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pcrecord 25 July 2019

Hi gang,

Like I do once or twice a year, I like to download pro recorded tracks and make the exercise of mixing a song. It gives an idea how different those tracks are from mine and also a degree of detachment I don't get from tracks I recorded myself.

If you want to join me, it would be fun to compare our results.
I chose well recorded tracks made with by the Telefunken team using their mics of course.

You can download the raw tracks here :
The song I chose is : Remember Jones - "I Smell Sex And Candy" (Cover)
https://www.telefunken-elektroakustik.com/live-from-the-lab-season-3
It's free, just enter your email address, I haven't received a publicity email yet ;)

You can post your mix or an extract here as an mp3, not a external link, that way they will stay available in this thread for future members.

While doing this, I will be posting videos of each steps and thought process of the mixing and mastering the song on my youtube channel, any comments are welcome. I guess the process will take a few weeks, so you got time to post your mixes :
This is part 1 session preparation ;

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Comments

pcrecord Thu, 08/15/2019 - 10:01

kmetal, post: 461943, member: 37533 wrote: Just wondering if you guys are using a mix bus compressor or eq?

You have everything I did in my video.. but since you asked.
Ozone, EQ, dynamic module (mutiband compressor) in M/S, the vintage compressor, the exciter, vintage tape.
Then my beloved Fabfilter PRO L2 ! ;)

Most plugins and modules barely touch the signal, it's very subtle, but it gets me a slighty larger sound with retaining most of the dynamics..
Well that's what I'm telling myself. what do you think ? :ROFLMAO:

kmetal Thu, 08/15/2019 - 10:55

miyaru, post: 461944, member: 49780 wrote: I use a Focusrite Red Compressor in the masterbus. NOthing fancy, came with my focusrite interface.

I was pleasantly surprised by the focusrite pluggin bundle. They are very usable. I liked the red comp when i messed around with it.

pcrecord, post: 461945, member: 46460 wrote: You have everything I did in my video.. but since you asked.
Ozone, EQ, dynamic module (mutiband compressor) in M/S, the vintage compressor, the exciter, vintage tape.
Then my beloved Fabfilter PRO L2 ! ;)

You did that on the mix bus? Then did a similar set of processing on the mastering too? Just trying to make sure i understand correctly.

pcrecord Thu, 08/15/2019 - 12:38

kmetal, post: 461946, member: 37533 wrote: I was pleasantly surprised by the focusrite pluggin bundle. They are very usable. I liked the red comp when i messed around with it.
You did that on the mix bus? Then did a similar set of processing on the mastering too? Just trying to make sure i understand correctly.

Oh.. sorry, I missread.. only on the mastering project..

While mixing I had L2 but I removed it before exporting for the mastering ;)

miyaru Thu, 08/15/2019 - 14:52

I use a compressor on the master bus quite often, it glues the song together. Just a little touch of compression threshold on -10 Db, slightly fast attack and a ratio of 1:1.7 or so...... finetuned to the song of course.

If I master I like to use multiband compressors, but I don't have a good one right now. I useally don't really master, just pseudo LOL. I think it is a trade by itself, with it's own tools and skills.

bouldersound Thu, 08/15/2019 - 19:47

Okay, I did a Mix 3. I was just going to touch up the mastering (which I'm doing in the same session as the mixing), but then I found some problems with my Mix 2 that I wanted to fix. Pretty soon I had heavily altered it so I just committed to a third mix.

[MEDIA=audio]https://recording.o…

Attached files Remember Jones_I Smell Sex and Candy_Bouldersound mix 3.mp3 (3.1 MB) 

DogsoverLava Sat, 08/17/2019 - 11:54

Here's my MasteredV1 version of my V3 mix. I tried to tighten up the bass a bit by focusing the attack to make it slightly more percussive, I also tried a little bit of saturation to warm it up a wee bit, and of course applied a pretty inelegant limiter that I tried to boost overall while taming some of the excited areas that might have been too aggressively dynamic. I like it but my objectivity between mixing and mastering is still something I have not developed.

[MEDIA=audio]https://recording.o…

Attached files sex and candy masterv1.mp3 (8 MB) 

pcrecord Mon, 08/19/2019 - 10:26

DogsoverLava, post: 461972, member: 48175 wrote: Here's my MasteredV1 version of my V3 mix. I tried to tighten up the bass a bit by focusing the attack to make it slightly more percussive, I also tried a little bit of saturation to warm it up a wee bit, and of course applied a pretty inelegant limiter that I tried to boost overall while taming some of the excited areas that might have been too aggressively dynamic. I like it but my objectivity between mixing and mastering is still something I have not developed.

[MEDIA=audio]https://recording.o…

To be honest without being disrespectfull. I think it's a bit heavy on the low end. could use more clarity.. to my taste anyway.. ;)

bouldersound Mon, 08/19/2019 - 11:00

pcrecord, post: 461910, member: 46460 wrote: All right, here is the complete song mastered at -14LUFS
The video of the making is 2 posts up ;)

[MEDIA=audio]https://recording.o…

What kind of metering do you use? Is it built into your limiter or do you use something separate?* What are your thoughts on intersample peaking? (I suppose you may address this in the video, which I'll have to watch, but I thought it would be interesting to discuss it here.)

I use the free Youlean meter. I maintain the headroom at 1dB to prevent intersample peaks exceeding 0dBFS.

*[Edit] Watched the video and saw that you use the meter built into Pro-L 2.

DogsoverLava Mon, 08/19/2019 - 16:33

pcrecord, post: 461995, member: 46460 wrote: To be honest without being disrespectfull. I think it's a bit heavy on the low end. could use more clarity.. to my taste anyway.. ;)

Thanks PC - I’m happy to get any kind of feedback from you - got lots of respect for what you and others on this forum can do so the only way I can get better and learn is if the criticism I get is delivered really directly and bluntly. I’ll dig into it more.

kmetal Tue, 08/20/2019 - 16:27

bouldersound, post: 461996, member: 38959 wrote: What are your thoughts on intersample peaking?

This is a good topic. I used to use waves L2 maximizer set to -.3dbfs instead of -.1 because its supposed to reduce intersample clipping. I havent had a proper chance to use my ozone and fabfilter stuff yet.

I honestly dont know what to listen for when it comes to intersample peaking. What are the audibl effects.

I think this is an important topic for mix and master.

bouldersound Tue, 08/20/2019 - 17:05

kmetal, post: 462000, member: 37533 wrote: This is a good topic. I used to use waves L2 maximizer set to -.3dbfs instead of -.1 because its supposed to reduce intersample clipping. I havent had a proper chance to use my ozone and fabfilter stuff yet.

If I understand it correctly, the EBU R128 standard specifies -23dBLUFS, which is much lower than the usual for music streaming. It also specifies -1dBTP ("true peak"), which might seem a bit overly cautious until you see what data compression can do to an audio file. I generally master to -14dBLUFS and -1dBTP.

kmetal, post: 462000, member: 37533 wrote: I honestly dont know what to listen for when it comes to intersample peaking. What are the audibl effects.

That's the thing, it may or may not have an audible effect depending on the DAC in question and how far above 0dBFS the intersample peaking goes. Some DACs have headroom to accommodate ISP and some don't, which I think is a legacy of the Red Book standard and early digital formats. You probably aren't likely to get ISP from a properly digitized analog signal, but you can get it from a digitally processed signal. Or something like that. I suspect ISP sounds different on different DACs. Most likely it sounds like some variation on clipping.

My approach is to be a bit overcautious and leave a full dB of true peak headroom on my uncompressed file. Then it should be nearly impossible for data compression (e.g. mp3 encoding) to cause it to go above 0dBTP.

Boswell Wed, 08/21/2019 - 05:33

The only place that an inter-sample over-limit can occur is in the analogue reconstruction filter in a DAC. Early in the advent of CD players, there were some models that could overload because the designer had not left enough headroom on the output amplifiers. This applied not only to the Walkman type, where battery consumption and maximising output sound level were both primary concerns, but also (embarrassingly) in certain hi-fi component CD players, where there was no excuse. I was involved at the time in generating "problem" CDs and testing them on commercial players, looking for waveform flat tops on an oscilloscope and corresponding abnormalities on a distortion measurement system.

When it comes to DAW processing at a fixed sample rate, it will not happen, since inter-sample levels never exist inside a DAW. However, you can route tracks that get near 0dB to effect modules that may then give overrange outputs, but in fixed-wordlength coding, it's the job of the module to report that samples in the effected result cannot be represented in the selected wordlength.

pcrecord Wed, 08/21/2019 - 06:00

Boswell, post: 462012, member: 29034 wrote: The only place that an inter-sample over-limit can occur is in the analogue reconstruction filter in a DAC. Early in the advent of CD players, there were some models that could overload because the designer had not left enough headroom on the output amplifiers. This applied not only to the Walkman type, where battery consumption and maximising output sound level were both primary concerns, but also (embarrassingly) in certain hi-fi component CD players, where there was no excuse. I was involved at the time in generating "problem" CDs and testing them on commercial players, looking for waveform flat tops on an oscilloscope and corresponding abnormalities on a distortion measurement system.

When it comes to DAW processing at a fixed sample rate, it will not happen, since inter-sample levels never exist inside a DAW. However, you can route tracks that get near 0dB to effect modules that may then give overrange outputs, but in fixed-wordlength coding, it's the job of the module to report that samples in the effected result cannot be represented in the selected wordlength.

So do you think this can still happens to modern CD players ?
Do you think it's still relevant to lower the final ouput half or 1 db quieter to be safe ?

Boswell Wed, 08/21/2019 - 07:14

I think that, many years ago, when we exposed this problem, the perpetrators took notice, so that modern designs are proof against it. In addition, most modern DACs perform their reconstruction and sinc (sin(x)/x) compensation in the digital domain and require only low-pass filtering in the analogue domain.

It's not something I bother to check in my own mixes these days, but then I never let peak values get to 0dBFS. Depending on the material and when the peaks happen, I am usually happy with 0.5dBFS or even closer. I've also not seen any studies showing what happens to occasional full-scale peaks in mixes that are subsequently processed to LUFS standards. They are supposed to be preserved if the average level is -14dBFS or below.

bouldersound Wed, 08/21/2019 - 11:05

What I read, and this may not be correct, is that having headroom beyond 0dBFS was actually a violation of the Red Book specification. Not that they wanted to cause a problem but that it was never considered at the time that signals would be processed as they are these days. The output voltages for given dBFS levels were specified in a way that didn't account for clipped digital signals that would be reconstructed with peaks above the originally specified peak voltage.

Boswell Wed, 08/21/2019 - 15:29

For "Red Book Specification" you could say "the laws of physics", since there is no way of storing over-range numbers on a digital medium. However, that assumes that all the numbers are independent, which is not the case when representing digitised sounds, leaving room for some exploitation.

There are two ways this anamoly can show up: one intentional and the other unintentional. To take the latter first, there is no specification in the Red Book about the exact form the DAC reconstruction filter should take, as long as it meets certain requirements. Different implementation of these filters can lead to subtly different characteristics while still meeting the prescribed requirements, and one of these differences is overshoot. It's not really relevant as long as the reproduction system can correctly handle the result, which was one of the characteristics I was looking into all those years ago in the early CD players.

The second aspect is the intentional one, and that is deliberately tweaking the recorded samples such that any type of reconstruction filter would cause the output to go over range. There were several rock bands that made use of this technique to gain even more points in the loudness wars. For obvious reasons I won't go into the details on a public forum of how it can be done, but the difference was very audible.

bouldersound Wed, 08/21/2019 - 23:24

I was referring to what dBFS values were specified to correspond to what voltages, which is arbitrary. If the specification includes something like "0dBFS shall produce 1.414v" and something like "the output shall not exceed 1.414v" then it wouldn't be surprising to have a some DACs that don't behave well with digital signals that reconstruct to more than 1.414v. But that's just speculation on my part.

My habit used to be to master things to -.3dBFS peak, but I found that peaks could rise when a file was compressed so I've shifted to a full -1dBTP ("true peak"), which happens to be the EBU R128 specification.

audiokid Thu, 08/22/2019 - 09:06

bouldersound, post: 462019, member: 38959 wrote: I was referring to what dBFS values were specified to correspond to what voltages, which is arbitrary. If the specification includes something like "0dBFS shall produce 1.414v" and something like "the output shall not exceed 1.414v" then it wouldn't be surprising to have a some DACs that don't behave well with digital signals that reconstruct to more than 1.414v. But that's just speculation on my part.

My habit used to be to master things to -.3dBFS peak, but I found that peaks could rise when a file was compressed so I've shifted to a full -1dBTP ("true peak"), which happens to be the EBU R128 specification.

I've found this as well. I always put a limiter on to stop this. Sequoia's or Fabfilter's limiter work well for me.

kmetal Thu, 08/22/2019 - 13:39

bouldersound, post: 462001, member: 38959 wrote: If I understand it correctly, the EBU R128 standard specifies -23dBLUFS, which is much lower than the usual for music streaming. It also specifies -1dBTP ("true peak"), which might seem a bit overly cautious until you see what data compression can do to an audio file. I generally master to -14dBLUFS and -1dBTP.

That's the thing, it may or may not have an audible effect depending on the DAC in question and how far above 0dBFS the intersample peaking goes. Some DACs have headroom to accommodate ISP and some don't, which I think is a legacy of the Red Book standard and early digital formats. You probably aren't likely to get ISP from a properly digitized analog signal, but you can get it from a digitally processed signal. Or something like that. I suspect ISP sounds different on different DACs. Most likely it sounds like some variation on clipping.

My approach is to be a bit overcautious and leave a full dB of true peak headroom on my uncompressed file. Then it should be nearly impossible for data compression (e.g. mp3 encoding) to cause it to go above 0dBTP.

This is interesting stuff. It makes me wonder how much dimminishing returns there are when making different masters optimized for different delivery formats.

Theres EBU, digital delivery in flac, wav, and mp3, vinyl, cd, hi res digital, then streaming, and probably more i cant think of.

Im not sure there has ever been more delivery formats at once in history. Maybe the closest was vinyl, 8 track, cassette, and radio mixes/masters at one very short crossover period?

These types of things like ISP facinate me since they can be one of those "cant put my finger on it, but something aint right here" kind of things. I bet its also something a dedicated mastering engineer is alot more cognizant of than an average joe like me.

(Edit i posted this before refreshing my browser and seeing the 7 replies i havent read yet)

Smashh Mon, 08/26/2019 - 06:38

Great stuff guys . I struggle to get clarity and I am starting to think that my DA converter is not helping.
I really need to experiment more with the plug ins that I have .

Can I compensate by using plug ins to create an exceptable clarity ?
Or . will I never get clarity with this DA ( studio live 16.0.2 )
Excuse my ignorance , but Im trying to understand the process better .
Thanks for the great insights

kmetal Mon, 08/26/2019 - 16:44

Generally using less processing is a way to maintain clarity. The most transparent 3rd party pluggins ive used are fabfilter and ozone, they may be worth a try. Since the DA is partly responisible what your hearing from your daw/plugins, no pluggins can compensate for the DA, everything passes thru the DA.

As far as the DA, the question becomes which of these is your weakest link- the room, the monitors, or the DA? Id classify the Studiolive as 'decent', and my general guess without knowing what your using is the room or speakers are probably lying to you more than your DA.

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