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> One For Ivan And Bogdan, trialing a new mastering compressor
tonymiro
post Feb 15 2010, 07:37 PM
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We've just been sent a new mastering compressor on evaluation for a few months by one of our suppliers. It's an Elysia Alpha and our supplier is pretty keen that we use it as well as/instead of our main compressor.

Anyway, Ivan very kindly provided me with an audio wave of a track he has recorded recently with Bogdan and their group Ljute Papricice called 'Nocni Gruv' to put through the Elysia. I should say at this point that what follows was aimed at evaluating the Elysia so Ivan and his group had very little chance for input in to what I did and so have been particularly kind to allow me to post the result here. Thanks guys smile.gif .

Ivan has very kindly posted the original recording as an mp3 - see post no. 13 below. Thanks Ivan smile.gif

Anyway here is my 'Nocni Gruv' Elysia test master for Ivan and Ljute Papricice.

Attached File  Nocni_Gruv_Master.mp3 ( 5.94MB ) Number of downloads: 177


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And here is an edited version of the mastering log to give you all some idea of what went on and why:

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Client: In house Elysia Alpha Mastering Compressor test.

Track: 'Nocni Gruv- Ljute Papricice'

Original: 96/24 Wave Required final format:44.1/16 Wave for Ljute Papricice plus MP3 [for upload to GMC]
Track length: 3:54 approx 110mb 96/24 wave.

[File was sent zipped via internet. Zipping here both reduces the file size and also provides a degree of data conformity checking.]


Internal DAW to 96/24 [to match the 96/24 wave; any final format changes required will be done as a final when I re-convert to 16/44.1/mp3.] Chain is :

Daw aux out -->Lavry DA1--->Elysia-->Ibis--->Lavry AD-->Daw aux in: daw main-->Lavry DA2 -->monitoring

Monitors and meters to K14 [I don't use peak meters for mastering. I use the K system and usually set out at k14 for rock/jazz. Monitors are pink noise calibrated on the K system. This means that I can accurately set, maintain and repeat spl, volume etc for mastering. This is important both wrt the Fletcher Munson effect and also so that I am much less likely to clip on transients.]

Normally a client would supply me with a reference track - a piece of music that they like the sound of that they want their recording to be like. I didn't have one and didn't discuss with Ivan what he wanted as this is about evaluating the Alpha.

Import into sequencer. No errors from sequencer. Save Session 1. (As most of the work here is OTB the EDL isn't as important. Instead I'm dealing with sessions - each one ending in a discrete audio print/take of the processed audio.)

1st audio play - impression - nice original studio recording number . Generally well recorded although, to me, has a slightly emphasised mid band. Levels fine.

2nd audio play [focus on what Elysia may be for]

Glue - bring the recording together and make a little more cohesive. Possible to bring up bass and add more punch to the drums as this should work for this type of funk. Final levelling to bring up rms for the 44.1/16 wave. Overall I'm looking to use the Elysia then three times - normally for most mastering sessions I tend to do one analogue pass. Most, possibly all of these three could be done in the Elysia in one pass but I've opted to do each one separately as it's easier for me to clearly hear what it's doing where and how.



Session 1 is the original, unprocessed 96/24 wave.

Session 2
I've put the Alpha in a very basic compression setting to bring the recording together. Threshold is down at -25, moderate ratio at 1:2, release at 280ms.

Save and AB with Session 1.

Playthrough 2- see previous comments. Note I'm AB'ing what I've done against the original - no point in making a change if it doesn't actually improve on the original. The Alpha actually allows you to monitor both the unprocessed and processed signal so you can AB in the Alpha. However, to keep things simple at this stage I'm going to monitor via the second DA on our Lavry and put down a temporary print between 'sessions'.

Session 3
Drums and bass - as this is quite a nice funk groove I want to bring both up in the mix and add a bit more punch, particularly to the drums. It would be possible to do this using a split band or multi-band compressor but MBCs very often introduce unwanted artifacts in to a recording. It's usually best therefore to use an MBC rarely and with caution in mastering. I want to add a bit of weight to the drums whilst maintaining a sense of groove to the track. To do this I've focused a lot on both the release time and the gain reduction limiter and I'm using the side chain of the Elysia on LP.

Save as Session 3.

Playthrough 3 - generally better to me - bass and drums have come up and there is more punch on the drums and they sound 'bigger' with a bit more weight and ambience. However, think some of the warmth of the original has gone and the track now sounds a little bright.

At this stage I would normally use a separate eq - either our Ibis or, if ITB a Flux or Sonoris plug-in. However, the Alpha has an Audio Filter so I'm going to use that as a sort of shelfing eq to bring the low end up by about 0.5 dB and the high end down by the same amount with a cross-over at about 1k.

Playthrough 4 - better, track sounds warmer but Ivan's guitar sounds a little brittle. As I want to focus on his guitar I'm going to bring in an external EQ.

Session 4
Ibis added to signal chain. EQ cut -o.8 dB, medium q, 610hZ, bell. Note - normally I'd put an eq before the compressor but in this particular case it works better after.

AB - better, Ivan's guitar still, to me, sounds suitably funky but without the previous brittleness.

Save as Session 4

Session 5
Last bit for the Alpha - I'm going use it to level the track. I want to try to bring up the rms by a few dB and so give the track increased perceived loudness but without using a brickwall limiter or hardclipping. The Alpha has a soft limiting function and I'm going to set the overall gain stage so that fast transients clip out at -.5.I'm aiming at -.5 rather than 0.0 or -.2 (which I normally use) as I'll have to process the final track as an mp3 and LAME tends to add between 0.2 to 0.5dB to peak levels - if I aim at -0.3 LAME could easily force hard clipping and produce some nasty digital distortion on the mp3. [I did the gain stage here manually just by listening for overshoots rather than watching the meters].

Downsample to 44.1k dither to 16bit and print as CDA Wave for Ivan.

Final listen/quality check on 44.1/24 wav
Plays through ok but there is, to me a little distortion on a couple of fast transients at, for instance, 31 seconds. To me this seems to actually be intersample peak distortion and may have been partly as a result of the multiple, sequential sessions saves. Even though the set up, etc have all remained at 24/96 and no dithering has taken place until the final 44.1/16 print it's a possibility that the multiple saves has accumulated distortion (the final track is a forth generation.) A single pass through the Elysia Alpha and one session take would probably eliminate it.

44.1/16 bit wave sent to Ivan as a zipped file (approx 38 Mb)



Format change to mp3 VBR for GMC. Added linear fade out at Ivan's request.

Total time - approx 1 hour 17 mins. - slightly longer then normal as quite a bit of time was experimenting with the Alpha and some additional time connecting the Ibis in to the signal chain.

This post has been edited by tonymiro: Feb 16 2010, 07:08 PM


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Staffy
post Feb 15 2010, 07:47 PM
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Very interesting Tony, but I can't see no file???
One question here, is it normal to do the mastering in several "passes" eg. first apply one level of compression, then maybe EQ and then another level of compression and then finally dither???

//Staffay

Very interesting Tony, but I can't see no file???
One question here, is it normal to do the mastering in several "passes" eg. first apply one level of compression, then maybe EQ and then another level of compression and then finally dither???

//Staffay


--------------------


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tonymiro
post Feb 15 2010, 08:04 PM
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You should be able to see the file now Staffy - didn't attach properly the first time around (been a long day wink.gif .)

I did several passes for one main reason - the Alpha is a very complicated bit of hardware. Also as it's hardware I can't insert several versions of it in the channel - only way to do that would be either to have several Alphas and with the price of it I doubt anyone will be able to do that wink.gif. I find with stuff that I've never used before that I get to know it better and quicker by doing things in small discrete phases. So I deliberately did this here and because I essentially got the Alpha to do several different things I had to print each stage. Once I get to know it well enough I'd be able to do all of them on a single pass.

Dithering - when it's needed - though is always, at least to me, the very last stage prior to rendering the CDA. I sometimes do that as a separate stage to all the previous processing since once its done its done. Even though you can bring the audio back up to say 24 bit from 16 dithered you will still have truncated the original and you cannot get it back - the new 'extra' 8 bits of data won't contain the data removed at dithering. Because of this I'll sometimes print a session with all the processing but without dither and then dither and render to cd as a separate, final session. That way if a client wants me to make any alterations I can just go back to the undithered session cool.gif . That make sense?


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Staffy
post Feb 16 2010, 08:20 AM
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QUOTE (tonymiro @ Feb 15 2010, 08:04 PM) *
You should be able to see the file now Staffy - didn't attach properly the first time around (been a long day wink.gif .)

I did several passes for one main reason - the Alpha is a very complicated bit of hardware. Also as it's hardware I can't insert several versions of it in the channel - only way to do that would be either to have several Alphas and with the price of it I doubt anyone will be able to do that wink.gif. I find with stuff that I've never used before that I get to know it better and quicker by doing things in small discrete phases. So I deliberately did this here and because I essentially got the Alpha to do several different things I had to print each stage. Once I get to know it well enough I'd be able to do all of them on a single pass.

Dithering - when it's needed - though is always, at least to me, the very last stage prior to rendering the CDA. I sometimes do that as a separate stage to all the previous processing since once its done its done. Even though you can bring the audio back up to say 24 bit from 16 dithered you will still have truncated the original and you cannot get it back - the new 'extra' 8 bits of data won't contain the data removed at dithering. Because of this I'll sometimes print a session with all the processing but without dither and then dither and render to cd as a separate, final session. That way if a client wants me to make any alterations I can just go back to the undithered session cool.gif . That make sense?


Sure it makes sense Tony! I would love to hear the raw file and compare it to this one!

One thing I noticed is that the original from Ivan & Bogdan is at 96/24. What is the point to record at 96KHz when You are supposed to downsample it all to 44,1 in the end anyway? 24 bit makes sense, but recording in 96 only makes You to dither it at the end. Wouldn't it be better to record at 44,1 and skip the dithering???

//Staffay


--------------------


Guitars: Ibanez AM-200, Ibanez GB-10, Fender Stratocaster Classic Player, Warmouth Custom Built, Suhr Classic Strat, Gibson Les Paul Standard 2003, Ibanez steel-string
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Ivan Milenkovic
post Feb 16 2010, 12:50 PM
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QUOTE (Staffy @ Feb 16 2010, 08:20 AM) *
Sure it makes sense Tony! I would love to hear the raw file and compare it to this one!

One thing I noticed is that the original from Ivan & Bogdan is at 96/24. What is the point to record at 96KHz when You are supposed to downsample it all to 44,1 in the end anyway? 24 bit makes sense, but recording in 96 only makes You to dither it at the end. Wouldn't it be better to record at 44,1 and skip the dithering???

//Staffay


During the time we prepared projects for album in Nuendo, I was working on a computer that had Audigy card in it smile.gif Having said that, all the projects were 96/24 by default, since that was a default mode for the audigy card. When I brought the projects in the studio our producer didn't notice it, and we started working directly in 96/24. Later on when he did saw that the projects are all running on 96/24 he asked about it, I explained, and he just continued working, it was not a big problem, and to rework the material before that point was not very effective.
I must say I'm glad that we did in such a high/unusual resolution, although it was unintentional at first. We can easily remaster/rework the album's components at greater sampling rate that will eventually be standardized in the future. The decision was already worth it, as when tony requested the files, him being a ME, he needed the files in 96/24 to process. No need to upsample, as the file was directly rendered out of a project.

EDIT: I'll post original wave file that I sent to tony for evaulation Alpha as soon as it uploads.

This post has been edited by Ivan Milenkovic: Feb 16 2010, 12:52 PM


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Staffy
post Feb 16 2010, 12:59 PM
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QUOTE (Ivan Milenkovic @ Feb 16 2010, 12:50 PM) *
During the time we prepared projects for album in Nuendo, I was working on a computer that had Audigy card in it smile.gif Having said that, all the projects were 96/24 by default, since that was a default mode for the audigy card. When I brought the projects in the studio our producer didn't notice it, and we started working directly in 96/24. Later on when he did saw that the projects are all running on 96/24 he asked about it, I explained, and he just continued working, it was not a big problem, and to rework the material before that point was not very effective.
I must say I'm glad that we did in such a high/unusual resolution, although it was unintentional at first. We can easily remaster/rework the album's components at greater sampling rate that will eventually be standardized in the future. The decision was already worth it, as when tony requested the files, him being a ME, he needed the files in 96/24 to process. No need to upsample, as the file was directly rendered out of a project.

EDIT: I'll post original wave file that I sent to tony for evaulation Alpha as soon as it uploads.


Oh well, if its 96/24 its gonna be a large one... ;-)
You have a point there with future formats, but I still not getting why the mastering has to be done in 96Khz when the outcome shall be a CD at 44,1 ........... ????

//Staffay


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Guitars: Ibanez AM-200, Ibanez GB-10, Fender Stratocaster Classic Player, Warmouth Custom Built, Suhr Classic Strat, Gibson Les Paul Standard 2003, Ibanez steel-string
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tonymiro
post Feb 16 2010, 01:53 PM
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QUOTE (Staffy @ Feb 16 2010, 08:20 AM) *
... What is the point to record at 96KHz when You are supposed to downsample it all to 44,1 in the end anyway? 24 bit makes sense, but recording in 96 only makes You to dither it at the end. Wouldn't it be better to record at 44,1 and skip the dithering???

//Staffay


Recording at 24/96 - 96 gives a clearer sense of the high frequencies and a much more open and transparent treble. Mastering that means you've got a much better insight in to the high frequency range and so can hear the upper harmonics and all the decaying tails of the cymbals etc much clearer. There are a couple of caveats/issues here with 96 though:

1-although a lot of 'prosumer' hardware will run at 96k very often the quality of the prosumer kit isn't enough to justify it. This is especially true of sound cards.

2- a 24/96 recording will take up more file space than a 16/44.1. Ivan's original 4 minute tune was about 110Mb and the final 16/44.1 was about 38. An audio CD holds around 720Mb of audio (an extended can go to 800) so if we used 24/96 you would get about 25 minutes of audio. (It also wouldn't be red book as that requires CDA at 16/44.1...)

3 - at some stage it needs to sample rate corrected (SRC'ed) and again a lot of 'prosumer' orientated daws etc whilst able to do SRC don't do it well. Although SRC isn't as problematic as dithering/word length reduction it's best not to do it a lot.

SRC by the way isn't the same as dithering - in the former you are changing the sample rate , in the latter you are changing bit depth and reduce the word length of the audio. Sample rate is about how often samples are taken of the analogue audio to produce the digital code and the higher the sample rate the better the higher frequencies come out. Bit depth is about the dynamic resolution (both dynamic range and the signal to noise). The higher the bit depth the better the dynamic resolution. Dithering is about reducing the word length /bit depth.

16 bit, although it's the standard for CDA has a reduced noise floor and dynamic range - ie 16 bit is @-96dB but 24 is @-142dB. Whilst it is perfectly possible to work in the reduced dynamic range and get a good recording at 16 bit the extra 8 bits here, with the improved dynamic range is generally worth having even taking in to account the final dither from 24 to 16. Pretty much all mastering hardware that will take digital in runs at 24 bit because of the improved dynamic range and lower noise floor. Hence why most of us MEs ask people to supply files as 24 bit.

The issues with changing bit depth from 24 to 16 is that you both remove audio data and you increase the amount of distortion in the audio due to quantisation error. In the former you can never get back the data that is removed so dithering 24 to 16 and back to 24 will not give you back the original data - you will get a 16 bit depth file plus 8 bits. In the latter there are different ways of dealing with the distortion - and this essentially is dithering. Dithering involves masking the quantisation error by putting low level noise over it. There are different ways to dither - how and where you apply the noise, and the type of noise you apply. So when you dither though you are 1 reducing the bit depth, 2 inducing quantisation error, 3 masking that by inserting low level noise. Hence, if you have to dither do it as little as possible...

You don't have to dither - you can instead just truncate the bit depth from 24 to 16 but then you will have quantisation error. In some cases this may not be too noticable but in the majority it will be.

So why 24 rather than 32? Quite simply as most ME grade hardware runs at 24 bit. Because of this there is no real advantage in 32 bit and it can actually cause problems with the truncation of those additional 8 bits. (There are other reasons but that tends to involve a complicated argument about whether 32 bit float is as good as 24 fixed maths. On a subjective level, and avoiding the maths, all the test files I've heard leave me to believe that 24 fixed is at the moment better.) Why 96 rather than 192 or higher - audibly there is very little noticable improvement above 96.

So overall 24/96 tends to give a good compromise provided your recording set up can do it well enough.


--------------------
Get your music professionally mastered by anl AES registered Mastering Engineer. Contact me for Audio Mastering Services and Advice and visit our website www.miromastering.com

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Staffy
post Feb 16 2010, 02:07 PM
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Thx. Tony, this makes sense! But since the most plug-ins etc. in the DAW domain doesn't operate at this sample rate (96Khz), I guess that f.ex a guitar with a delay added are already downsampled in the DAW ??? The same must go for every effect used in my belief, or am I wrong here?? Also, wouldn't the greater sample frequency put a heavier load on the computer, since a larger amount of data must be transferred between the discs and the RAM... ..... (just trying to motivate myself why I should record at 96Khz tongue.gif )

//Staffay


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tonymiro
post Feb 16 2010, 03:28 PM
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QUOTE (Staffy @ Feb 16 2010, 02:07 PM) *
... most plug-ins etc. in the DAW domain doesn't operate at this sample rate (96Khz), I guess that f.ex a guitar with a delay added are already downsampled in the DAW ??? The same must go for every effect used in my belief, or am I wrong here?? Also, wouldn't the greater sample frequency put a heavier load on the computer, since a larger amount of data must be transferred between the discs and the RAM... ..... (just trying to motivate myself why I should record at 96Khz tongue.gif )

//Staffay


Most, maybe all (I can't think of one that can't) Mastering grade plug ins can be run in an over-sampling/high quality mode that upsamples it to 96 or higher. But you are right that most standard vsts don't/can't upsample.

With the Daw - the ones we use here for mastering we can set the sample rate and bit depth and it will then auto convert any stream to that on the fly. In this way you can have three audio tracks recorded at different sample rates - say 48 and 96 - processed by vsts at 44.1 and 96 and the daw will auto change the stream to what ever you tell it to. However, generally you don't want a lot of SRC going on so we tend to set everything to a single sample rate - usually 96 and so minimise the amount of src taking place in the daw - we don't let our daw downsample unless we need it to.

As an aside I can think of at least one major recording daw that routinely applies SRC and bit depth changes without user intervention - most users probably don't even know that it is tbh. (Not sure if it truncates or dithers for the bit depth though.)

Yes higher SR does increase the load on the pc. For mastering I'm not likely to have lots of lanes open and lots of plug-ins running and latency isn't much of a consideration for me. I have a quadcore i7, a lot of ram, and sata drives. The only software on that pc is the OS and mastering stuff - nothing else, no virus guard, no internet browser - nada. Background processes are kept to an absolute minimum. If I pull up a stereo main place it at 24.96 and put up say an LP EQ, a MP EQ, a compressor and a limiter and a dither plug the cpu trips up to between 15-32%. It's the consequence of running mastering grade quality stuff - it's hardware/pc intensive.

Other bit is file space - Ivan's number is just under 4 minutes and at 24/96 is about 110Mb. Five session takes and I've got 500Mb of audio stored on a hard-disc. We archive sessions for quite a while in case a client comes back to us in the future and wants a re-master. Fortunately hard drive disc space nowadays is pretty cheap smile.gif .

One thing I find kind of interesting is that many home users know what the bit depth etc of their chosen daw is but know little about why and how it affects what they do.


--------------------
Get your music professionally mastered by anl AES registered Mastering Engineer. Contact me for Audio Mastering Services and Advice and visit our website www.miromastering.com

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Staffy
post Feb 16 2010, 05:14 PM
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Yeah, I understand what You mean here Tony, but I was thinking of using effects in the DAW when running the recording/mixing process. Eg. if I shall make a recording, I see no point in using 96Khz since my VST-plugs will downsample the signals anyway. The only way to keep the whole recording 96Khz would be to use external hardware for the effects - and then we are talking money....
At some point i tried to use 96Khz, but as far as I remember some plugs started to sound weird, but I will make one more try with using just Powercore that supports 24/96 for the effects.

//Staffay


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tonymiro
post Feb 16 2010, 05:34 PM
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QUOTE (Staffy @ Feb 16 2010, 05:14 PM) *
Yeah, I understand what You mean here Tony, but I was thinking of using effects in the DAW when running the recording/mixing process. Eg. if I shall make a recording, I see no point in using 96Khz since my VST-plugs will downsample the signals anyway. The only way to keep the whole recording 96Khz would be to use external hardware for the effects - and then we are talking money....
At some point i tried to use 96Khz, but as far as I remember some plugs started to sound weird, but I will make one more try with using just Powercore that supports 24/96 for the effects.

//Staffay


Yes that is a problem. I'm kind of lucky here as when we do record and have to add effects I can go out to hardware at 96 anyway.

There are some non-mastering vsts though that can upsample. PSP make a few Lexicon emulations that upsample; Voxengo have some compressors and some eqs; I think all the URS Pultec and Neve emulations support upsampling...


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Staffy
post Feb 16 2010, 05:40 PM
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QUOTE (tonymiro @ Feb 16 2010, 05:34 PM) *
Yes that is a problem. I'm kind of lucky here as when we do record and have to add effects I can go out to hardware at 96 anyway.

There are some non-mastering vsts though that can upsample. PSP make a few Lexicon emulations that upsample; Voxengo have some compressors and some eqs; I think all the URS Pultec and Neve emulations support upsampling...


Hmmm, thx for the tips, I will check it out, but I will probably go out and get me a new computer first, since switching sample rate will put some heavy load on the box due to larger files to stream and to store in the cache... and Im already maxing my 4GB quad core out .... sad.gif

//Staffay


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Ivan Milenkovic
post Feb 16 2010, 06:07 PM
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Good points here. In this case all the plugins are capable of 96/24, and the truth is there aren't that many "bells and whistles" in these. Waves L3, puncher, multiband compressor, RVerb, stock delay etc... Although it may sound "modest" to anyone who is into mixing, I must say that the gear is nothing without the man. Lucky for us we have a true guru here who doesn't need to use anything that he doesn't have too smile.gif
Whole album takes about 18GB of storage, which is relatively small. Drum, bass, guitar, vocals @ 96/24 wave files and that's pretty much it.
Here's the original file I sent to tony. Bare in mind our sound engineer mastered this track using 4 (basic) software plugins on the main stereo out bus. This is the version with those plugins bypassed.

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Attached File  Nocni_gruv.mp3 ( 3.58MB ) Number of downloads: 95
 


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Staffy
post Feb 16 2010, 06:28 PM
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QUOTE (Ivan Milenkovic @ Feb 16 2010, 06:07 PM) *
Good points here. In this case all the plugins are capable of 96/24, and the truth is there aren't that many "bells and whistles" in these. Waves L3, puncher, multiband compressor, RVerb, stock delay etc... Although it may sound "modest" to anyone who is into mixing, I must say that the gear is nothing without the man. Lucky for us we have a true guru here who doesn't need to use anything that he doesn't have too smile.gif
Whole album takes about 18GB of storage, which is relatively small. Drum, bass, guitar, vocals @ 96/24 wave files and that's pretty much it.
Here's the original file I sent to tony. Bare in mind our sound engineer mastered this track using 4 (basic) software plugins on the main stereo out bus. This is the version with those plugins bypassed.


Suddenly, this thread makes sense!!! Thx Ivan! And I must say that You have a great job on this one Tony, the mastering sounds much more "open" without beeing too maxed out. Here it sounds very good with a "live" sound that I like.

Nah, the gear is nothing worth if the tech doesn't know what he is doin.... And if the musicians are good, and there are a good sound in their instruments, there isn't much to do! smile.gif You dont really need those bells and whistles to something anyway!

//Staffay


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tonymiro
post Feb 16 2010, 07:03 PM
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QUOTE (Ivan Milenkovic @ Feb 16 2010, 06:07 PM) *
... I must say that the gear is nothing without the man...


I think Bob Ohlson (who is one of the big Motown MEs) generally says that a good engineer will get a good result using poor equipment whilst a poor engineer will get poor results using amazing equipment. I would add to that is it far easier to master a well mixed tune then a poor one - and Ivan's engineer did a good job on the tracking and mixing smile.gif .


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Staffy
post Feb 16 2010, 07:59 PM
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QUOTE (tonymiro @ Feb 16 2010, 07:03 PM) *
I think Bob Ohlson (who is one of the big Motown MEs) generally says that a good engineer will get a good result using poor equipment whilst a poor engineer will get poor results using amazing equipment. I would add to that is it far easier to master a well mixed tune then a poor one - and Ivan's engineer did a good job on the tracking and mixing smile.gif .


Yeah, that was exactly was I meant! Its a much harder job to get something sounding decent when there is bad sound/playing from the beginning. I've had some nightmares bout that by the time I was a studio owner in the 80'ths. We had one reggae guy who's supposed to sing a tune, and he sang it a quarter up from the original key - eg. when it was Cminor he sang in Fminor. So what did I do? I transferred the whole song to Fminor and told him to sing like he did before resulting that he sang in Bb-minor instead... and on top of that, he was out of tune... OMG. !!!! laugh.gif
But strange enough he had a minor success in Lebanon with that song (he was from there), I still dont get it.... tongue.gif

//Staffay


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tonymiro
post Feb 16 2010, 08:15 PM
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QUOTE (Staffy @ Feb 16 2010, 07:59 PM) *
Yeah, that was exactly was I meant! Its a much harder job to get something sounding decent when there is bad sound/playing from the beginning. I've had some nightmares bout that by the time I was a studio owner in the 80'ths. We had one reggae guy who's supposed to sing a tune, and he sang it a quarter up from the original key - eg. when it was Cminor he sang in Fminor. So what did I do? I transferred the whole song to Fminor and told him to sing like he did before resulting that he sang in Bb-minor instead... and on top of that, he was out of tune... OMG. !!!! laugh.gif
But strange enough he had a minor success in Lebanon with that song (he was from there), I still dont get it.... tongue.gif

//Staffay


laugh.gif

- Nowadays thank god for autotune/melodyne cool.gif.

I lost track of the number of drummers who had problems counting to four wink.gif .

'So you come in on the count of four ok? One, two, th..'

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Ivan Milenkovic
post Feb 16 2010, 10:12 PM
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I think frustrated singers must have developed melodyne. If they didn't, they should give their respect to the the makers of this fine piece of software - it saved them a lot of money around the world! smile.gif


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Staffy
post Feb 16 2010, 11:32 PM
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QUOTE (Ivan Milenkovic @ Feb 16 2010, 10:12 PM) *
I think frustrated singers must have developed melodyne. If they didn't, they should give their respect to the the makers of this fine piece of software - it saved them a lot of money around the world! smile.gif


.... as well as giving us a whole bunch of "singers" that can't even perform "Twinkle, Twinkle little star" live without even their mothers crying for mercy ...... laugh.gif laugh.gif laugh.gif


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Bogdan Radovic
post Feb 17 2010, 12:19 AM
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Elysia sounds great. I must also note that this mastering sounds pretty close to our band's idea for the songs and style we did with our producer. So without consultation - Tony captured the "idea/feel" so well with cool results! smile.gif

Thanks Tony! smile.gif

This post has been edited by Bogdan Radovic: Feb 17 2010, 12:19 AM


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