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Topic: Natural vs Beefed up Sound

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  1. #1

    Natural vs Beefed up Sound

    One thing I do mostly in my music that I compose is leave volumes as what they are. I don't add limiters, maximizes, or other volume boosting effects.

    I have done a bit of research, and a lot of the market is about LOUD. But has anyone ever compared tracks to tracks? By that I mean some with the quieter recordings such as Schindler's List's main theme. Those who own the soundtrack will know what I am talking about. The solo violin is quiet as compared to big hit scores in these days.

    When you listen to the theme from Schindler's List, everything is so natural, it is quiet but it feels more... well... I can't explain it well but real.

    I have noticed when I use maximizing plugins they tend to disrupt one thing or another as compared to leaving it silent and natural.

    Does anyone have these same thought or experiences?

  2. #2

    Re: Natural vs Beefed up Sound

    When dealing with orchestral instruments, and unless you are designing a hybrid track, natural sound, every time.

    Others will have their own opinions but everything you will read about orchestral mixing will state to lay off as much as you can from compressors and certain other plugins. I find though they are often talking about mixing a real orchestra. Samples need some help. I use a glue compressor on the master bus to very lightly solidify the mix. Some people would probably say never do this.

    The factors involved are your samples quality, and if they need EQing, a compressor to boost the sound, and other plugins. Your natural dynamics of samples across different libraries will likely require a lot of volume tweaking. Once you have that, tiny adjustments on compression (percussion for example) can help your instruments be heard.

    I recommend a limiter on the master bus, always. I just try to avoid hitting the ceiling. But I don't want to adjust the dynamics of a track or section just because one tiny part spiked over 0db - so a limiter prevents that happening.

    I use an orchestral mastering suite to give the tracks some lift, expand the stereo field, some multiband compression if required.

    Even after all this time I'm still experimenting. I'm not necessarily doing it correctly, because I hear better orchestral mixes than my own, and finding out how is tough. People keep these secrets very close to their chests. Some hand their stems over to professionals.

    I look at my wave files after an export. You should be able to clearly see the natural dynamics, quiet sections, loud, and no horrible 'block' wave files. I despise tracks that are produced that way. Even if they are of the genre, like dance music.
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  3. #3

    Re: Natural vs Beefed up Sound

    It's a good, perpetually interesting subject for those of us into home recording, Richard.

    Graham has given you an excellent, very helpful reply. It's a condensation of the things he's talked about on various threads over time, and I recommend you take his advice to heart.

    All of us are always trying to up the quality of our recordings, Graham, and there's no question that at this point, what you're producing is Way up the ladder of quality. I think the procedures you're using are really good adaptations of the standard advice to not use much, if any, plugins while mixing orchestral recordings. Exactly as you point out, an engineer working with tracks of a live orchestra is dealing with one thing, but we're dealing with something different - our sample libraries which will be in need of a lot of "tweaking" to get the best blend. To use a subtle amount of compression to help smooth things out, and to limit the top DB level is extremely helpful, as you point out.

    I completely agree that virtual orchestra recordings can benefit from the judicious use of plugins, despite the common admonition to not use any mastering FX. All of us have heard recordings that aren't serving the compositions very well, with a weak sounding, limp sound which dulls the impact of the music.

    Quote Originally Posted by sururick View Post
    ...I have done a bit of research, and a lot of the market is about LOUD. But has anyone ever compared tracks to tracks? By that I mean some with the quieter recordings such as Schindler's List's main theme. Those who own the soundtrack will know what I am talking about. The solo violin is quiet as compared to big hit scores in these days.

    When you listen to the theme from Schindler's List, everything is so natural, it is quiet but it feels more... well... I can't explain it well but real.

    I have noticed when I use maximizing plugins they tend to disrupt one thing or another as compared to leaving it silent and natural...
    As I started this reply with, it's a big subject, Richard. There is a problem nowadays with the "loudness wars" which result in pop recordings which have the "block" or brick wave file profile Graham's talking about. If you look at the audio display for a modern pop song in an audio editor, instead of a naturally undulating rise and fall of volume, you'll see the screen almost completely filled up - the signals are all pushed up to the very top.

    A really unfortunate thing is that when you watch/listen to a lot of YouTube videos, the sound will be almost unbearably loud and compressed, because the amateurs making the videos simply strap on compressors and limiters, use default settings, and think they're making the sound be like commercial recordings - but their results are grotesquely squashed.

    But here's an important thing to understand about loudness. Even though we don't want to produce those solid brick wave files that are unwavering in their decibel level, we Do still want to have our loudest peaks go up to the same highest possible level (0db) as those overly compressed recordings. To be more accurate, we usually want to aim for around -0.3 DBs.

    Here's a screenshot I posted some time ago. The top light blue image shows a section of a Forum member's recording as it was originally posted. The dynamic range, the variety of volume levels, looks fine, but the highest levels are way below the 0db mark, probably around -15.db. That's a lot of potential volume wasted.



    The lower, black image shows the same section after I use Normalization to bring the volume up.

    Normalization keeps the relative volumes all exactly the same, it just brings the loudest peaks up to the maximum. The quietest sections still maintain their relationship to the loudest. You can set what the maximum is, and it should be just a fraction under 0db.

    That track could have been improved even more. See that peak in the middle of the bottom shot? It probably didn't need to be that loud. That one peak could have been reduced automatically by a Limiter, or hand edited to be lower, without any sonic damage to the track. That would have left the potential volume of the entire track even higher, because the loudest peak would have been lower than that one.

    You really need to finish work on your tracks in a dedicated audio editor like Sound Forge that I use, or the free Audacity. There's always improvements to be made to your export, maximizing the volume being the primary one.

    The advantage of raising the wave file to its maximum volume is that it will then be compatible with the audio levels the listener is accustomed to. He shouldn't need to be reaching for his playback volume knob to turn your track up or down in order for it to be in the pocket of a professional recording's level.

    But doing that kind of work on your master isn't engaging in the "loudness wars." That refers to compressing audio files so the dynamic range is very limited, with an almost constant volume for all sections. THAT you don't want to do, as Graham explained. You simply want to use the entire available db range, but keep the dynamic range intact, and in orchestral music, that can be Very wide.

    Your example of the "Schindler's List" music is perfect. That violin solo is sweet and soft, as it should be. But:

    Volume is relative. Something can only be perceived as quiet in comparison to other sections of the same piece which are loud. The loudest sections of the "List" soundtrack will go up almost to 0dbs, and then soft sections like this will be way down in the range of -20dbs, even -40dbs. It's the variety of volume levels which give that soundtrack, and any good recording, it's dramatic, natural and dynamically changing sound.

    So, don't be afraid of using the kinds of tools Graham talked about - just learn to use them with some finesse. You don't want to make horrible solid brick recordings, but you also don't want to make wimpy tracks too low in volume. It's out of control peaks you want to also be concerned about - Nothing's quite as ugly as the sound of digital clipping!

    Randy

  4. #4

    Re: Natural vs Beefed up Sound

    I may be going a little off topic, but it got me thinking about RELATIVE volume (and other) levels before you get to the final mix.

    One thing that I wonder about is why sample libraries are recorded at different volume levels. For example, I am using the last pre-Aria versions of GPO and JABB and Kontakt 2 (rather than the Kontakt 2 Player they shipped with). When I open a GPO instrument, the default maximum volume is louder in Kontakt 2 than the JABB instruments. I have to manually match the values with every JABB instruments -- otherwise, it's a lot more tweaking in the individual tracks to make the JABB trumpets louder than the GPO harps, and a lot less range to work with on the dynamics for each.

    I usually start with the max set for every instrument in Kontakt or the other players. I set the cc 7 value for each track to 100, then change it for individual tracks to compensate for differences in the overall loudness of some instruments when compared to others (for example brass vs strings), their "position" in the virtual orchestra (degree from front to back and L or R) and any weirdness in the sample recordings themselves. My goal is to get everything sounding balanced (i.e. like a real orchestra would). Then I set the levels for the dynamics of each piece using cc 1 in Garritan libraries and velocity values in the others.

    I prefer this method to a real or virtual mixing board for several reasons. Since mixing is highly subjective and will sound slightly different to me every time I hear it, having specific numeric values for volumes (and volume ranges) helps me not to veer too far from what my ideal relative values should be. It also keeps me on top of the peculiarities of instruments whose character markedly changes at different volumes. (For example, the French Horn can sound a lot brassier in louder passages than the normal, mellow, sound that I usually associate with the horn. I like to be able to decide that myself, rather than relying on the defaults or x faded articulations.)

    With regard to reverb, I think there was a whole other thread about that some time ago. I think most of what was said can be applied to other effects. Basically, I use them when I want to "correct" the sound of individual instruments to get them to blend better. For example, Garritan instruments are very dry, so they all need some help. Since other libraries are less flexible, I am usually trying to get the GPO and JABB instruments match my other samples.

    As for compression and limiting, I pretty much agree with what has been said in this thread. I would just add that pop music production practices date back to the days when listening devices (we called them record players and tape decks back then) were primitive by today's standards. A transistor radio could only decently reproduce mid-ranges, so record producers tended to roll off the highs and lows (at least for "singles). But toward the end of that era, audiophile recordings were being released with extended dynamic ranges (i.e. they assumed that if you were paying the extra bucks you had a decent turntable which wouldn't skip when the kettle drums rattled). But I feel your intended playback device and environment should still dictate how you tailor the sound. For example, I think you can go with much softer and louder passages in movie soundtracks (where you know even the softest sounds are going to tremendously amplified) than you would if your intended listener is going to be someone with headphones and an MP3 player (or else he will be constantly having to turn the volume up or down).

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  5. #5

    Re: Natural vs Beefed up Sound

    I saw your great post on this thread earlier, ejr - Now I have time to add a few things in response:

    Quote Originally Posted by ejr View Post
    ...One thing that I wonder about is why sample libraries are recorded at different volume levels. For example, I am using the last pre-Aria versions of GPO and JABB and Kontakt 2 (rather than the Kontakt 2 Player they shipped with). When I open a GPO instrument, the default maximum volume is louder in Kontakt 2 than the JABB instruments...I set the cc 7 value...for individual tracks to compensate for differences in the overall loudness of some instruments...I prefer this method to a real or virtual mixing board...
    The default volume levels certainly do vary wildly, certainly between Libraries from different companies, but even among the Garritan Libraries, and even inside just one particular Library. All those samples are from so many different recording sessions, with extremely different circumstances, that's one factor. And the programmers can't just aim for the samples to all play at full (0db) value - obviously they would peak out as soon as they're played with any volume control. It makes me think of how assembling samples for a virtual instrument is an art form, not a science, and so we
    have some widely differing ideas of how much of a volume range should be provided, and what default level should be used as a result.

    But nothing really accounts for how the default balances in just one Library, take GPO, for instance, aren't really useable when using more than just a few instruments. There will always be imbalances that need to be rectified. And then, for some reason, the entire COMB Library is extremely soft compared to GPO and the others. Whenever I use COMB instruments, I have to push their volumes way up, and pull everything else way down.

    No matter what templating I'll do once in awhile, I find the question of balance is really unique for each piece. I much prefer to just start fresh each time as I start the basic work of mixing, which is setting the basic relative balances. That part really doesn't take very long. It's the constant juggling of volume levels throughout a piece which is the trickiest part.

    I think you know I always bounce my MIDI tracks to audio, and so that does cause some differences in the work flow.

    You said you don't like to use a virtual mixing board, but of course using one is essential if you're doing the extra steps of working with audio rather than just staying in MIDI.

    Since I work with audio, that's the biggest reason that getting an absolutely spot-on balance between instruments in MIDI doesn't matter to me too much - I'll be doing that more specific kind of adjustment with the audio tracks.

    The virtual mixer has MIDI tracks in it too, however, and is actually easier to use than the one in ARIA. When using ARIA in a host DAW program like Sonar, the host actually usurps control over the faders in ARIA, so without some changes made in the default way the program works, the user has to use Sonar's sliders to work ARIA's sliders. And I find it much preferable. For one thing, all of the MIDI volume sliders for the string of ARIA instances are all next to each other on one screen, instead of all tucked away in the separate ARIA GUIs.

    Quote Originally Posted by ejr View Post
    ...With regard to reverb...I use (it) when I want to "correct" the sound of individual instruments to get them to blend better. For example, Garritan instruments are very dry, so they all need some help...
    Well, having the Garritan instruments dry was a very specific decision and concept, a very important one. The point is to leave it up to the user in what space each project is placed in. Digital reverberation is the equivalent to natural reverberation in venues.

    By having the instruments dry, the user has complete flexibility over this crucial part of his recordings. He can place his group of virtual musicians in a parlor, recital hall, concert hall, opera house, or at the bottom of the Grand Canyon or in outer space.

    Reverb isn't "correcting" the individual instruments, its placing them in an environment, without which, they sound completely unnatural, since we never hear any sound in real life without some reverberation coloring its tone, even in the smallest of rooms. The Garritan instruments don't need "help" in this regard - they're blank slates, ready for the user to bring to life with all the tools available, including placing the instruments in virtual spaces.

    Getting back to the original topic, you had a very important point in your opening paragraph:

    Quote Originally Posted by ejr View Post
    ...it got me thinking about RELATIVE volume...
    You're right. That's the key. It's what I was trying to explain in my previous post. We want our recordings to have wide dynamic ranges, unlike pop music which has its range purposely narrowed in an extreme way. And the highest peaks in both kinds of recordings, pop and orchestral, still aim to the optimum level of 0db - If we leave space between our loudest points and the highest potential volume, then we're creating all sorts of unnecessary problems for our recordings. But from start to finish, it's all about how the various levels relate to each other. Our soft sections can only be perceived as Soft in comparison to our sections which are Loud. And even the most even, peaceful piece will have its peaks and valleys. The peaks just need to be at 0, and those valleys will still be soft - as compared to rock and pop which has a dynamic range of Loud and Even louder.

    Randy

  6. #6

    Re: Natural vs Beefed up Sound

    I don't think there needs to be a versus mentality when it comes to something like this. Some things will sound better natural and others will sound better compressed and louder. There is no need to limit yourself, you should let the song/your personal taste decide how the recording sounds. To me it's like saying "I refuse to write for woodwinds because they aren't powerful enough". I say use everything in the toolbox and if it sounds good it is good. If you aren't afraid to write something atonal you shouldn't be afraid to use a limiter or a compressor. If it sounds bad just take it off.

  7. #7
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    Re: Natural vs Beefed up Sound

    Sorry to have been absent from the Forum for so long, but, other important things had to be done.

    I was pleased to see this topic being discussed and the suggestions are very helpful. I found that some GPO instruments, such as the triangle, were too soft to be used well. Bouncing the track in place while normalizing the track solved this. Are there potential problems with this practice that I need to know?

    I am working on a piece that has a unison horn passage that needs to start and end medium loud but approach the max volume for a horn section in the middle. I used 4 GPO horns plus the FF overlay. When each instrument is played alone or with the overlay the effect is correct. When all of the horns are played together the effect is also correct. If all the horns are played together with the rest of the orchestra the passage goes all the way to +6 DB. Fading the horns to a lower level makes the start and end of the passage too soft. I'm considering grouping the horn section as a sub mix and putting some compression on the grouped sound to try to maintain the correct timbres while reducing the peak and bringing up the beginning and end of the passage. Can this work? Is there a better way to go?

    Thanks to all of the people who have responded to this thread.

    Norman

  8. #8

    Re: Natural vs Beefed up Sound

    Hi, Norman - Excellent to see you here, as it always is.

    Your example of needing to compensate for the low volume of the triangle is a perfect example of how it can be difficult to balance levels when working only in the MIDI realm. To get the triangle up where you need it calls for pushing its slider all the way to the top in ARIA, and pulling everything else down to at least half-- and that's not a very satisfactory solution.

    Bouncing the triangle track and normalizing is a fine solution. You may know I always bounce all my MIDI tracks to audio in order to have more volume control, just as in your example. I actually don't normalize a bounce very often, having bounced the tracks at healthy volumes to begin with, and then pushing the audio volume sliders up into that head room if needs be, pulling other sliders back etc - But normalizing is a fast, easy solution, and after you've normalized, then you actually need to pull the fader down, as I'm sure you've noticed - unless you want a GIGANTIC triangle that sounds like the dinner bell on The Ponderosa--hehe.

    Your other challenge, with the horns, is another perfect example of how tricky mixing/balancing instruments can be:

    Quote Originally Posted by jandjnelson View Post
    ...I am working on a piece that has a unison horn passage that needs to start and end medium loud but approach the max volume for a horn section in the middle...If all the horns are played together with the rest of the orchestra the passage goes all the way to +6 DB. Fading the horns to a lower level makes the start and end of the passage too soft. I'm considering grouping the horn section as a sub mix and putting some compression on the grouped sound to try to maintain the correct timbres while reducing the peak and bringing up the beginning and end of the passage. Can this work? Is there a better way to go?...
    And this particular challenge illustrates the truth in the statement, "Mixing is an art, not a science." Because of the particular combination of many factors, making that horn passage do what you want it to do in the mix is unique and calls for experimentation - as you've already started doing. Here are some thoughts:

    Using a compressor to bring up the beginning and end and reduce the level of the middle would basically be eradicating the CC1/11 volume work you did on the horn tracks. The compressor would also be coloring the sound to some degree - So I would only try the compressor as a very last resort. But see what I mean, that you'd be using that plugin to basically erase the work you already did with MIDI volume control.

    What you said about the horn passage making the mix go into the red when all the tracks are playing - that is always the challenge, dealing with the increases in volume as the accumulation of sound from all tracks is summed. So when first setting up your mix, you need to pull all your audio track faders down to half, and use that as a starting point. Go to the loudest peak in the piece, and adjust the faders so that those loudest moments reach up into the volume meter's head room, but without the peak indicator showing you've peaked out. That's your highest position for the faders. They'll all be at slightly different levels of course, since you will have wanted to improve the balance between all tracks.

    Next up - You just have to start using track automation, if you're not already doing that. Record the track automation in real time, or hand insert nodes to control each track's volume throughout a mix. When that horn passage comes up, you'll need to bring all the other tracks down a bit.

    And that brings up the next thing - You need to start using group buses, if you're not already doing that. With each section of the orchestra having its own bus, then you have just a few faders to deal with when you need to compensate like I just described. Automate the faders down a bit for the other sections, then have them come up again when needed.

    You may also need to automate the group bus for the brass or horns, bring the fader up slightly for the start, down a bit for the middle, up again for the ending. That will have the same basic result as using a compressor, but without interfering with the sound so much, and with you having more control.

    AND - there's also the very important issue of your arrangement. It's very possible that you have too many things competing for the same frequencies during that horn passage, and they're partially canceling out the sound of the horns. So, you may need to try taking out some instruments at that point - your arrangement simply may need some adjustments.

    Randy

  9. #9

    Re: Natural vs Beefed up Sound

    Wow! This quite bit of information to read. I want to thank all of you for your responses, tips, and help. As for the master bus, what I usually do is lower all the main volume nobs to low volume levels and raise the volume on my speakers. Once I am done with the composition and mix it down I then normalize it to expand the volume, this is how I know I will never reach that digital clipping. It really is hard for me to make since of the EQ in placing it into my music. When I compose, my ears like it, but again its just me. Perhaps if I learn to use the EQ better, but again does the style of music come into place? A lot of the stuff I make is dreamy like.

  10. #10

    Re: Natural vs Beefed up Sound

    Quote Originally Posted by sururick View Post
    ...As for the master bus, what I usually do is lower all the main volume nobs to low volume levels and raise the volume on my speakers. Once I am done with the composition and mix it down I then normalize it to expand the volume, this is how I know I will never reach that digital clipping. It really is hard for me to make sense of the EQ in placing it into my music. When I compose, my ears like it...does the style of music come into place?...
    Hi, Richard - The topic of your thread is of perennial importance in our world of making music in our home studios. And, as in the replies here, it's a big topic that encompasses many aspects of the mixing art - emphasis on "art" not "craft."

    In response to your new post, - some questions and replies:

    --I'm not understanding about lowering the "main volume knobs" and compensating by raising the volume of your speakers. Are you trying to avoid accidentally going into the red and blasting your ears out? By "main volume knobs"- do you mean the individual sliders for the tracks, the slider on your Master bus, or the sliders for your soundcard/interface?-- If the latter - you don't want to ever move the volume control for your interface. Only at optimum, 0DB, can you get a playback level that's showing you exactly what's happening in your mix.

    This procedure of lowering volumes but raising the speaker volume is self cancelling - You would have the same sum volume if you would just leave your speakers where they should be, at optimum level, and only use your software's faders. Even before you start mixing, and you're working on the music itself, you may as well get at least an approximate volume for the project early on.

    Normalization doesn't guarantee that you won't have digital clipping. If there's digital clipping distortion in the track, it will still be there once you've Normalized. As the Audacity information page about Normalizing says, "Normalize does not allow clipping above 0 dB"--and that's right - the process is bringing the loudest peaks up to the value you set in the Normalize control, and it can't be higher than 0, but if there's distortion in the track, that will still be there when you're done.

    Audacity's info page also agrees with what I always do and have recommended many times - You don't want to Normalize to a full 100%, which is up to 0 dB - You want to aim for -3. When you make your MP3 copy of that 2-track master, it's much less likely to peak out. MP3 compression raises the volume slightly, so you need headroom for that. Even after you've made your MP3 at -3, it's best to run a "detect clipping" filter to see if there are any peaks in the file. If there are any, careful use of the pencil tool in an audio editor like Audacity can be used to move those peaks down - Just make sure they aren't audible, distorted clips - Those never go away once they're in a file. You can pull distortion way down in volume, and it's then simply distortion at a lower level.

    It's actually impossible to accidentally have clipping in your master if you just look at the meters in your recording program, Sonar, or any other. There are peak meters that show you what's going on. In Sonar, clipping is shown by a horizontal red bar at the top of the fader's module. Keep your eyes on the Master fader. If you see that red bar appear there, then it will always also appear at the top of that sound interface module at the far right in the mixer. Re-adjust your Master fader accordingly, bringing it down just a tiny bit. If you feel that the lower level is hurting the overall mix, then seek out the specific track with the bit causing the clipping, and with a volume envelope, notch down just that single track at that point. There are no unpleasant surprises if you look at your meters while mixing. And if you didn't notice the moment your mix peaked out, you'll still know it did at some point, because that red peak display stays there after the peak has gone by.

    EQ can be the most mysterious element in the mixing process. In your post you're absolutely right that the style of music Is a very important factor in making EQ decisions. Many people use no EQ at all when working with orchestral pieces. Some of the best sound engineers in the world who work with live orchestras say they use very little if any EQ, because, as you're indicating, the idea is to produce a natural sounding recording of the orchestra.

    BUT - and it's a very important But - working with tracks of a real orchestra and working with virtual orchestra tracks are very different things. The samples we work with have been given a thorough working over, and theoretically the EQ of the samples is already what it should be - But in reality, problems come up fairly often as we work on our mixes which are combinations of those sampled instruments. The accumulated sound of a virtual orchestra simply isn't going to be as transparent and natural sounding as a live orchestra. We always need to use reverb in order to create a venue for our virtual orchestra to play in - just that process alone can introduce unwanted band frequency problems.

    Still, even though we may need to use some EQ in our orchestral mixes, depending on the particular blend of sound we're getting in a given piece, we would never need to use as much as pop music producers use in order to get the shimmery, punchy sound they're after.

    The process can be much simpler than what you're puzzling over, Richard. The ideal situation is to have accurate monitor speakers in our studios so we really know what our mix is sounding like. Then we just need to be extremely critical of our own work as we listen to the playback. Is it seeming a bit muddy? That's the most common problem, because bass frequencies pile up quickly, with the problem getting more pronounced with every additional track in a mix. All you need to do is go to the tracks with the most bass content and turn on the track EQs, selecting the pre-set which rolls off bass. Try that on some tracks, and you'll definitely hear more clarity.

    Less commonly, we may have a pile up of high frequencies, like if we have tinkly bells along with instruments like the piccolo. We may need to use the EQ preset that dampens the highs on some of the tracks, taking out a potentially unpleasant emphasis of the highs.

    And so on!

    Randy

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