tips and tricks

[mersisblue]

Id like to say we've been given a chance to start from scratch here lets do

this right


finding correct volume levels in the mix
one tip is to turn the master fader down and listen to see if anything is drowned out if so it may need to come up

another way is to turn the master fader up and walk into the other room and take a good listen to the mix

you may hear something from this perspective

 

[polygen]

...and never be afraid to bring ALL faders down if the mix just won´t feel right. then bring up kick, snare, bass and vocal. tweak till these fit eachother nicely, then gradually bring in other stuff.

 

[boody]

...and never be afraid to bring ALL faders down if the mix just won´t feel right. then bring up kick, snare, bass and vocal. tweak till these fit eachother nicely, then gradually bring in other stuff.

amen to that 8)

recent visits to gearslutz lead me to the wonders of automation. I used to correct stuff with it, but now I'm experimenting with automated eqs, different compressors/ settings for different parts of the songs etc.

All this after I read that for example some of the bigger engineers tweak eq to the vowel.... and I used to be happy with just one eq/compressor/level setting for the whole song. Guess I was lazy :wink:

 

[sniper]

well i'm NO expert on this matter but the use for eq automation might be in place in a scenario like this:

2 guitars, 1 centered, one panned full right. suddenly a 3rd guitar jumps in and performs a solo, centered.

now instead of lowering the centered backing guitar maybe it would be a better idea to eq it and make place for the frequencys that are dominant on the solo guitar.

i've been meaning to experiment with this as i too NEVER automate. but i still haven't come to the mix phase of my current project. (which i subconsciously keep pushing forward.. 8-[ )

 

[GP]

Id like to say we've been given a chance to start from scratch here lets do

this right


finding correct volume levels in the mix
one tip is to turn the master fader down and listen to see if anything is drowned out if so it may need to come up

another way is to turn the master fader up and walk into the other room and take a good listen to the mix

you may hear something from this perspective

How about using "crappy" computer speakers as reference to getting the levels in the mix? Actually, I beleive it was SUNTOWER who started a topic at the old forum on this. And yes Suntower, I have the "cambridge soundworks" besides the yorkvilles. As much as I hate the cambridges, they come in handy as a reference to levels.

 

[cAPSLOCK]

There is the 'go out of the room' like tip from Craig Anderton where you mix a track of noise with your mix to get an idea of what falls out. Wacky but cool.

http://archive.keyboardonline.com/colum ... 0003.shtml

cAPS

 

[GP]

Good link cAPS. I actually tried this a while back with pink noise. Interesting results. Something different to say the least. But just another tool to utilize during the mix process. :wink:

 

[j2]

and the best way to get levels perfect in a mix........

10 Steps to improving the way you listen
-----------------------------------------------
(given that you've recorded all tracks with appropriate levels)


1] compress every instrument, even vocals, with the same compressor and same compressor settings (fast attack + fast release for this exercise)

2] bring all faders to approximately 60%

3] close your eyes but imagine that you are looking straight at the center of your mix, playback at a level louder than what you typically use, and listen to 10 seconds of a busy section of the song keeping mental notes of distance, front and back, side to side of each instrument

4] stop playback, open eyes, remember what the purpose of the song is and uncompress the tracks that seemed too far away, leaving the remainder compressed

5] hand on spacebar, close eyes, go.......again, listen for this change in space between the tracks and how quickly they come at you as opposed to earlier when they all came at you with nearly identical response

6] 20 seconds of listening this time then stop playback, open eyes, and reverse the process of enabling/disabling compressors as in step 4

7] give your section one more listen for 30 seconds (patience)

8] by this time, you have made mental notes of what needs to be changed strictly devoted to the space, response (quick or slow response), and lastly volume

9] EQing after compression can give more presence to your track, but if you're a rock engineer, you'll probably want every instrument to sit in it's spot except for the vocal (which can easily have EQ+comp+EQ+more comp...)

10] after you've spent about 10 minutes with a decent mix, push all tracks up to approximately 85%, throw a compressor with fast release and slow attack on the main buss, slowly decreasing the release (making it slower) and listen again for:

- space front and back
- response time
- overall equalization

if you can't learn anything about how you mix or record your tracks from these 10 steps, you might want to consider selling gear instead of using it

=)

 

[MatsD]

I think there are never just one perfect level for an instrument in a mix, you need to ride the faders if you're into traditional analog recording or use automation if you have a modern mixer console or a DAW. Also, most problems with levels comes from bad arrangements, too much going on at the same time in the same frequencies ranges. Curing this by referring certain instruments to strict different frequency ranges (that is a popular advice) often results in a lifeless synthetic and dated sound. If instruments doesn't compete as much the relative levels are not as critical and you do not need to abuse the individual tracks as much, which gives your song a more natural sound.

The most critical level is the lead vocal relative the rest of the mix, so important that engineers often make several mixes for the mastering house. But how loud the vocals should be varies very much between genres I think. Other critical tracks are snare, kick and bass tracks, they define much of what basic impression you get from the song. But again the correct levels differ a lot for different genres.

Since ideals vary so much, I think the only apparent advice is to have a professional mix of a similar song from your favourite record as a reference. Listen to how the tracks are out of the way of each other, is it by means of pure level, is it by panning, eq, ambience or just clever arranging? For arranging clues, listen to big band recordings before stereo and multi track recording was established or classical music, there is a lot to learn from there that can be used in mixing with modern equipment and in contemporary genres.

/Mats D

 

[frans]

in the car

Levels and EQ: i listen to mixes in my car while driving as this adds all kinds of noises which, when something is lacking volume or fudamental frequencies just makes it drop out.

I think this will be helpful for beginners.

 

[Drammy]

Thought I'd add a tip here and see what you guys think about it.

I have recently been told never to use the export Audio function of a sequencer to mixdown tracks with.

I use Cubase SL2.0 and up to now have always exported audio to create the final mixdown.

I have since been told to go about it a completely different way. Apparently it is best to send the audio out through a digital out and then record back into a wave editor or seperate sequencer track via a digital in.

Apparently it is quite surprising how much life is taken out of tracks due to exporting a track as audio. I was played two examples, one using this technique and the other not. I could certainly hear what they were talking about, the exported file sounding a lot more compressed than the recorded track.

I haven't had chance to put this theory into practice yet, but hope to give it a go on the track I am currently doing.

I will post some examples somewhere for comparison but if anyone else has any comments regarding this tip, feel free to criticise or pass them on.


Martyn

 

[Ericcc]

Thought I'd add a tip here and see what you guys think about it.

I have recently been told never to use the export Audio function of a sequencer to mixdown tracks with.

I use Cubase SL2.0 and up to now have always exported audio to create the final mixdown.

I have since been told to go about it a completely different way. Apparently it is best to send the audio out through a digital out and then record back into a wave editor or seperate sequencer track via a digital in.

Apparently it is quite surprising how much life is taken out of tracks due to exporting a track as audio. I was played two examples, one using this technique and the other not. I could certainly hear what they were talking about, the exported file sounding a lot more compressed than the recorded track.

I haven't had chance to put this theory into practice yet, but hope to give it a go on the track I am currently doing.

I will post some examples somewhere for comparison but if anyone else has any comments regarding this tip, feel free to criticise or pass them on.

Martyn

Hmmmm...that sounds quite interesting...anyone ?

 

[Eurocide]

Why should the 1s and 0s sound different by sending them 1:1 trough a digital device??:?
Sorry, but this is mindf**k, because the same 1s & 0s you are listening to via the Cubase main outs are recorded at the wave editor. So no difference to an audio export is possible.
It would make sense to send the mixdown through a hi-end outboard device like Avalon, Manley etc.
But that will already be done in the mastering process.

Or does anybody know if the audio export does someting else to the material than putting the digital data into one file?

 

[j2]

Glad this was brought up.

I do know that Nuendo 2 has the best downmixing algorithms to date, even if you don't use a dither on the master bus.

I too have recently decided to send my mixes through my 01v (strictly staying digital) - and hate to admit it, but using Monster cables. I've done two mixes sending out of FruityLoops, one over ADAT and one over SPDIF. I've also tried using different kinds of cables and sending versions out through ADAT vs. SPDIF outputs. Lightpipe seems to have a more even sound and SPDIF seems more analog, in a sense that there's more noticeable events taking place in the overall spectrum. Maybe because one's pure light and one's still using wire.

The main difference that he's talking about is simply more life to the mix. This is because you have to compensate for the output by sending it back a little hotter, thus creating more overall volume to the mix, even though you're balancing it back to a similar volume as the original. I've noticed that if I overcompensate sometimes, I get a \"compressed\" sound to my mix, but only positive compression (volume enhanced).

Some people are paranoid and don't trust the downmixing algorithms because they're still analog (real-time) fiends. I have experimented like I said and I find it to add, in the spectrum side, an overall boost in certain lower frequencies, only because if the song contains this information, the volume balancing you're doing going through the external device is bringing this level up. Don't quote me on saying \"some guy said if you send your digital mix out to an external digital device and back, you get more low end\" - I just noticed it with *my* equipment and only in the frequency range of 80-125hz. These are fundamental \"feeling the music\" frequencies, aside from the obvious pounding lower frequencies.

Tests were done using:
--------------------------

] FruityLoops 4.5.2 (FLStudio)
] Nuendo 2.1.1
] SoundForge 6 - always recording back at 32-bit
] Hammerfall HDSP - using Hammerfall DSP Mixer as hotter sends to 01v
] Yamaha 01v w/ ADAT expansion

One last thing for all the Fruityloops users. If you want the best sound possible period, even if your session is as low as 44.1/16, always downmix a 32-bit float (0.24) with interpolation set to Sinc 256....6-point Hermite will sound almost as good, but take a lot less time. You will get a mixdown that sounds just like what you heard through playback. Of course, this is if you're satisfied with having a 'safe limiter' (ElephantHQ) on the master fader to keep levels below 0db - I tend to always have a snare or vocal going over 0db, thus downmix realtime.

If anyone's *THAT* interested, I could whip up some a/b's

 

[MatsD]

I think it's bullshit that export should sound any worse than realtime. If realtime sounds better, it would be explained by you running it thru something that adds something you happen to like to the sound. Certainly mixing down to analog tape would make a big difference in sound.

It's fairly easy to test the quality of the export algorithms. Do an export, import it to a stereo track in the same song, let someone else switch back and forth randomly between the original song before mix and the stereo track with the imported mixdown with half a second of total muted silence between changes (to allow for switching to the same option several times in a row) and guess away. Do it enough times for it to have any statistical value (100 rather than 10).

Edited:

There could of course be an explanation in some plugins performing incorrectly during export. I haven't noticed this in Cubase but sometimes when rendering effects in Wavelab, which is a simliar task. The most recent example of this problem I've come across is the new stereo-widener in the Blue Tubes Bundle from Nomad Factory. It sounds totally different after rendering compared to realtime. I wrote Nomad Factory's Support about that and some installation issues, but like most support departments they don't consider it their job to address problems or answer questions from users. :evil:

/Mats D

 

[japut]

I have to agree here about the export functions on most audio programs. I started using a channelgrabber to take my audio in realtime and there is a difference. Try it and maybe you will see it the same way. Give tobybears channel grabber a run, it's free and works very nicely. Even at extreme cpu usage no clicks or pops even though my soundcard is spitting like a cobra... 8)

 

[jcat]

Hmmm, I can't say I notice much difference if I export audio rather than re-record in through my 02r. In fact the 02r only has a 20bit wide audio path, so I'm loosing resolution if I mix through it. I do all my mixing internally now in SX2, then export the audio at the end, hencing removing the need ot dither and keeping everything 32bit for mastering.

If anyone wants to prove it (and can be bothered), just do a short mix, one exported, and one re-recorded. Then do a reverse phase null test on one of the channels, just make sure there isn't any time delay on the re-recorded track so as they're lined up properly.

My money's on no difference. If they sound different like louder or something then you may have set something up wrong. If they sound different like more bass or something, then I still reckon you've got something set-up wrong or are imagining it maybe.

All the mix down does is play the song through, and record the result to a file. If that file's 32bit float, there will be no difference at all.





Cheers,

jcat

 

[japut]

Like I said I'm using a channelgrabber so the audio still is not leaving my soundcard to be mixed and I know for a fact there is a difference in my ear. I'll get on with the test tonight or tomorrow but if you could return the favor and download a channel grabber so that we can get a get a full testing scenario going on. It would be much appreciated.

 

[chewie]

It´s not that hard to check if there´s a difference...just import the two versions to two stereo tracks, phase invert one and if there is a difference you should actually HEAR the difference This requires that both versions are EXACTLY the same length e.g. that they start at the same sample.

Chewie

 

[j2]

About the 02r word length....how old is yours? My 01v isn't the 96k version, but it still has 24bit as the highest option.

What I've mostly noticed sending my mixes out of my computer and back in is that my cables are giving it a not-as-digital sound, even though they're digital cables!

I'm not going to argue that for the purest possible sound, a 32bit float mixdown through SX2 or N2 is definitely the cleanest. But in today's world of mp3's, the more digital your file is from the start of encoding, the more digital the mp3 will sound. Seriously, if there is more digital content to a file, Fraunhofer will still try its best to retain some of those upper frequencies if they exist and in the process, not retain it all. If you've given your mix a pass and made it lose a negligable amount of high end data, or if you're coming straight from A/D converters off a 1/4\" master - the mp3 won't suffer at 320k. Only the obvious mp3 compression sound will occur, but the frequency range will stay close enough.