@vinco said:

how do you get the right balance of the instruments and automation if your instruments are not panned while automating in midi?

The most common way to spread phantom sources in the stereo field are:

a) Each stereo track is on two faders, one fader for the left channel, and one fader for the right channel. Each of this two single channel fader has one pan knob. This gives all options as when mixing with a console.

b) Interaural intensity differences (IID), or better called interaural level differences (ILD), are differences in the intensity of sound arriving at the two ears, and are important cues to localise sounds.

c) Interaural Time differences (ITD), are the main cues for localizing the azimuthal position of sounds. For example: even when the left channel is far louder then the right channel, but the left channel is delayed by a few macroseconds against the right channel, then we localize the sound comming from the right, because the sound from the right channel arrives a few macroseconds earlier at our ear.

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@Another User said:

But I'd love to use your info to have a nice calibrated template session.
Something to start with that I can rely on.

I rather think using a template doesn't save much time. I never had twice the same instrumentation.

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Angelo,

I'm not an English-native speaker so sometimes the words I'm using aren't right.

When I'm automating Midi tracks, I use a combination of velocity, volume (CC7) modulation (CC1) and the fader automation on the aux return of each instrument.

So, if I understand correctly, while composing, all your instruments are fully panned in the total stereo (100% left - 100% right)

When I said balance, I meant the loudness of the different instruments.
I always believed (please correct me if I'm wrong) that an instrument being centered in the pan would sound louder than the same instrument being panned hard-left (for the same gain on the fader).

This is why i'm asking this question.

If your instruments are not panned appropriately when you're composing, then all your fader moves/automation would need recalibration when you're at the point of panning the instruments. (either panning in Midi or in Protools on the Aux Return)

@Angelo Clematide said:

The biggest difference when mixing with a console is, that the best hardware outgear processors are used, instead of plugins.

Of course but I thought you meant the calibration setup/workflow would be different if it's done with a external console.

Vincent

Vincent,

two questions:

1) how many fader do you have per stereo track in your virtual mixer, one stereo fader or two single channel fader?

2) do you know what "pan laws" are?

Here a good explanation what pan law is:
http://logicquicktips.blogspot.com/2006/10/laws-of-pan.html

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Until now I always used stereo faders but it is possible to use two mono faders if needed (Logic Pro).

I'm actually considering using the stereo aux in Protools as reference (instead of using the output of the virtual instrument on each track) for panning.

Why?

Vincent

@vinco said:

Until now I always used stereo faders...

Why?

Deactivate the Universal modus in Logic, then you are flexibale to chose whatever fader type you like. The numbers of faders and pan knobs double, and you are in a professional console like environment.

Faders on a console are always single channel faders, there is no such thing as stereo faders in pro audio.

Think about the consquences having two faders for each stereo track, and each of this two faders has also a pan knob. You not only can pan each side of the stereo seperately, but also change the level of each side independently, you can overgo the pan laws, even before pan laws where introduced in Logic. Further you can narrow the stereo field, as well several other techniques can be used now to place a source in the stereo field.

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That's a great tip you just gave.

I only experimented with the stereo panning PT has but I never thought of using 2 separate faders to get more flexibility.

So that's something you would do at the mixing stage, right? Not when you're composing?

Vincent

@Another User said:

Not when you're composing?

When I compose, I need all time and concentration for getting the idea into the box as fast as possible.

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@Angelo Clematide said:

That's a great tip you just gave.
That's not a tip, that's how we all work eversince.

Thanks for the training then since I didn't have that knowledge [:)]

Thanks again for being so patient.

Once you understand it, it feels obvious though!

Vincent

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Why are you quoting my post? Your last notions don't relate to what I've written whatsover... [*-)]

Hi Angelo,

there are three more questions, I have:

1. When U describe

"III. The measured loudness of an orchestra recording

The decibel values are from a recording who has dynamics from ppp (as soft as possible) to fff (as loud as possible):

ff = -12.5 dB RMS (as loud as possible)*
mf = -18.0 dB RMS (normal loudness, not loud)
mp = -20.0 dB RMS (normal loudness, not soft)
pp = -27.0 dB RMS (soft)
ppp = -48.0 dBFS Peak (as soft as possible)"

These values are before setting your track faders and sample patches to -6dB, right?

2. Since using reverb and track-positioning, my instruments change their original volume. So this is why it would be helpfull to know the dynamic range of e.g. the VI-16 after setting the volume faders and inserting reverb and other plugins.

To avoid an answer like "It depens on the song...", how are the dynamic ranges of your instruments (ariving at master output) in the Sunrise_master?

3. Why do I change the single tracks an not the sum-output by 6 dB?

@Felix Bartelt said:

1. When U describe

"III. The measured loudness of an orchestra recording

The decibel values are from a recording who has dynamics from ppp (as soft as possible) to fff (as loud as possible):

ff = -12.5 dB RMS (as loud as possible)
mf = -18.0 dB RMS (normal loudness, not loud)
mp = -20.0 dB RMS (normal loudness, not soft)
pp = -27.0 dB RMS (soft)
ppp = -48.0 dBFS Peak (as soft as possible)

These values are before setting your track faders and sample patches to -6dB, right?

No!

The values in the loudness table are the circa minimum to maximum dynamic. This values are as measured on a finished stereo master who is mixed with wide range dynamics. Most orchestral classical recordings have an approximately min/max dynamic range as in this loudness table.

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@Felix Bartelt said:

2. Since using reverb and track-positioning, my instruments change their original volume.

I don't use any additional 3D positioning tool. I checked them once and didn’t like the sound.

I make all the 3D positioning and phantom source in the stereo field "by hand" as in the old days, and without any additional plugin.

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@Felix Bartelt said:

So this is why it would be helpfull to know the dynamic range of e.g. the VI-16 after setting the volume faders and inserting reverb and other plugins.

To measure every single instrument seperately, that would to be too much work, work who would not produce meaningful and helpful results for the mixer.

The most basic and important thing for the mixer are:

a) that the balance of all instrument track combine to a good stereo master.
b) that at the master stereo fader no artefacts are produced during printing a mix, i.e. dB 0+ values or severe clipping.
c) The stereo master has the required/wanted musical dynamics.

This may sound a little dry, and as if mixing is a boring process, but everybody who mixes knows that it is an extension of composing, quasi the last arranging.

In my case, the wanted dynamics are always “wide range,” similar to the dynamics a recording made in a concert hall or studio has. Whenever a stereo master has to be volume maximized, i.e. for tv-spots, then we do that in a second work process, this simply because we want to own a wide range dynamic master, otherwise we could process the additional maximizing plugin chain right on the stereo master fader and print the mix mastered.

My proposed system for mixing a wide range stereo master is nothing new, but a description of what every mixing engineer work consist of since decade.

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@Felix Bartelth said:

...", how are the dynamic ranges of your instruments (ariving at master output) in the Sunrise_master?

The "Sunrise" stereo master is the reference track after which the loudness table in question was made. The loudness values in the table accurately represent what the loudnesses are at the stereo master fader while printing the mix.

I chosed this track, because the overall dynamics can be easely measured and also visually be followed on the meter and wave editor, this since the recording has a gradually raising loudness from the beginning to the end.

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@Felix Bartelth said:

To avoid an answer like "It depens on the song…

No, it doesn’t depend on a single song!

I depends on an overarching dynamic concept. A dynamic concept who permitts to record and mix song after song, for years and years. And where on a future compact disc, any song, no matter if produced years ago or just yesterday, will fit between any song, and this without level adjustment or loudness modification.

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The original discussion thread is closed, so I'll post here and you can move it if you like.

To me, the challenge is not total orchestra dynamics. The dynamics of the full orchestra can be compressed (for film scores) and is subject to taste. What is much more interesting to me is relative dynamics inside the orchestra. I've searched for a chart that shows dB SPL for individual instruments at several dynamic levels. For example,

Trumpet, pp: 51
Trumpet, mf: 75
Trumpet, f: 93
Oboe, f: 77
...etc.

I would assume that some university grad student must have done this, but I could not find a paper after several google searches. Simply a list of instruments playing forte (perhaps muted and unmuted) would be useful. The instruments could be then be balanced in our templates. There's also the concern of relative volume versus pitch (a low forte C on trumpet will be softer than forte C an octave or two above), but this is hopefully built into the sample and outside our control in VI.

"Use your ears" is a valid response to this question, but empirical data is always nice to start from. Perhaps VSL could provide this data on their stage if it doesn't already exist. [:)]

Thomas J Bergersen talked about this relative balance in his column for Virtual Instruments mag. Once the relative balance is complete, then the combined dynamics can be studied.

@Another User said:

"Use your ears" is a valid response to this question...

That's what is left to do when it comes to what you call "relativ dynamics," if I understand that terminology right.

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That's the reason why we need to balance the instrument levels (and panning) ourselves. Even a chart that's within 10dB would be helpful.