How I master (step by step), please read and give advice

[Ben Godin]

I have been into mastering for a good while so i decided to go and buy the Waves masters package and the Trueverb and IDR plugin, also my WAVELAB comes with the UV22 and a resampling machine.

So this is what i do...

I was in charge of a prog. rock album that was in many ways like the smashing pumpkins, this is what i did...

the song came on a cd so it was in 16/44.1, i loaded it into wavelab, and in the master channel, i loaded the resampler, convert to 48 khz, and after that in the dithering channel i loaded the uv22 and set it to normal 24 bit dither (so i dithered up from 44 to 48 and from 16bit to 24 bit), next i rendered a new file, so this file was a 24 bit 48khz new version , i listened ... sounded about the same, good, next i applied a noise reduction program, and worked with the noise reduction untill just enought noise was removed to the point where the song retained all of its life and energy, and some of the headhphone bleed and mixerboard noise was removed. Next effect is the eq. i use the waves broadband eq and cut 1 db at 300 and boost 1 db at 6000 and 1 db at 10000. Next effect is the multiband compressor, i apply a fairly powerful threshold ( multi electro masterting, i stop lowering when the DynamicLine is in the middle of the freq. blue band and stays fairly put, usually this happens at at least -30 threshold to anywhere near -70 threshold in the uper most band). Then i apply a reveb, in this case waves truverb, i am careful not to leave trails and cut out any early reflections, in this project i used the vocal plate preset and tweaked the settings until right. Finally i use the waves L2 and apply a fairly strong threshold untill the compressed file comes close to the -.2 but i am careful not to overcompress, i leave the process to as 24 bit not 16. Next i resample using the resamler function and change the 48 khz to 44.1 . I do not use the L2 dither function so i turn it off. I then go to the dither stage, i either use the Waves IDR or the Apogee UV22, whichever sound better, i can often hear a difference in the two and chose accordingly. The final file came out faily well if you want to hear it this is the link...

TITLE: SONG 1
LINK: http://www.soundclick.com/bands/7/beng410music.htm

Now here are my questions:

1. please comment about mistakes or things that i do that seem incorrect such as chain placement or other things

2. Any constructive criticism would be great (about the song itself or how i achieved this sound)

3. Do you recomend any hardware gear which would furthur my chain, such as replacement eqs. After buying the masters package i have about 400 dollars left so anything useful will be considered.

Thank you for reading , i created this because many people wanted a simple tutorial, i know i have no right to make one because my methods may be wrong, but if any of the ME or anyone else wants to do something of this type, this is the kind of golden information that i and other rising ME would love to be able to read, may this post spark others to do similar posts.

[gelbardn]

AFAIK, you aren't gaining anything by upsampling from 16/44.1 to 24/48 for processing, and then downsampling again. You're probably introducing
more noise, if anything. You can't add anything that's not already there.

I would suggest keeping your processing station at the sample rate that the original material came in.

[Michael Fossenkemper]

Ok, mistake 1) do not resample from 44.1 to 48k unless you need to and only then, use a excellect SRC. Wavelab is not. Your just introducing distortion and such without any benifit. Next) you don't have to rewrite the file to 24 bit. Almost all DAW mixers are 24bit. even if you have a 16 bit file and you move the fader .1db, it will become a 24bit file. if you do any processing with plugins and such, it will become a 24 bit file. You do need to dither down to 16 at the very end of your chain. Other than that, use your ears in any decision you make. A/b the before and after to make sure your not doing any damage. don't let the volume fool you.

[gelbardn]

Also, from 3:37 and out, with the last little picked guitar chord, it sounds like the guitarist is standing in front of a computer monitor (that buzz in the background).
It doesn't SEEM present through the rest of the recording, however.

[Ben Godin]

thanks for the reply, i seem to have trouble understanding resampling, see i can't do much with a 16bit, 44.1 file, and yet with the 24 bit file 48 khz, it seems to me that the sound comes out better...

Also for the last comment, that last riff there was just a screw up in the recording stage, and too much noise from the amp was picked up, i couldn't do much about it without having drastic effects on the rest of the song, believe me, it was MUCH worse when i got the song, and it was throughout the whole song

[Ben Godin]

also if someone would care to review my song , this would help in knowing what i could do better next time...

[Michael Fossenkemper]

your most likely hearing the benifit of operating in 24 bit rather than 48k. most daws operate in 24 bit or higher even though the soundfile is 16bit. if you take a 16 bit file and bounce it as a 24bit file without doing anything to it, it's still a 16bit file. the last 8 bits are empty. The internal mixer of the daw should be operating at 24bit or greater. this is bit depth and is seperate from sample rate conversion. here is a test you can do to see how well SRC is doing. take a 44.1k file, SRC it to 48k, SRC it back to 44.1k and compare it to the original 44.1k file. It's going to sound worse. now if you take a 16 bit file, bounce it as a 24bit file, then bounce it back to a 16 bit file. It's going to sound exactly the same IF your daw is bit transparent. So SRC is a desctuctive process that should only be done if it needs to be, Like if you have a 48k file and need it to be 44.1k.

[mixandmaster]

Using the waves masters bundle, I think the place where you can make up the most ground with this tune is in the multiband compressor. The cymbals are dominating the high end. Maybe compress the highest part a little more - which will cut the high end when the cymbals are struck but leave the rest pass through. As with all compression, pay CLOSE attention to the ALL the paramaters, (amount, threshold, attack, and release times on the waves MB). Make sure you don't hear the compressor go on and off (unless you want to :? ). I also want to hear more low and lo mids...which you can add by eq or manipulate the multiband compressor.

Good luck. 8-)

[Ben Godin]

hey thanks turtletone and mixandmaster,

turtletone- i performed the test and you were right, now here is a question for you, how come MEs say that the first step is to convert to 24/96 when they get the file off of its orig 16/44.1 ?

mixandmaster- you are very much correct, the cymbals were very uncompressed, and i am re-doing the song and fixing that, i guess that with only a year's expirience i still have much to learn, i am working with the mid-low (300-1000) where the quitars are dominant, i have raised them some.

Can someone recomend what to add into my mastering chain, as long as its under 400, well, make that 800 (bday money comes to a use sometime) , ill gladly go out and buy it. Or a number of things that add up to 800. Super thanks to you guys. :D 8-)

[Michael Fossenkemper]

The reason why ME's convert to higher sampling rates is because 1) they have excellent outboard SRC's that are virtually transparent 2) because the filters in mastering gear when operating at higher sampling rates are smoother in the audible range. But this is really only a benifit at 96k, not at 48k because the filters have distortions at 48k that are still within the audible range.