This was super helpful bro. I really appreciate you taking the time to explain what was going on in great depth. I feel lucky to have found your channel tonight. So many helpful tips to help get us non-sound engineers to a good spot. LOL For someone new to production (and I mean 1 year new so I'm still learning ALOT!)
what a crazy tutorial you just opened my eyes to a punch of things that i never thought about and some plugins which i never thought i would use them , now i think my mastering mentality had changed a lot , thank you so much for sharing these tips
thanks for sharing all these tips 🫶 a video about giving sounds more body would be great - for example how to add more body to a snare or to a synth lead sound that has good high frequencies but that misses some energy in mid / low mid frequencies
Thanks so much for this great tipps! Your tutorials are helping me a lot improving my mixing and master skills. Also your masterclass was amazing! Please more of theese types of vids. ✨🙏 Best
Dig this- Is this your normal process, or you normally using more analog gear in your masters? Would be sweet to see a video (or a bunch Lol) of you doing a mostly analog master.
Exactly, so in a normal mastering session I work with tons of analog mastering gear. Great idea for more videos on that! This video is very much a demonstration of some techniques that work in the box for achieving loudness.
cool video! are you able tell us the difference between the hardware ngbus comp and a digital compressor. like does it add depth and hardmonic distortion that difital cant do? thanks in advance if you are able to answer:)
Great to hear! Absolutely. The ngBus is a beast. Next to its compression qualities being superior to any SSL type compression plugin, it’s able to operate in M/S and has a THD control for added harmonics. Keep in mind that a plugin is digital, so an analog unit in essence operates differently. Love the ngBus!
Brilliant! Once you have it all set, if you find it a little too much, you could just pull back the input 1db or so from the first input. Giving less of the whole chain affect?
Hi Bram, thanks for sharing your experience and information. I assumes that you have had the opportunity to compare various studio monitors suitable for Top Mix/MST work (Pop & Dance music). I am mainly interested in the comparison between PMC 6-2 vs HEDD Type 20 MK2 (if you have any experience). Thanks for your reply, MIIlO
How can you clip in plugins in a 32 bits floating point DAW ? Do you have any technical knowledge about digital audio ? I am refering to "gain staging" part of your video.
Absolutely. However, some plugins react differently when pushed harder. I’m not referring to digital clipping, since the DAW indeed operates at 32-bit floating point.
@@thismixissick Well at 3:30 you explicitly said " Going way too hard into plugins, unintentionally CLIPPING in a lot of places, is what you want to avoid because all these ARTIFACTS and DISTORTION adds up..." If this is not referrig to digital clipping, then what is it ? I like your content, and it's alright to make mistakes, but just be honest with your audience and make a correction in futur videos instead of twisting your own words to negate the obvious.
The most simple answer to your questions is DBFs. DAW-s are projected in the way of creating some limit point in order to achive maximum digital clean loundness. So if you push your signal above 0 DBFs your signal will start to clipping not because our computers can't handle this operation (we have 32 floating point) but becouse we have to find some reference point and make decisions about it. I will give you an example a you can try it on your own. 1. Run your session at 32-floating point 2. Take any sound and push it above 0 DBFs 3. I will start clipping 4. Export your file with 32-bit floating 5.Open new 32-bit DAW session 6. Import your previously clipped sound of choice 7. Normalize it to level under the 0DBFs 8.If you did all steps correctly, your audio will not clip anymore. Why? Cuz the reason I have said in the beginning. The DAW and audio programmers have to find a solution to manipulate audio it to export it to analog driven world with the limitation of sound physics and quality of audio drivers (speakers). And in real world our speakers have audio translation limitations. In our DAWs we just sets some abstract reference points to operate and manipulate audio. And then 8bit is not enough data to smoothly operate audio manipulation without clipping and distortion. 16 bits are enough but still (CD-R are on them) they can lack flexibility when manipulation of audio. Then we have 24bit which are more than enough to manipulate audio and finnaly 32bit is just pure freedom do do what you what to do. Remember only that our software is programmed relatively to some reference points and in our music production case is our DB full scale where digital Lourdes is measured from (-infinity to 0) from silence to maximum potential loundes of signal without distortion due to lack of source data to write.
@@jeanpaul.mma_ The clipping you are refering to is caused by getting over 0 dbfs when bit resolution is reduced to 24 or 16 bits. This is not what he was talking about since he is using true peak limiting in a mastering context. He was talking about getting over 0 dbfs in multiple process (EQ COMPRESSION SATURATION ETC..) in the 32 bit floating domain before hitting the limiter and before reducing the bits depth, thinking it would cause distortion and artifacts. That's simply false statement.
This was super helpful bro. I really appreciate you taking the time to explain what was going on in great depth. I feel lucky to have found your channel tonight. So many helpful tips to help get us non-sound engineers to a good spot. LOL For someone new to production (and I mean 1 year new so I'm still learning ALOT!)
That’s great to hear. Keep going!
what a crazy tutorial you just opened my eyes to a punch of things that i never thought about and some plugins which i never thought i would use them , now i think my mastering mentality had changed a lot , thank you so much for sharing these tips
That’s great to hear man! Truly is. Thanks for sharing, and have a good time mastering!
Bam is one of the best in the business. Great advice as always 👌🏼😎
Thanks for kind words Mark!
thanks for sharing all these tips 🫶 a video about giving sounds more body would be great - for example how to add more body to a snare or to a synth lead sound that has good high frequencies but that misses some energy in mid / low mid frequencies
Great idea. Thank you!
Thanks so much for this great tipps! Your tutorials are helping me a lot improving my mixing and master skills. Also your masterclass was amazing! Please more of theese types of vids. ✨🙏
Best
That’s great to hear, and exactly the aim! Love pushing this out for you.
great Video man!!!😊
Great to hear!
Siiick video
Great to hear!
Dig this- Is this your normal process, or you normally using more analog gear in your masters? Would be sweet to see a video (or a bunch Lol) of you doing a mostly analog master.
Exactly, so in a normal mastering session I work with tons of analog mastering gear. Great idea for more videos on that! This video is very much a demonstration of some techniques that work in the box for achieving loudness.
cool video! are you able tell us the difference between the hardware ngbus comp and a digital compressor. like does it add depth and hardmonic distortion that difital cant do? thanks in advance if you are able to answer:)
Great to hear! Absolutely. The ngBus is a beast. Next to its compression qualities being superior to any SSL type compression plugin, it’s able to operate in M/S and has a THD control for added harmonics. Keep in mind that a plugin is digital, so an analog unit in essence operates differently. Love the ngBus!
Good Video, def make some more, Good Job! How did you setup the outboard?
Great to hear! More are absolutely coming your way. Excellent idea for a future video!
legend ! 🔥
Great that you’ve enjoyed it!
Brilliant! Once you have it all set, if you find it a little too much, you could just pull back the input 1db or so from the first input. Giving less of the whole chain affect?
Exactly!
Hi Bram, thanks for sharing your experience and information. I assumes that you have had the opportunity to compare various studio monitors suitable for Top Mix/MST work (Pop & Dance music). I am mainly interested in the comparison between PMC 6-2 vs HEDD Type 20 MK2 (if you have any experience). Thanks for your reply, MIIlO
Absolutely. I initially wanted to go for ATC, until I heard PMC. Yes, the HEDDs are fantastic, yet don’t come close to the PMC’s.
@@thismixissick Thanks a lot, see y'soon
thank you
Welcome!
Bro! Where's the studio tour? I will goto every video of yours and ask for a studio tour 🤪
Excellent idea for a future video!
Thx for this
You’re super welcome!
Good
Happy to hear so!
How can you clip in plugins in a 32 bits floating point DAW ? Do you have any technical knowledge about digital audio ? I am refering to "gain staging" part of your video.
Absolutely. However, some plugins react differently when pushed harder. I’m not referring to digital clipping, since the DAW indeed operates at 32-bit floating point.
@@thismixissick Well at 3:30 you explicitly said " Going way too hard into plugins, unintentionally CLIPPING in a lot of places, is what you want to avoid because all these ARTIFACTS and DISTORTION adds up..."
If this is not referrig to digital clipping, then what is it ? I like your content, and it's alright to make mistakes, but just be honest with your audience and make a correction in futur videos instead of twisting your own words to negate the obvious.
The most simple answer to your questions is DBFs. DAW-s are projected in the way of creating some limit point in order to achive maximum digital clean loundness. So if you push your signal above 0 DBFs your signal will start to clipping not because our computers can't handle this operation (we have 32 floating point) but becouse we have to find some reference point and make decisions about it.
I will give you an example a you can try it on your own.
1. Run your session at 32-floating point
2. Take any sound and push it above 0 DBFs
3. I will start clipping
4. Export your file with 32-bit floating
5.Open new 32-bit DAW session
6. Import your previously clipped sound of choice
7. Normalize it to level under the 0DBFs
8.If you did all steps correctly, your audio will not clip anymore. Why?
Cuz the reason I have said in the beginning.
The DAW and audio programmers have to find a solution to manipulate audio it to export it to analog driven world with the limitation of sound physics and quality of audio drivers (speakers).
And in real world our speakers have audio translation limitations. In our DAWs we just sets some abstract reference points to operate and manipulate audio. And then 8bit is not enough data to smoothly operate audio manipulation without clipping and distortion. 16 bits are enough but still (CD-R are on them) they can lack flexibility when manipulation of audio. Then we have 24bit which are more than enough to manipulate audio and finnaly 32bit is just pure freedom do do what you what to do. Remember only that our software is programmed relatively to some reference points and in our music production case is our DB full scale where digital Lourdes is measured from (-infinity to 0) from silence to maximum potential loundes of signal without distortion due to lack of source data to write.
@@jeanpaul.mma_ The clipping you are refering to is caused by getting over 0 dbfs when bit resolution is reduced to 24 or 16 bits. This is not what he was talking about since he is using true peak limiting in a mastering context. He was talking about getting over 0 dbfs in multiple process (EQ COMPRESSION SATURATION ETC..) in the 32 bit floating domain before hitting the limiter and before reducing the bits depth, thinking it would cause distortion and artifacts. That's simply false statement.
Thank you for providing this additional and valuable information for everyone!