Become a Site Supporter and Never see Ads again!

Author Topic: Audacity (osx): How to mix two AUD sources?  (Read 26476 times)

0 Members and 1 Guest are viewing this topic.

Offline justink

  • Trade Count: (4)
  • Taperssection All-Star
  • ****
  • Posts: 1803
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #90 on: September 07, 2017, 09:51:08 AM »
This thread is great - I spent an hour trying to line up a two hour matrix last night in 2.1.3, and figured out 3DP wasn't enough. Came here, downloaded 2.1.0, and I'm back on with the task.

Has anyone informed the Audacity team?

probably not.  we definitely need a fix.  i still can't figure it out.
Mics:
DPA 4028 (wide cards)
DPA 4023 (cards)
Earthworks TC25 (omnis) 

Pres and a/d's:
Oade ACM Grace Lunatec V3 (FOR SALE - Make Offer)
bm2p+ Edirol UA-5 (FOR SALE - Make Offer)

Recorders:
Oade CM Edirol R-44
Sony PCM‑M10

Offline Humbug

  • Trade Count: (5)
  • Taperssection All-Star
  • ****
  • Posts: 1979
  • Gender: Male
  • Humbug Hum Rejection Unit
    • Trading site
Re: Audacity (osx): How to mix two AUD sources?
« Reply #91 on: September 07, 2017, 02:48:39 PM »
Messaged the Audacity team on Facebook this evening; will post if I hear back.
UK based taper: SPCMC2(AT831)>CA9100 / PIPsqueak>iRiver H140/Nomad JB3
AT933(853C)>CA9100 / PIPsqueak>iRiver H140/Nomad JB3/iRiver H108 CF mod/Sony PCM-M10

http://www.archive.org/bookmarks/Humbug66

Offline JiB97

  • Trade Count: (12)
  • Needs to get out more...
  • *****
  • Posts: 2523
  • Gender: Male
    • My Archive Bookmarks
Re: Audacity (osx): How to mix two AUD sources?
« Reply #92 on: October 26, 2017, 07:49:22 PM »
posting in here so I can reference later.

been having issues recently combining sources; everything is coming out clipping even when both source are max peaking @ -4.  wish there was a program where I could tell it to have source A @ 60% and source B @ 40%...
AKG ck3/ck8 | c460b  + Naiant Actives | PFAs
Audio Technica u853r (omnis/mini-guns)
Tascam DR-70D

My Archive Links

Offline bvaz

  • Trade Count: (6)
  • Taperssection All-Star
  • ****
  • Posts: 1323
Re: Audacity (osx): How to mix two AUD sources?
« Reply #93 on: October 27, 2017, 07:06:24 AM »
posting in here so I can reference later.

been having issues recently combining sources; everything is coming out clipping even when both source are max peaking @ -4.  wish there was a program where I could tell it to have source A @ 60% and source B @ 40%...
I usually play the file in audacity at spots and look at the meters to see how high it is hitting before exporting.  not full proof, but it gives me an idea.

Offline Life In Rewind

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 843
Re: Audacity (osx): How to mix two AUD sources?
« Reply #94 on: October 27, 2017, 08:05:03 AM »
posting in here so I can reference later.

been having issues recently combining sources; everything is coming out clipping even when both source are max peaking @ -4.  wish there was a program where I could tell it to have source A @ 60% and source B @ 40%...

Percentages aren't very useful or meaningful...use your ears.

And yes - its normal to have higher peaks when combining sources.

Solutions are - record in 24 bit and don't run so hot. (like -8db instead of -4db)

Or - once you have your mix - use the faders (on the left) to reduce the gain by equal amounts on each file set.

Just nudge each one down by 1db until the peaks go away.

Then export!
« Last Edit: October 27, 2017, 08:07:24 AM by Life In Rewind »
Audix M1280/Avantone CK-1/APEX 435/Altec 626A/TEAC ME-120/Sony ECM-999PR/Sony ECM-MS5
TASCAM DR-70D/Tricorder

Offline morst

  • Archivist: Camper Van Beethoven & Cracker
  • Trade Count: (1)
  • Needs to get out more...
  • *****
  • Posts: 2760
  • Get in touch if you wanna record Cracker or CVB!
    • Soundscape Preservation Society
Re: Audacity (osx): How to mix two AUD sources?
« Reply #95 on: February 14, 2018, 02:11:40 PM »
posting in here so I can reference later.

been having issues recently combining sources; everything is coming out clipping even when both source are max peaking @ -4.  wish there was a program where I could tell it to have source A @ 60% and source B @ 40%...
I usually play the file in audacity at spots and look at the meters to see how high it is hitting before exporting.  not full proof, but it gives me an idea.
You can also just quickly render the peak area, so you don't have to wait for the whole thing to run, just to find out that the hot part went over!?

Run a test render, import it, if it peaks, lower everything & repeat. If the test doesn't peak, then render the whole thing, import and check for peaks!
Teams: Neumann, Bay Area Tapers, Multitrack, Pioneertown Tapers, Mac Geeks, Cassette Masters, Poster Collectors, Alumni of teams St Louis, Upper Midwest & Milwaukee / Southern Wisco

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 570
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #96 on: February 15, 2018, 12:20:59 PM »

been having issues recently combining sources; everything is coming out clipping even when both source are max peaking @ -4.  wish there was a program where I could tell it to have source A @ 60% and source B @ 40%...
I usually play the file in audacity at spots and look at the meters to see how high it is hitting before exporting.  not full proof, but it gives me an idea.
[/quote]
You can also just quickly render the peak area, so you don't have to wait for the whole thing to run, just to find out that the hot part went over!?

Run a test render, import it, if it peaks, lower everything & repeat. If the test doesn't peak, then render the whole thing, import and check for peaks!
[/quote]

It isn't brain surgery, it's post production.  Experiment and write down what you do.  After doing it a couple times, you will know where to have the levels on all tracks.  If it helps, the way that I find works best for me is to use my ears and eyes.  In that order!  I listen to determine which tracks I want louder...or possibly the same.  Once I figure that out, it's easy from there.  When initially recording the show, I record low (ALWAYS at 24bit)... ideally peaking around -12db or lower.  I raise the tracks (or lower them if the band got crazy loud or the board patch was unmanageable during the show) to have the mix I want.  An example, being the last show I did, the board patch was brought to -5.5db and the microphone tracks were brought to -8.5db.  I liked the mixed ratio with the soundboard being a little more dominant in this case, but it isn't always like that for me.  From there, I export and save my work.  In this stage, I have gotten used to where to adjust my levels to so they are not peaking when combined.  Now I have two files in that folder, the original (which I will never delete) and the work in progress.  From there, I basically start over, opening up a new Audacity window, and dropping that new file into it for further editing.  This is where I do the bulk of my post work.  If there are only a few spikes, I'll bring them down (by highlighting the spike, then hitting the "+ magnifier" symbol 9-10 times, I isolate the spike to the point that I'm ONLY reducing that one jump in volume).  Once I'm happy with that stage, I bring both tracks up to -0.50db (some prefer -1.0db, while others prefer 0.00db.  I say too each their own on that point).  From there, if I'm happy, I'm almost done...but let's be honest here, I'm never happy at that point!  I will look at spikes, if there are whistles or screams, etc. to see what else needs to be done.  It is at this stage that a normal human being will normalize or use other features to essentially compress the music and give a more uniform wave.  I do not do this.  Ever.  I like the dynamic range of the live show and do my best to keep the highs and lows, while only eliminating the spikes to bring up everything else to a good volume.  This may sound like normalizing or compressing, but when I reduce a dozen or so spikes, around 1/1000 sec each, throughout one set, in order to bring everything else up, it is very different.  Once I've finished this, I will individually bring up the two tracks to -0.50db if they aren't already there.  Once this is finished, I like to do a 10 second fade in and fade out before saving this now completed file.  I prefer to track out the songs in CDWave...
And that, boys and girls, is how I have figured out how to reduce my 4-5 channel recordings down to a two channel recording that I can burn to CD, share, email to the band, etc.  I hope my step by step is helpful (and that I didn't leave something out because I take it for granted). 
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline morst

  • Archivist: Camper Van Beethoven & Cracker
  • Trade Count: (1)
  • Needs to get out more...
  • *****
  • Posts: 2760
  • Get in touch if you wanna record Cracker or CVB!
    • Soundscape Preservation Society
Re: Audacity (osx): How to mix two AUD sources?
« Reply #97 on: February 18, 2018, 07:12:23 PM »
  From there, if I'm happy, I'm almost done...but let's be honest here, I'm never happy at that point!  I will look at spikes, if there are whistles or screams, etc. to see what else needs to be done.  It is at this stage that a normal human being will normalize or use other features to essentially compress the music and give a more uniform wave.  I do not do this.  Ever.  I like the dynamic range of the live show and do my best to keep the highs and lows, while only eliminating the spikes to bring up everything else to a good volume.  This may sound like normalizing or compressing, but when I reduce a dozen or so spikes, around 1/1000 sec each, throughout one set, in order to bring everything else up, it is very different.
Please correct me if I am wrong, but AFAIK, normalization is arithmetic and linear. Compression is decidedly NOT linear. They are very different.

My super-detailed approach involves making a test mix just a LITTLE hotter than I think I can get away with. I then import it, and find volume spikes that cause over-level peaks in the current render. I then lower those peaks manually, one at a time. I usually use the "amplify" plugin with a negative value like -1 or -3 dB. Once I have done this, I bounce a new mix and compare to make sure I got 'em all! If there are more than 10-100 peaks, I just master it a little lower, depending on how much work I want to put in for that last little bit of diminishing returns!!!
Teams: Neumann, Bay Area Tapers, Multitrack, Pioneertown Tapers, Mac Geeks, Cassette Masters, Poster Collectors, Alumni of teams St Louis, Upper Midwest & Milwaukee / Southern Wisco

Offline noahbickart

  • Site Supporter
  • Trade Count: (30)
  • Taperssection All-Star
  • *
  • Posts: 1861
  • Gender: Male
  • So now I wander over grounds of light...
Re: Audacity (osx): How to mix two AUD sources?
« Reply #98 on: February 19, 2018, 10:45:52 AM »
Peak Normalizing does not affect dynamic range.

“Eliminating peaks” = limiting. This is, by definition, a reduction of dynamic range.

I see no problem with either depending on the circumstances. But I know you are anti compression.

If so, why do the latter but not the former?
Recording:
Capsules: Schoeps mk41v (x2), mk4v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, Darktrain 2 and 4 channel KCY extensions:
Preamps:    Naiant Littlebox, Naiant IPA, Naiant PFA, Sound Devices Mixpre6
Recorders: Sound Devices Mixpre6, Sony PCM m10

Home Playback: Mytek DSD 192> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: Grace m903> AKG k701 / Hifiman HE-400

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 570
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #99 on: February 19, 2018, 12:56:38 PM »

Please correct me if I am wrong, but AFAIK, normalization is arithmetic and linear. Compression is decidedly NOT linear. They are very different.

My super-detailed approach involves making a test mix just a LITTLE hotter than I think I can get away with. I then import it, and find volume spikes that cause over-level peaks in the current render. I then lower those peaks manually, one at a time. I usually use the "amplify" plugin with a negative value like -1 or -3 dB. Once I have done this, I bounce a new mix and compare to make sure I got 'em all! If there are more than 10-100 peaks, I just master it a little lower, depending on how much work I want to put in for that last little bit of diminishing returns!!!
[/quote]

I'm not sure what the difference is, or if there even is one.  This is how I learned to do my edits and I like the results.  I wouldn't mind an easier way, but not at the expense of my recording.  What we do doesn't seem much different, I just prefer to minimize the redlines, although it happens and I lower them the same way you described at 1db at a time.


Noah:
"Peak Normalizing does not affect dynamic range.
“Eliminating peaks” = limiting. This is, by definition, a reduction of dynamic range."

I'm not debating "Peak Normalizing" because it isn't something I know enough about.  Perhaps you or someone can give me an explanation of this so I can better understand the differences as compared to what I do now.
However, I will argue that what I do by reducing a few peaks (as defined by me as spikes that are high enough, like a single overenthusiastic drum beat, to keep the balance of the recording down at a lower than desirable level, but did not necessarily peak or redline during live recording.) is not the same as using a limiter, which will essentially cap off the recording level causing an almost automatic compression by keeping all of the highest peaks at the same level, thereby eliminating much of the dynamic range.  The only time I use a limiter is in a stealth situation where I have to set and forget my levels.  I'll still set them low, in hopes of never utilizing the limiter's function, but know it's there just in case.  Noah, you have seen how I record...do you really think I peak out at a live show?
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (13)
  • Needs to get out more...
  • *****
  • Posts: 12542
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #100 on: February 19, 2018, 03:16:24 PM »
I will argue that what I do by reducing a few peaks (as defined by me as spikes that are high enough, like a single overenthusiastic drum beat, to keep the balance of the recording down at a lower than desirable level, but did not necessarily peak or redline during live recording.) is not the same as using a limiter

What you describe above is the same as limiting with the threshold set to only catch the most excessive peaks.  The only difference is that you are selecting and reducing each peak manually, rather than having the limiter algorithm find them (as determined by the threshold value) and reduce them (the reduction ratio value) for you automatically.

Quote
.. is not the same as using a limiter, which will essentially cap off the recording level causing an almost automatic compression by keeping all of the highest peaks at the same level, thereby eliminating much of the dynamic range.

That's describing a much lower threshold setting which captures not just a few excessive peaks, but is effecting most peaks.

Both are "limiting peak excursions" and reducing dynamic range, as is your manual reduction technique.  The difference is simply one of degree based on different settings of the limiter (or your manual process), but are otherwise fundamentally the same process.




"Normalization" is simply an automated way of raising the level of the entire file by a certain value, where the normalization routine helps determine what that value should be to get the result you want.  It is a linear process and does not affect dynamic range - it raises the level of everything by the same amount.  Normalization with regards to the polishing of music files for release means "peak normalization", where the highest peak value is of interest (to avoid it going over) even though everything is being increased by the same value across the board.   It is the same as manually finding the highest peak, figuring out how much you can increase the signal level of the entire file by to increase everything as much as you want, and applying that amplification.

Sometimes you may hear of "RMS normalization."  This is also linear.  The only difference is that the RMS value of the signal is what is monitored instead of the peak values.  This is useful for matching the average energy level of two files for comparison so that they both have the same loudness so you don't deceive yourself by preferring the otherwise consciously imperceptibly louder one (RMS correlates to loudness much better than peak values).  But RMS normalization is not what you'd want to use to raise the level of your files so that the peaks are just below full-scale, as you don't really know where the peaks land until you render it.  It can accidentally cause overs since the highest RMS value will always be lower than the highest peak value and the two are not strongly correlated.

To complicate things, some programs will combine RMS normalization with a secondary limiting function to avoid overs when the RMS value is pushed to high.  This is two separate functions being applied on one step.  The second limiting function limits the dynamics, the RMS normalization it self does not.

« Last Edit: February 19, 2018, 03:22:57 PM by Gutbucket »
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<<

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 570
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #101 on: February 19, 2018, 04:40:56 PM »
I will argue that what I do by reducing a few peaks (as defined by me as spikes that are high enough, like a single overenthusiastic drum beat, to keep the balance of the recording down at a lower than desirable level, but did not necessarily peak or redline during live recording.) is not the same as using a limiter

What you describe above is the same as limiting with the threshold set to only catch the most excessive peaks.  The only difference is that you are selecting and reducing each peak manually, rather than having the limiter algorithm find them (as determined by the threshold value) and reduce them (the reduction ratio value) for you automatically.

Quote
.. is not the same as using a limiter, which will essentially cap off the recording level causing an almost automatic compression by keeping all of the highest peaks at the same level, thereby eliminating much of the dynamic range.

That's describing a much lower threshold setting which captures not just a few excessive peaks, but is effecting most peaks.

Both are "limiting peak excursions" and reducing dynamic range, as is your manual reduction technique.  The difference is simply one of degree based on different settings of the limiter (or your manual process), but are otherwise fundamentally the same process.




"Normalization" is simply an automated way of raising the level of the entire file by a certain value, where the normalization routine helps determine what that value should be to get the result you want.  It is a linear process and does not affect dynamic range - it raises the level of everything by the same amount.  Normalization with regards to the polishing of music files for release means "peak normalization", where the highest peak value is of interest (to avoid it going over) even though everything is being increased by the same value across the board.   It is the same as manually finding the highest peak, figuring out how much you can increase the signal level of the entire file by to increase everything as much as you want, and applying that amplification.

Sometimes you may hear of "RMS normalization."  This is also linear.  The only difference is that the RMS value of the signal is what is monitored instead of the peak values.  This is useful for matching the average energy level of two files for comparison so that they both have the same loudness so you don't deceive yourself by preferring the otherwise consciously imperceptibly louder one (RMS correlates to loudness much better than peak values).  But RMS normalization is not what you'd want to use to raise the level of your files so that the peaks are just below full-scale, as you don't really know where the peaks land until you render it.  It can accidentally cause overs since the highest RMS value will always be lower than the highest peak value and the two are not strongly correlated.

To complicate things, some programs will combine RMS normalization with a secondary limiting function to avoid overs when the RMS value is pushed to high.  This is two separate functions being applied on one step.  The second limiting function limits the dynamics, the RMS normalization it self does not.

Thank you for the explanation.
I'm not sure if I was initially clear as to what I do to bring my levels up or down.  When doing so over the entire file, nothing is altered other than the "saturation".  More often than not, I'm raising the level of an entire file (set) by + "Xdb" (or lowering if I'm balancing out multiple tracks to blend a certain way...more live mics or more soundboard...).  When I refer to spikes, I'm thinking in terms of those random drum beats, a blast of feedback from an instrument (which I'll try to reduce even more), a thud on a microphone, etc. that are causing the entire set to be lower than I'd like.  Here comes the part that I may not have been clear on:  I don't uniformly drop those spikes to match the next highest levels, I reduce them by 1db - 3db depending on their severity, essentially maintaining their higher level, just not as a "run-away" spike.  Does that make more sense?  I know I'm thinking of what I do (I'm at work now), but may still be leaving out some fine details.  When I do this, I do it individually as to maintain their relation to the rest of the music.  I don't just say, "OK, I'll reduce these few spikes to X", but rather I may reduce one by 2db, and another by 1db, allowing me to bring the entire file up another couple db's.  So, if I used the normalization feature instead, wouldn't that all be done at a single level, say 1db for example?
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (13)
  • Needs to get out more...
  • *****
  • Posts: 12542
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #102 on: February 19, 2018, 06:51:43 PM »
Okay to be clear, you are doing three things-
1) identifying the offending peaks
2) reducing the level of those peaks by some amount
3) raising the level of the entire file into the "now unused top part of the scale" where you've made more room via lowering the highest peaks.

The first two steps are limiting.  The third is normalization, regardless of if you are doing all this manually or using a limiter and normalization algorithms to assist you.

Okay, now the details-
Quote
When doing so over the entire file, nothing is altered other than the "saturation".  More often than not, I'm raising the level of an entire file (set) by + "Xdb" (or lowering if I'm balancing out multiple tracks to blend a certain way...more live mics or more soundboard...). 
  ^This describes manual normalization.

Saturation is a magnetic tape term.  Good magnetic tape saturation meant finding an appropriate balance between low-level tape hiss noise at low saturation levels and non-linear compression effects at high saturation levels.  Digital audio is linear and does not suffer (or benefit depending on your perspective) from those distortions.  It's just digital level, measured in terms of dBfs (decibels, full scale) where 0 dBfs is the highest level possible without clipping distortion.  Normalization raises the level of everything equally to closer to 0dBFS.  It's good practice to leave a small buffer of a dB or so at the top, with the highest peaks topping out at a few dB below full-scale, say -2dBfs or whatever, since that can avoid something called intersample-overs, where the reconstructed analog waveform my occasionally clip even though the highest sample value to either side of that peak is actually below 0dBfs.

Quote
I don't uniformly drop those spikes to match the next highest levels, I reduce them by 1db - 3db depending on their severity, essentially maintaining their higher level, just not as a "run-away" spike.  Does that make more sense?  I know I'm thinking of what I do (I'm at work now), but may still be leaving out some fine details.  When I do this, I do it individually as to maintain their relation to the rest of the music.  I don't just say, "OK, I'll reduce these few spikes to X", but rather I may reduce one by 2db, and another by 1db, allowing me to bring the entire file up another couple db's.  So, if I used the normalization feature instead, wouldn't that all be done at a single level, say 1db for example?

^
Okay so this is the limiting steps (finding and lowering the peaks), not the normalization part (raising the level of everything) which is applied after the peaks are lowered.  The threshold control on the limiter is the detector which determines what will be affected.  Only the peaks which exceed the threshold setting might be affected (might depends on other settings of the compressor/limiter.  If limiting, might becomes probably or definitely (which is partly what differentiates limiting from compression).  The threshold is only a sensor.  It does not affect the sound itself.  It's just doing the identification of peaks part.  Part 1) in the list above.

The limiter's compression ratio and time-constraint settings (attack time, release time, and on a digital limiter- look ahead) affect how the peaks which are detected as exceeding the threshold are reduced and by what amount.  Limiting implies a high compression ratio and a very short attack time to more aggressively control excursions above the threshold.   A short attack time catches the shortest onset peaks, (look ahead essentially enables the ability to achieve an zero or immediate attack time, catching everything).

The ratio determines how much reduction in level is applied above the threshold.  Unless the ration is set to infinite, it won't crush all peaks to exactly the same peak level, unless its a "brickwall limiter" intended to absolutely never let anything exceed a certain peak value (which in essence is infinite compression with zero attack time above the threshold).  Instead it turns down the level by the ratio amount.  So a larger peak and a lower one, both of which exceed the threshold value, will still peak at different levels, but both will be reduced by a percentage of their original value.


Here's the thing- It can be hard to wrap your head around all these settings to determine exactly what the limiter (or compressor - they are the same thing essentially, only differing by degree in settings) is doing.  It can do what you describe, but when you go in and manually find each peak and manually reduce it, you know exactly which peaks are being targeted and by how much they are each being reduced.  The trade off is between the effort to do that manually verses developing the skill of how to set the limiter to do the same thing.  Those who really know their way around compression and limiting and do it well (not the loudness war casualties) will often use several passes instead of just one pass of compression and or limiting, catching only the very highest peaks with a more aggressive setting while perhaps using a less aggressive setting with a slightly lower threshold on a seperate pass to sort of "feather" the reduction and make it more transparent.

None of this is to argue that you should do it this way.  Do what works for you and you are comfortable with.  It is only to explain that limiting and normalization can be done manually or with limiting and normalization tools, but is essentially the same process.  Like musical performance and like taping itself, the quality of the end results depends more on the operator and how one uses the tools available to them rather than the tools themselves.
« Last Edit: February 19, 2018, 07:46:18 PM by Gutbucket »
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<<

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 570
  • Gender: Male
Re: Audacity (osx): How to mix two AUD sources?
« Reply #103 on: February 19, 2018, 07:15:55 PM »
^^ OK, that was helpful information.  Thank you for taking the time to explain it to me   :)
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline morst

  • Archivist: Camper Van Beethoven & Cracker
  • Trade Count: (1)
  • Needs to get out more...
  • *****
  • Posts: 2760
  • Get in touch if you wanna record Cracker or CVB!
    • Soundscape Preservation Society
Re: Audacity (osx): How to mix two AUD sources?
« Reply #104 on: February 19, 2018, 07:34:12 PM »
^^ OK, that was helpful information.  Thank you for taking the time to explain it to ALL OF US   :)

Fixed that for us!  :D
Teams: Neumann, Bay Area Tapers, Multitrack, Pioneertown Tapers, Mac Geeks, Cassette Masters, Poster Collectors, Alumni of teams St Louis, Upper Midwest & Milwaukee / Southern Wisco

 

RSS | Mobile
Page created in 0.221 seconds with 43 queries.
© 2002-2018 Taperssection.com
Powered by SMF