW 4: Consorts and Cousins: A Tale of Two Trombones

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A “Bb” Tenor Trombone by C.G. Conn (Foreground) and a “G” Bass Trombone by Hawkes & Sons (Background) lounging.

The two instruments you see depicted are roughly contemporaries (the Bass Trombone is actually a little later, in the 1910’s, and English rather than American, but contemporaries they are just fine enough).

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Image of the Week 3

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Possibly early 1900’s or late 1800’s Classical Trombone Mouthpiece

I actually rescued this mouthpiece from a pile that were going to be scrapped, and boy what a find! A little polish and a great background shot for a future library is possibly born.

Some consultation with a more experienced individual points towards this mouthpiece being based on the sort of mouthpiece measurements one might find from the Classical period- large, relatively flat rim, smallish cup, somewhat sharp transition into backbore, small bore.

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Image of the Week 2

Interior close-up of a grand piano sampled in December, 2014.

Interior close-up of a grand piano sampled in December, 2014.

The above shot comes from a brief sampling session in which I completed a basic sampling of a grand piano for the upcoming VSCO 2 and other applications. It’s a bit dusty, but sounds pretty nice!