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You do the master, and the the radio stations add their own flavor. What do you think?
Old 25th March 2021 | Show parent
  #31
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looks like you misread me on purpose, didn't get my analogy or i cannot get my message across...

(imo it ain't necessary to fully use all functions of a broadcast processor to get a somewhat more informed idea on what's happening when tracks are getting aired: it's enough to emulate to most crucial aspects.

same when mixing for pa: it's enough to mix on hornloaded system, maybe using woofers of roughly similar sizes - no one needs to mix on a large line array 100m away from the system.

in both cases: if you can, even better but i doubt anyone around here can do so very easily!)
Old 25th March 2021 | Show parent
  #32
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Quote:
Originally Posted by deedeeyeah ➡️
looks like you misread me on purpose, didn't get my analogy or i cannot get my message across...

(imo it ain't necessary to fully use all functions of a broadcast processor to get a somewhat more informed idea on what's happening when tracks are getting aired: it's enough to emulate to most crucial aspects.
I got that and disagree. You don't seem to understand what a broadcast processor does.
Quote:
Originally Posted by deedeeyeah ➡️
same when mixing for pa: it's enough to mix on hornloaded system, maybe using woofers of roughly similar sizes - no one needs to mix on a large line array 100m away from the system.

in both cases: if you can, even better but i doubt anyone around here can do so very easily!)
I find the analogy pointless, sorry. Its like you think that a speaker type is the thing that matters, and I couldn't disagree more.

In the film industry they mix in a theater called a Dubbing Stage, which is an actual theater of medium size. The difference between a Dubbing Stage and a commercial cinema is that the Dubbing Stage is properly maintained and working well, and calibrated to industry standards, and commercial cinemas don't maintain calibration or achieve the low noise figures. So, in many stages consoles you'll find various "monitor degrade" buttons. Some simulate bad theater EQ, some simulate more typical theater HVAC noise, and so on. They check their mixes because when you mix on a really excellent system in a quiet room, the mix turns out to be overly dynamic, and then when played in a typical cinema, important elements get lost.

There have been attempts to simulate the large Dubbing Stage in small systems, but only really one that was close enough to translate well.

So yeah, if you want to know how your stuff sounds in a specific situation, you have to replicate that situation.
Old 25th March 2021 | Show parent
  #33
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Quote:
Originally Posted by Trakworx ➡️
Yes they are a sort of an auto-mastering, but a pretty bad one in most cases. It's part of the compromises they make in order to overcome the various technical limitations of broadcasting.
I agree. Sames applies, if not even stricter, to vinyl production, club music (with a 15 year old DJ riding the EQ), or the end-user's ear buds, his speakers, acoustics, and and and.

This is mastering, finding the right compromise for your audio to survive this mess without too many creative drawbacks. Keeping it simple, dynamic and rather dry is likely the best way out - don't overprocess. Sometimes over-enthusiastic mastering is the only problem around.

Last edited by FabienTDR; 26th March 2021 at 03:55 AM..
Old 25th March 2021 | Show parent
  #34
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🎧 10 years
Quote:
Originally Posted by jaddie ➡️
.
I've been meaning to ask anyone lately, what is Stokke mask?

The Thimeo Stereo Tool (which costs under $100, right in my income bracket ) that was mentioned above has some features that seem to point in the direction of simulated broadcasting.

Do any of these features in your opinion seem like they could be used consistently to help predict masters broadcast via FM?

https://www.stereotool.com/documenta...m-transmitter/

FM Transmitter section
All the settings to generate a compliant FM signal.

All the settings that are related to the FM signal can be found here, such as pre-emphasis, stereo and RDS encoding, RDS texts, Stokkemask (ITU-R SM.1268) and ITU-R BS.412 compliance and multipath distortion protection.

FM processing is more difficult that normal processing. Because of Pre-emp you often loose highs when enabling the FM processing settings (Composite Clipping helps a lot against this), Stokkemask and multipath distortion protection can have some impact on the amount of stereo separation, and BS412 lowers the level a lot.

Some people prefer to use Stereo Tool only for processing and use a separate RDS and stereo coder. While this works and still gives a reasonable sound, it makes it impossible to use Composite Clipping which gives several dB's of extra headroom for mainly the high frequencies, and features such as multipath distortion protection. If the whole composite/MPX signal is generated from within Stereo Tool, the sound that can be generated is a lot better.

If you are planning to use Stereo Tool on an FM transmitter, please also read FM Output. If you have multiple transmitters and want to synchronize the sound between them, see Synchronize FM transmitters.
Old 25th March 2021 | Show parent
  #35
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Quote:
Originally Posted by traumerei1838 ➡️
I've been meaning to ask anyone lately, what is Stokke mask?
A standardized means of measuring the occupied bandwidth of a frequency modulated carrier, ITU-R SM.1268.
Quote:
Originally Posted by traumerei1838 ➡️
The Thimeo Stereo Tool (which costs under $100, right in my income bracket ) that was mentioned above has some features that seem to point in the direction of simulated broadcasting.

Do any of these features in your opinion seem like they could be used consistently to help predict masters broadcast via FM?

https://www.stereotool.com/documenta...m-transmitter/

FM Transmitter section
All the settings to generate a compliant FM signal.

All the settings that are related to the FM signal can be found here, such as pre-emphasis, stereo and RDS encoding, RDS texts, Stokkemask (ITU-R SM.1268) and ITU-R BS.412 compliance and multipath distortion protection.
StereoTool is an impressive bit of software, no question. You run into the same problem though when trying to use it for simulating FM. It produces a composite baseband signal for modulating a transmitter, and that signal is not directly monitorable. It has to be decoded, and de-emphasized first, and the easiest way to get there is to modulate a low power transmitter, then receive the signal with an FM Stereo receiver.

The one caution is, though the names of the processing sections (multi-band, composite clipper, etc.) are fairly universal, the actual functions differ significantly. It may be fine, but would be concerned about the StereoTool processes matching the big processors that everyone uses.
Quote:
Originally Posted by traumerei1838 ➡️
FM processing is more difficult that normal processing. Because of Pre-emp you often loose highs when enabling the FM processing settings (Composite Clipping helps a lot against this), Stokkemask and multipath distortion protection can have some impact on the amount of stereo separation, and BS412 lowers the level a lot.
Every country that has FM has the concern of keeping the stations output bandwidth within some sort of "mask" that defines the amplitude of sidebands vs frequency from the carrier. Technically, when you narrow the FM bandwidth, you reduce separation and increase distortion. However, the mask itself doesn't reduce high frequency response, stereo separation or increase distortion significantly. The mask is wide enough that good audio can pass quite well. It's there to protect other stations on adjacent frequencies. However, we run into problems when with we modulate the carrier very heavily with dense and clipped signals, which of clipped in the composite signal, can splash outside of the mask. Composite clippers are a very old and crude way to control peaks without as much dulling, but they cause lots of other problems. The big processors have moved quite a ways past composite clippers and still achieve loud and reasonably bright response.
Quote:
Originally Posted by traumerei1838 ➡️
Some people prefer to use Stereo Tool only for processing and use a separate RDS and stereo coder. While this works and still gives a reasonable sound, it makes it impossible to use Composite Clipping which gives several dB's of extra headroom for mainly the high frequencies, and features such as multipath distortion protection. If the whole composite/MPX signal is generated from within Stereo Tool, the sound that can be generated is a lot better.
You can use an external composite clipper after StereoTool and before the transmitter. They used to be marketed as stand-alone boxes. CRL was a big manufacturer of them in them, as was Modulation Sciences in days when composite clipping was a new idea. There's a Modulation Sciences composite clipper on eBay right now.

A couple of words about multipath distortion. While careful control of the L-R subchannel can help a bit, the real culprit in transmitter-aggravated multipath is a function called Synchronous AM, an amplitude modulation component synchronous with the FM component. Its a bit difficult to understand, but if you took a carrier with a -40dB Synch AM component, and got into a multipath reflection situation where the two signals were skewed in time the equivalent of 180 degrees of a modulating audio signal, the two carriers (the direct and reflected) sum in way that the signals begin to cancel, but the AM vectors add, which increases the sync AM by many times, up to and over 100%. When this happens, the receiver can't maintain lock on the carrier, and huge amounts of distortion crop up. The fix is to transmit a carrier with as little sych AM as possible. This is done with wide-band RF stages, and wide-band, well-matched antennas and transmission lines. Meeting actual RF mask is less important to multipath, and limiting the subcarrier only hides the effect. You can't really pre-process out multipath, but you can transmit a clean RF signal with minimal sync AM and significantly reduce the effects.
Old 26th March 2021 | Show parent
  #36
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Quote:
Originally Posted by jaddie ➡️
I got that and disagree. You don't seem to understand what a broadcast processor does.
lol - yeah, sure, i got no idea: that's why i get to design, install, operate and fix radio stations across the globe (well, except north america and antartica) since 1978!

Quote:
I find the analogy pointless, sorry.
only to come up with an analogy yourself?!

do you get the point about an analogy, approximation or emulation at all?

Quote:
Its like you think that a speaker type is the thing that matters, and I couldn't disagree more.
well, they do matter: experienced ears can differentiate between different type of speakers even if they measure the same in the frequency domain, have mostly similar dispersion and are driven at identical spl levels... - i assume you are neither mastering nor mixing, certainly not live then?

Quote:
In the film industry they mix in a theater called a Dubbing Stage, which is an actual theater of medium size. The difference between a Dubbing Stage and a commercial cinema is that the Dubbing Stage is properly maintained and working well, and calibrated to industry standards, and commercial cinemas don't maintain calibration or achieve the low noise figures. So, in many stages consoles you'll find various "monitor degrade" buttons. Some simulate bad theater EQ, some simulate more typical theater HVAC noise, and so on. They check their mixes because when you mix on a really excellent system in a quiet room, the mix turns out to be overly dynamic, and then when played in a typical cinema, important elements get lost.

There have been attempts to simulate the large Dubbing Stage in small systems, but only really one that was close enough to translate well.
i admit the mixing for commercial film release is the area in which i'm not much experienced so i cannot comment.

Quote:
So yeah, if you want to know how your stuff sounds in a specific situation, you have to replicate that situation.
there it is again, arguing on absolute terms, lacking any understanding of how getting close can be good enough to make a more informed choice..

__

i suggest we leave it at that - good luck!
Old 26th March 2021 | Show parent
  #37
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Quote:
Originally Posted by deedeeyeah ➡️
lol - yeah, sure, i got no idea: that's why i get to design, install, operate and fix radio stations across the globe (well, except north america and antartica) since 1978!
It seems that neither of us are ready to accept the others experience and authority on this. I'm OK with letting the information in my posts be the validation, and I always try to include actual technical detail in my posts that can be easily fact-checked via other means. I avoid sweeping generalities without factual basis. I wan't someone to be able to take what I say, look into, say, an Omnia 9 or Optimod 8700i processor manual, and verify what I said to be true. Or even a vintage 8100 manual. Check to see if what I post matches reality or not.

You can choose your own means of validation. Just know, perhaps some do, but I don't readily accept hand-waving as validation of fact.

Quote:
Originally Posted by deedeeyeah ➡️
well, they do matter: experienced ears can differentiate between different type of speakers even if they measure the same in the frequency domain, have mostly similar dispersion and are driven at identical spl levels...
Correct, but irrelevant to the discussion.
Quote:
Originally Posted by deedeeyeah ➡️
- i assume you are neither mastering nor mixing, certainly not live then?
Your assumption is based on your disrespect for my opinion, without knowledge of what I do or have done. And is incorrect anyway, but I'm done blowing my own horn. But thanks for that disrespect. I guess we both do that.
Quote:
Originally Posted by deedeeyeah ➡️

there it is again, arguing on absolute terms, lacking any understanding of how getting close can be good enough to make a more informed choice..
Well, I'm an Engineer. Close can be good enough, but when someone's idea of "close" eliminates several major sound modifiers (pre-emphasis based HF limiting and clipping), and the desire is to get a preview of what broadcast systems, off of which use those heavily, will do to your mix, then I don't agrew with that concept is "close" or "good enough". Others can have different ideas, just don't expect me to agree. I will point out the flaws in the logic.
Quote:
Originally Posted by deedeeyeah ➡️

i suggest we leave it at that - good luck!
And wouldn't that be nice?
Old 26th March 2021 | Show parent
  #38
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Thank you so much for taking the time to write this out. Lots of good information to digest here. And I see why it's futile to emulate FM at the mastering stage.

Seems like you and deedeeyeah are debating like Westinghouse & Edison. Time for a duel!
Old 26th March 2021 | Show parent
  #39
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Quote:
Originally Posted by traumerei1838 ➡️

Seems like you and deedeeyeah are debating like Westinghouse & Edison. Time for a duel!
Funny! I'll be Westinghouse so I don't have to kill any animals.
Old 26th March 2021 | Show parent
  #40
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Quote:
Originally Posted by darkalex ➡️
Is there a way to intentionally simulate this compression?

Like adding this on music to get that radio rock sound even if those tracks never make it to the radio
Sure, but it only sounds right when you are hearing it in your car, at a minimum of 50 MPH.

Although, there may be a plugin that will simulate a 70dB noise floor, and then you'd be golden!

PS after re-reading the thread (which I tend to skip when the personal crap starts) it seems dubbing stage consoles already have something like this.
Old 27th March 2021 | Show parent
  #41
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Quote:
Originally Posted by edva ➡️
Sure, but it only sounds right when you are hearing it in your car, at a minimum of 50 MPH.

Although, there may be a plugin that will simulate a 70dB noise floor, and then you'd be golden!

PS after re-reading the thread (which I tend to skip when the personal crap starts) it seems dubbing stage consoles already have something like this.
A little back-story on this. In the early 1980s, a guy named Tom Holman joined Lucasfilm. His first big project was to work on improve theater sound, because George was so disappointed when he watched one of is movies in a commercial theater. The screening room/dubbing stage was the test bed for his research. The very short version...the result was the first set of "THX" theater specifications that included, among many other things, an acoustic noise figure that was challenging to meet. They went with that of course.

When mixing Return of the Jedi in 1983, in that now that very quiet THX room, they kind of goofed and mixed the dialog too low for the "Luke and Lea on the Bridge" scene. When it was tested in a common-man's theater, the dialog got buried under HVAC noise. They went back and re-mixed the film, now based on real-world theater noise floors.

The outgrowth was a noise generator made from a set of noise-source diodes made by Koep Precision (6 channels of it, IIR), and filtered appropriately, then mixed into the dubbing stage system at various preset levels to simulate normal (non-THX) theater noise floors. The noise generator was nicknamed "Sonic Perfume" because it covered a multitude of sins. They also worked on the upper end of the dynamic range because optical tracks on release prints had a hard limit when the galvanometer shutters creating the variable area track banged into each other, "shutter clash". I don't recall how that was simulated, but it was on the "monitor degrade" panel too.

Much of that work has propagated to other dubbing stages, now nearly 40 years down the road.
Old 27th March 2021 | Show parent
  #42
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Quote:
Originally Posted by jaddie ➡️
A little back-story on this. In the early 1980s, a guy named Tom Holman joined Lucasfilm. His first big project was to work on improve theater sound, because George was so disappointed when he watched one of is movies in a commercial theater. The screening room/dubbing stage was the test bed for his research. The very short version...the result was the first set of "THX" theater specifications that included, among many other things, an acoustic noise figure that was challenging to meet. They went with that of course.

When mixing Return of the Jedi in 1983, in that now that very quiet THX room, they kind of goofed and mixed the dialog too low for the "Luke and Lea on the Bridge" scene. When it was tested in a common-man's theater, the dialog got buried under HVAC noise. They went back and re-mixed the film, now based on real-world theater noise floors.

The outgrowth was a noise generator made from a set of noise-source diodes made by Koep Precision (6 channels of it, IIR), and filtered appropriately, then mixed into the dubbing stage system at various preset levels to simulate normal (non-THX) theater noise floors. The noise generator was nicknamed "Sonic Perfume" because it covered a multitude of sins. They also worked on the upper end of the dynamic range because optical tracks on release prints had a hard limit when the galvanometer shutters creating the variable area track banged into each other, "shutter clash". I don't recall how that was simulated, but it was on the "monitor degrade" panel too.

Much of that work has propagated to other dubbing stages, now nearly 40 years down the road.
Very interesting, thank you. "Sonic Perfume" - great name
Old 29th March 2021
  #43
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Michael Grafl's Avatar
 
🎧 10 years
Radiostations are often listened to in running cars and as a backdrop in busy areas, so it makes sense to use heavy processing to make sure quiet parts don't get lost in dynamic material, or that the loud parts aren't too jarring.

I don't think many people are listening to the radio expecting great sound quality. If I like a song on the radio I'll get the original recording and listen to that. I'm not a mastering engineer, but I don't see it bothering me if I were.
Old 29th March 2021
  #44
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Because it's frequency modulation (FM) broadcasters rely on the maximal average modulation to cover the largest possible area. The louder, the better. The second point is the sound signature of the radio station which is mostly done by exciter, stereo widening and bass enhancement. Any solid mastering can survive to this chalenge.

Last edited by Laurend; 29th March 2021 at 07:02 PM..
Old 29th March 2021 | Show parent
  #45
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Quote:
Originally Posted by Laurend ➡️
Because it's frequency modulation (FM) broadcasters rely on the maximal average modulation to cover the largest possible area.
Not exactly. There is a hard legal limit on maximum peak modulation, which is always the first concern. But, since this is a pre-emphasized system, raising the average implies drastic high frequency peak control, which is done with a high frequency limiter and post-pre-emphasis clipper. The result is always a degree of dulling as compared with the original pre-processing. There's no free lunch.

Anyone in a market with HD Radio can verify this by switching manually between FM and HD1 on a "loud" station. The only difference you should hear, assuming good reception, is that the HD1 has a cleaner, brighter, more open high end, though on some stations that don't get the transition right, you might her other artifacts. HD Radio is not pre-emphasized, so doesn't require special HF control or HF clipping.

Raising the average is important to keep the station competitive, but does not actually change coverage, which is a function of signal strength as presented to the FM detector (receiver) and its capture ratio. When you loose enough signal that the detector can achieve adequate quieting, it's just a dB or so away from inaudible.
Quote:
Originally Posted by Laurend ➡️
The louder, the better.
Unfortunately, mostly true. There are a number of exceptions in special formats like Jazz, Classical and public radio. And there's the occasional station that realizes they don't get more listeners by being louder (because everybody has a volume control!), but the get better TSL if the station is less fatiguing. TSL=Time Spent Listening, a metric reported by ratings services, and important for advertisers.
Quote:
Originally Posted by Laurend ➡️
The second point is the sound signature of the radio station which is mostly done by exciter.
Incorrect. The exciter is the device that generates the modulated FM signal. It's input is broadband, 0-100kHz, and it's output is usually broadband, certainly way beyond 200kHz. If the exciter is not ancient, broken or defective, it has the least impact of anything in the total chain. Even the RF power stage has more impact, often being bandwidth-limited.

Before the exciter is the Stereo Generator, which forms the multiplex baseband composite signal of Left plus Right, 19kHz pilot, 38kHz suppressed carrier with Left minus Right, plus whatever other signals above 50kHz the station might want to use. The stereo generator is also quite transparent, but because of the necessary pilot filtering, they are better if integrated with the final stages of the broadcast processor. The stereo generator itself, though, is quite neutral.
Quote:
Originally Posted by Laurend ➡️
stereo widening
Stereo widening always increases modulation in the 38kHz subcarrier because to add an sort of inter-aural crosstalk cancellation signal (that's how that works, btw) you also add much more to the difference of Left and Right, which universally aggravates multipath reception problems. Artificial widening is used very sparingly, if at all, because of this. Multipath reduction processing works with the subcarrier left-right signal to mitigate multipath degradation to some extent. Increasing the 38kHz subcarrier modulation simply adds to the total modulation, it's not free, and doesn't get you loudness.
Quote:
Originally Posted by Laurend ➡️
and bass enhancement.
Bass is processed by its own band (or two) in the multiband processor. Modern processors add "bass clipping" as a loudness tool, but it is then followed with a means of subtracting to some degree the more objectionable harmonics produced by bass clipping. Remember, this is a pre-emphasized system, and if you modulate at 100% below 500Hz, there's no headroom for anything higher.

Every time gain is adjusted by a processor, harmonic distortion is added, and IMD is increased. This happens because of the rapid attack and release time constants which tend to "track" the waveform, modifying it, and producing distortion. If you take two fast limiters, each producing distortion, and put them in a chain one after the other (like mastering followed by broadcasting), you get release times that are now twice as fast (they both release together) and the result is a doubling of harmonic and intermodulation distortions.

Each time a clipper threshold is exceeded and clipping happens, an extremely rapid increase in both types of distortion occurs.

You can hear the effects of aggressive loudness mastering on the air because bass notes can be heard to modulate high frequency notes. It's not just the fault of the broadcast processor or the mastering processing, but rather the chain of the two. But I'll warn you, once you've heard this, you can never ignore it again. It's one of the things that makes FM hard to listen to long-term.
Quote:
Originally Posted by Laurend ➡️
Any solid mastering can survive to this chalenge.
If you've read this far in the thread and still believe that, perhaps a careful re-read might help.
Old 30th March 2021
  #46
Gear Maniac
Very interesting thread.

Be careful when you use compression. Play a record anywhere out in the world, streaming services, online, terrestrial radio, anywhere, and it will be put through some form of compression.

Less is more is not, just an empty, old, saying.
Old 30th March 2021 | Show parent
  #47
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Quote:
Originally Posted by LLSentelle ➡️
Be careful when you use compression. Play a record anywhere out in the world, streaming services, online, terrestrial radio, anywhere, and it will be put through some form of compression.
Yes always be careful.

But streaming services don't put compression on tracks. Neither does satellite radio. Level matching tracks is not compression.

Just clearing that up for the zillionth time.

Cheers,
Old 31st March 2021 | Show parent
  #48
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Quote:
Originally Posted by Trakworx ➡️
Yes always be careful.

But streaming services don't put compression on tracks. Neither does satellite radio. Level matching tracks is not compression.

Just clearing that up for the zillionth time.

Cheers,
Actually, satellite radio does do a bit of compression as their big target market is listening in cars. It's not very aggressive though.
Old 31st March 2021 | Show parent
  #49
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Quote:
Originally Posted by jaddie ➡️
Actually, satellite radio does do a bit of compression as their big target market is listening in cars. It's not very aggressive though.
I actually tried to find info about that and couldn't find any - only stuff about data compression - so I concluded that they don't use dynamic compression.
Old 31st March 2021 | Show parent
  #50
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Quote:
Originally Posted by Trakworx ➡️
I actually tried to find info about that and couldn't find any - only stuff about data compression - so I concluded that they don't use dynamic compression.
Just because it's not published doesn't mean its not being done. It follows the radio model, processing for the destination, and competing with radio because that's where you most often find satellite radio...as a feature on a radio.

Radio stations never publish anything about their processing either, but we know it's being done. Satellite radio is not a pre-emphasized medium, so processing is easier, and can be lighter in the peak control area. But it absolutely is there.
Old 2nd April 2021
  #51
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🎧 5 years
Quote:
Originally Posted by UncleGroOve ➡️
Quick pop into the Mastering Forum to ask how do pro mastering engineers feel about radio stations that use processors like https://www.orban.com/optimodfm-8700i to "homogenize" the "sonic signature" of their broadcast. Does mastering become irrelevant at that point?
Do you instinctively compensate in your workflow to account for downstream processing?
Can we think of these processors be a sort of auto-mastering unit on their own?

Cheers,
Paul
Imho, this is one of the reasons why mastering becomes RELEVANT
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