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A Good Structure Question - Audiobook Narration


CJ8073

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Hi, Everyone, I have two quick questions:

 

1) Is a track that is placed in a folder and hid (entire folder and content) from view in both the track/console safe from inadvertent edits?

2) I am doing some volunteer audiobook narration; however, I would like to pursue this as a career at some point and I am trying to get things well organized inside of Cakewalk.  Right now I have each section/chapter separated on individual tracks and placed in their corresponding folders.  I have no plugins on any of those tracks.  I set up a "Voice" stereo bus that I run each section/chapter through after I'm done editing that has some plugins on it.  Very minimal - small, small EQ and a light de-esser.  I also have set up a separate stereo bus for each track (named for each track as track name_bus) that also has some plugins on it - an expander and limiter.  When I'm done for a track I export each one individually - so, like this - Track output ------> Voice bus ouput -------> Bus for associated track output (my headphones).  (I do individual buses for each track because I find that sometimes my volume levels are ever-so-slightly different when I record a track for a section and I have to mess with the limiter to get the same volume levels for each track - I don't know if I am doing this right, but it has seemed to work to keep everything exporting at the same volume level and I don't want to have to write down and change the limiter every single time back and forth for each track.... am I making sense? I'm sorry if I am wording anything wrong).  So, in the end, I only have switch out the outputs for whichever track I want to export.  However, I recently came across something called a patch point and I am wondering if I should be sending all of my tracks to a patch point with the "Voice" plugins there instead of using the "Voice" bus that I set up?  Is there a benefit to doing it that way and am I using the Stereo bus for "Voice" wrong?  Lastly, overall, does this seem like a good setup (separating each section/chapter on a different track in folders)?

Thank you for any input and I'm sorry if I sound silly at all - I am still learning.

Sincerely,

Cori

Edited by CJ8073
Clarifying a bit more.
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  • CJ8073 changed the title to A Good Structure Question - Audiobook Narration

To make my tutorials I first record all the narration so as to not babble. I suck at live recording of narration so I write a script and I read it from my second monitor.   I would think my workflow would be exactly how I would do what you are doing. The only difference I can see is your tracks would be much longer. Mine are never more than 30 seconds.  

I have a template with 30 Audio tracks. It is easy to add or subtract these. 

I don't bother,  but you can name each track. 

After I record a track I listen and fix and edit mistakes. I then turn off record and Mute the track. 
I move to the next track and set in record. And so on. 

I'm very careful to stay same distance from my mike and I always record at a safe level.  

After I'm done I might top and tail the tracks with fade ins and fade outs to remove unwanted noises like breathing. 

The master bus has a few compressors, limiters etc. to make sure the voice stays at the same level with a target peak level of -1.0 db. 

That's it, no folders nothing complicated at all. 

 

Then I export as "Tracks through entire mix" at 48/24. This creates a stem ( track) for each of the " Chapters" 

 

Edited by JohnnyV
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3 hours ago, Byron Dickens said:

Cakewalk is massively overkill for what you're doing but check out these tutorials:

 

 

Thank you for the recommendation.  I am currently subscribed to this channel.  I know cakewalk is massively overkill, but I tried both audacity and Cubase and I did not like either.  Cakewalk feels like home and it was easiest for me to get comfortable with.

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2 hours ago, JohnnyV said:

To make my tutorials I first record all the narration so as to not babble. I suck at live recording of narration so I write a script and I read it from my second monitor.   I would think my workflow would be exactly how I would do what you are doing. The only difference I can see is your tracks would be much longer. Mine are never more than 30 seconds.  

I have a template with 30 Audio tracks. It is easy to add or subtract these. 

I don't bother,  but you can name each track. 

After I record a track I listen and fix and edit mistakes. I then turn off record and Mute the track. 
I move to the next track and set in record. And so on. 

I'm very careful to stay same distance from my mike and I always record at a safe level.  

After I'm done I might top and tail the tracks with fade ins and fade outs to remove unwanted noises like breathing. 

The master bus has a few compressors, limiters etc. to make sure the voice stays at the same level with a target peak level of -1.0 db. 

That's it, no folders nothing complicated at all. 

 

Then I export as "Tracks through entire mix" at 48/24. This creates a stem ( track) for each of the " Chapters" 

 

Thank you, for your very helpful reply.  I really appreciate it.  And thank you for the videos, they are so helpful!  So, if I don't use folders and I use only tracks, can I hide the finished tracks so that I don't accidently edit them while I'm working on an active track?  When exporting tracks through entire mix, can you export all at once?  I've been doing one track at a time.

Again, thank you, so much for your reply!

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1 hour ago, Byron Dickens said:

There's definitely a lot to be said for being comfortable with the tools you're working with.

Yes!  I tried Audacity, but it was awful and I felt like every change was so permanent.  (I learned a little about destructive vs non-destructive editing after that)  I tried Cubase and it was so confusing to me.  I found Cakewalk and since it was free I figured I'd give it a shot and it's been great so far!!  Honestly, I wish I knew more about music because I feel like the music that can be put together with a DAW is pretty amazing.

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Yes you could definitely just hide finished tracks. It would be a choice between that and the folders, which ever seems easier. 

And yes, the export Tracks Trough entire mix feature  will do all tracks.  It shows a confirmation list before you hit ok so you can make sure the correct tracks are selected. 

The Export dialogue is very powerful and there’s a lot of options like in how the track are named and if you want automation and track or bus effects added.  I use the entire mix  selection because it includes the master bus. 

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16 hours ago, JohnnyV said:

Yes you could definitely just hide finished tracks. It would be a choice between that and the folders, which ever seems easier. 

And yes, the export Tracks Trough entire mix feature  will do all tracks.  It shows a confirmation list before you hit ok so you can make sure the correct tracks are selected. 

The Export dialogue is very powerful and there’s a lot of options like in how the track are named and if you want automation and track or bus effects added.  I use the entire mix  selection because it includes the master bus. 

Thanks for your help.  Am I over complicating things by using multiple buses, do you think?  If so, how else should I deal with the volume issue when my Chapters might be slightly different between chapters?  Eventually, I'd like to be able to narrate audiobooks for ACX and I know they have a specific dB range that final submissions have to be.  In order for me to get each chapter close to each other in the end and within spec, I've had to tweak the limiter ever so slightly differently for each one.  If I put all the plugins on a single master bus and run the tracks through the entire mix, I'm afraid the final volume levels will sound a little different between chapters.  Is there anything else I could be doing better?  Thanks again for any help!

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I was going to say to lock the clips. That is a great way to protect them some. I like folders better than hiding clips. That way all of the other tracks are only one click away. There are several ways to level the different take.  First, you should use the volume to adjust them as close as you can. Some people use compressors and limiters to help level audio. These are good, but my favorite plugins are an all in one solution. There are many on the market. Some of my favorite are Waves MV2 and Vocal Rider (these two can be used together) or WA Productions Outlaw. Be sure to check out the deals forum before you buy any plugins! Outlaw will let you set a LUFS range which might be a plus later on.

There are some good tools built into the cakewalk pro channel as well.

Others -

Waves - playlist rider

Plugin alliance - Master Desk, Master Desk pro

Sonible - Smart Limit

Hornet Audio - several inexpensive gain rider plugins

 

You should probably look at Outlaw 1st!

https://www.waproduction.com/plugins/view/outlaw
 

Good luck with your project. There is plenty of good help here. 

 

 

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A few quick things with narration and long tracks in addition to the above:

  1. Keep your setup and narration as consistent as possible so that variations between sessions are as minimal as possible. Doing a chapter per track is a good way to keep things organized, and you can copy FX chains between them (or run them through a buss), so you can tailor FX as you work and only apply (render) them in a final export when they are set and consistent.
  2. Be sure to always select things properly before editing. This one is important to any workflow, since edits take to what you have selected.
  3. When working with long narration tracks, using something that will insert a visual transient into the wav file (similar to a clap) will make edit locations easy to visually find when editing. I typically bounce each track into a single wav file after level across it is consistent. Pick a standard to work to for all tracks, then the FX chain will also be consistent for them all with minimal tweaking required for final exports.
  4. Another general item is that Cakewalk does not delete audio, so inadvertent destructive edits will still leave the original / prior version in your audio folder. Use track names to define chapters (best habit is to do immediately when creating a new track), then the audio files are also easier to find in case you want to use another wav editor program. This is also why I bounce tracks, so that I will end up with only one wav file per chapter with a proper name. To clean up these audio folder afterwards (they are prone to filling with multiple wavs during editing), it is easiest to do a "file->save as..." to a new folder and select "copy all audio with project." What this does is copies only the audio in use by the project to the new folder, which can help with cleanup when you finish. The original folder will still be there, so can either archive or remove that project folder to your preference.
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Narrative for audio books is not like music and the voice is all that is heard. There’s no music to mask imperfections. Its definitely a talent not to be underestimated. 
My son in laws Dad has been going into a studio in Scottsdale AZ for years and doing this. 
He said they only let him record for an hour because your voice gets tired quickly.   
 

The big secret to doing this is a treated room, the right mike and mike technique. You certainly don’t want noticeable compression and limiting on the tracks. I do have the Loud max and the SL63x on the master but my goal is they are not working, they are just watchdogs. 
Cakewalks Boost 11 is often overlooked but I find if you use it lightly is a very good expander limiter. I set my peak at -4 db and if a track needs a little boost I use no more that 2 or 3 db. But that’s the lazy man’s method. I usually manually edit but I mentioned it because manual editing of a 1 hour track would be very time consuming and I’m not sure Audio Book pays good enough for that.


I have my set up with a SM7 and I point it at a 45 degree angle towards my left of my face  so I’m talking across its windscreen. I keep my right cheek almost touching the wind screen. This needs to never change in a session. I find this totally eliminates Plosives. 
After a lot of trial and error I have an exact setting on my interface for a level that never peaks.  Note that this a bit hotter than the setting I would use for singing. 

The tracks should never peak past -4 db. If they  do I will find the error and I apply clip gain to the peak. 

I don’t,  but you can also use Process Normalize to set all tracks the same peak levels. But peaks don’t tell you about actual RMS or LUFS level. 
For that I use the paid version of You Lean Loudness meter. You can drag and drop the track into its GUI and it analyzes the level and LUFS. I would highly recommend it for this reason alone. LUFS are way more important than peaks as far as perceived levels go. 
I put it in the Master bus bin as part of the template. I can open the GUI from there and drag any track to quickly analyze. 

This is a screen shot of my Template. Notice the setting of the Loud Max Limiter which is what I use to achieve my goal of -1db for You Tube videos. 

You could use the  Boost 11 or the Loud Max on a track. I put it in the screenshot to demonstrate how I would use it but it is not part of the template. The SL63x is just there for the odd time I feel a need for a little compression.  The Boost 11 and Loud Max are interchangeable. I tested about a dozen free limiters and those 2 where the keepers. The Loud max is the winner for actually keeping within 0.03 db of it's  settings. The You Lean is also the most accurate analyzer I tested meter for peak readings.  

Screenshot(1312).thumb.png.48649c01ded04cf540ae13ce05d44cd6.png

 

Edited by JohnnyV
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I want to say thank you! so much to everyone for your very helpful and insightful replies!!  I'm learning so much!  I wanted to upload a screenshot of how I've got everything set up so you could see it all.  Does it makes sense the way I've got it set up? 

MyCakewalk.thumb.PNG.b3e69a4e575b17ec972431649e05b5a1.PNG

 

I might consider not using folders... I'm not sure, yet.  I was also thinking about this:

On 8/26/2023 at 5:06 AM, Max Arwood said:

There are several ways to level the different take.  First, you should use the volume to adjust them as close as you can.

So, as you can see the way I've got things; I don't use any plugins on the individual tracks, I run them one at a time through the voice track (when I am exporting them) and then I set the voice bus to output through a corresponding bus for each track where I can tweak the volume at varying amounts a little here and there per track so the end result is that all of the tracks have roughly the same volume.  However, based on the above suggestion from Max, it seems I am doing this a little backwards?  Should I be using the volume control per track to even them out first?  Then use only a single, final bus for the voice bus to run through in the end?  One that I could just set and not have to touch?  Or should I just be using a single final bus altogether and put all of the plugins on the one bus?  I don't change anything on that Voice bus at all.  (The reason why I don't put the plugins on any of the tracks individually is because that's too many plugins for me to have to keep track of, plus it seems a little redundant since I really wouldn't be changing the settings of any of them, and I I like to keep the actual recording completely untouched as far as initial sound is concerned)

When I export, I've been doing each track individually using the following output path:  Track to voice bus to corresponding bus.  But I feel like, if I can somehow only have to have a single final bus (or combination of the two Voice to Master buses, and get rid of all of the individual buses) then this would make the export "tracks through entire mix" make more sense?  I think?

That's where I was thinking (after learning a little bit about patch points) that maybe I was supposed to be using a patch point for the voice plugins, instead? 

So, again, to come back around, ultimately, in order to not have to use so many buses for each of the tracks, am I suppose to level out the volume using the individual volume on each of the tracks, first?  This way there is only one master bus at the end to finish getting all of the sounds up to the final volume level required at the end? (I use the Boost11 to up the volume and limit to -3.5dB)

 

As far as editing:

On 8/26/2023 at 3:39 AM, Bristol_Jonesey said:

You can prevent unwanted editing on a clip by right clicking it and select Clip Lock. You can lock the data, position or both 

  Okay, I will keep this in mind, then.  I'm not sure if I'm going to keep using the folders or just use tracks, but I will definitely consider the idea of Clip Lock on my final tracks so I don't accidentally edit them.  Thank you!

======================================

On 8/26/2023 at 10:18 AM, JohnnyV said:

Narrative for audio books is not like music and the voice is all that is heard. There’s no music to mask imperfections. Its definitely a talent not to be underestimated. 
My son in laws Dad has been going into a studio in Scottsdale AZ for years and doing this. 
He said they only let him record for an hour because your voice gets tired quickly.  

Congratulations! to him!  I think narrating audiobooks is fun so far.  And even the editing has been such an interesting learning experience.  I haven't done anything that pays, yet, I've only done Volunteer work, but I'd really like to turn it into something more, which is why I am trying very hard to get decent at this before diving in for a paying project.  I want to make sure I have something to offer.  If my narration isn't well received, I'd like to maybe edit audiobooks instead because, honestly, I just really love working in the DAW and editing the audio.

My environment and setup:

So, for the room I record in - lol, it's a tiny closet.  It has a popcorn ceiling and carpet.  I had an old eggshell twin mattress topper I cut into rectangles and attached to my wall in front of me, behind me and on the door of the closet next to me.  I layered every wall and the door way with lightweight, different density throw blankets and then layered that with thick drapes that reach from the ceiling to almost the floor around every wall and the door.  I layered an extra comforter on top of the carpet and a few more throws around the baseboards to block extra noise from my computer (it sits about two feet outside the door of my closet).  So, it obviously won't block the neighbors lawn mower when he's mowing outside or an extremely loud car driving by, but acoustically, I think it does well.  (My husband kindly helped me tacking all of this up)

The equipment I use is an AT2020 XLR microphone and the Steinburg UR12 audio interface with input gain set to the 8th little dot).  I have my little tv stand/tray covered with fabric and my microphone is attached to that with a pop filter in front of it.  I have it angled slightly towards my left cheek and slightly downward and I sit about 7/8 inches from it.  (I noticed that I have a whistle "s" a bit, so, I read that if I make it slightly off axis and angled downward, it would help with that) To help me keep a relatively close volume level each time, I've been resting my arm on my chair and kind of getting into that position every time I step away and sit back down, whether it's needing to get up for a moment or until the next day.  However, I still sometimes still struggle to get volume levels matched, so that's why I had implemented all of the individual buses, to balance out and raise volume levels using the Boost11 plugin.  But I kept thinking I was doing something wrong by doing that.  

I still get a little confused with the whole RMS vs LUFS thing and I've watched several videos.  I finally found this plugin:  dpMeter5, which I have been using on the final output bus to get a feel for the RMS value of a track before I export it.  (I got tired of exporting a gazillion times and running the same file through a file tester to see if I was meeting the RMS requirement for ACX)  That seems to work to get decent feel; however, I am thinking, if I a supposed to be leveling volume on individual tracks first, maybe I should be putting that little guy on each track to check RMS there first?  Does that sound right?

Some of the plugins that have been mentioned, do they cost?  I am trying to stick with free plugins at the moment because I really can't afford to invest in those just yet, so if any of you know of any other great free plugins that would be helpful, I appreciate the suggestions!  (I was looking for a free mouth "de-click" plugin, but so far I can only find paid ones.  Right now, I'm editing out any weird clicky noises manually ..... literally takes forever!  haha, fun, but forever!  (This probably has to do with better recording techniques, also, so I am trying to get better at that, too)  So, if anyone has a suggestion for that, in particular, I'd love to know about it!) 

Thanks! to all who have made it this far, I appreciate it.  Honestly, I've been trying to tackle learning this on my own and you guys/gals here on the forum have been such a great help to me whenever I've had a question.  I am very grateful to those of you who help and contribute to these forums because I have learned so much, even from reading others' posts!  And to all who make Youtube videos! many thanks!

~Cori 

 

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53 minutes ago, CJ8073 said:

So, as you can see the way I've got things; I don't use any plugins on the individual tracks, I run them one at a time through the voice track (when I am exporting them) and then I set the voice bus to output through a corresponding bus for each track where I can tweak the volume at varying amounts a little here and there per track so the end result is that all of the tracks have roughly the same volume.  However, based on the above suggestion from Max, it seems I am doing this a little backwards?  Should I be using the volume control per track to even them out first? 

The primary reason for normalizing audio to some standard is when using a universal FX chain (from a buss) you want the first compressor in that chain to see relatively the same signal (so the knee is consistent for each track). Many FX have thresholds to them, so that first compressor makes the rest of the FX chain yield a consistent output. You can (and often do) use another compressor/limiter/maximizer on the final result, but there is an audible difference between tracks that were processed properly and those that were not (i.e., if that first compressor did nothing because the knee wasn't triggered and affected the rest of the chain) and just had the volume raised at the end.

There is no right or wrong for what you are doing, and using a buss has the benefit that you can adjust only one processing chain rather than copying adjustments between tracks. That can be burdensome when the material you are working with is essentially the same anyway and you decide late in the game that you want to change settings. Max's advice on consistent levels is what makes that possible. With a consistent recording setup, you should be able to get away with normalizing a bounced track (once your edits are completed); I habitually will normalize to -3dB since I already set up compressors expecting that signal level, which makes tweaking them simpler.

Another trick you may not be aware of is Patch Points/Aux Tracks, which let you record the output of your chains to a new track (similar to exporting, but recorded to a new track). It can be quicker than exporting and you can drag/drop those tracks out of Cakewalk  into a Windows folder. Depending on your workflow, this may or may not be beneficial, but something to keep in mind should you need it.

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16 hours ago, CJ8073 said:

I finally found this plugin:  dpMeter5, which I have been using on the final output bus to get a feel for the RMS value of a track before I export it. 

This is why I recommend the paid version of the You Lean meter. It is the only meter that allows instant results by drag and drop of audio (and video)  files to the GUI. There is a free version that works in real time as well but the upgrade is only $50 Can. If your time is worth anything to you this is a very good investment.  https://youlean.co/youlean-loudness-meter/

 

I think I already explained clearly my workflow so I won't elaborate. Key point is that if each track has been optimized and adjusted then there would be no need for buses or a separate export per track. It would all be done in way less time. Good workflows are ones that might take more prep work but save time in the long run. That prep work includes a proper recoding environment and set up and the right mike and equipment. If you tried a SM7B you might not need to do any editing. I used to own an AT 2020. It does not work at all for me for the reasons you just mentioned. Noisy. 

Edited by JohnnyV
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On 8/28/2023 at 7:25 PM, mettelus said:

The primary reason for normalizing audio to some standard is when using a universal FX chain (from a buss) you want the first compressor in that chain to see relatively the same signal (so the knee is consistent for each track). Many FX have thresholds to them, so that first compressor makes the rest of the FX chain yield a consistent output. You can (and often do) use another compressor/limiter/maximizer on the final result, but there is an audible difference between tracks that were processed properly and those that were not (i.e., if that first compressor did nothing because the knee wasn't triggered and affected the rest of the chain) and just had the volume raised at the end.

There is no right or wrong for what you are doing, and using a buss has the benefit that you can adjust only one processing chain rather than copying adjustments between tracks. That can be burdensome when the material you are working with is essentially the same anyway and you decide late in the game that you want to change settings. Max's advice on consistent levels is what makes that possible. With a consistent recording setup, you should be able to get away with normalizing a bounced track (once your edits are completed); I habitually will normalize to -3dB since I already set up compressors expecting that signal level, which makes tweaking them simpler.

Another trick you may not be aware of is Patch Points/Aux Tracks, which let you record the output of your chains to a new track (similar to exporting, but recorded to a new track). It can be quicker than exporting and you can drag/drop those tracks out of Cakewalk  into a Windows folder. Depending on your workflow, this may or may not be beneficial, but something to keep in mind should you need it.

 I would, honestly, really like really understanding everything you are explaining to me here.  I mean this sincerely.  I can, after almost a year of reading and researching understand a basic level of what you are saying, but there is still so much I'd like to learn.  (I found the used book, Modern Recording Techniques, 7th Edition by David Miles Huber, that I have been reading.  I am currently on Chapter 4, Microphones.  I've loved learning from it so far!)  So, when you say, "knee" are you referring to the strength of the compression?  Or, the strength of the compression after the compression starts?  I have listened to quite a few youtube videos about this and at first I thought ratio and knee were one in the same, but after doing more research and video watching, I think they are two different things.  Ratio being the overall compression strength with knee being complementary to that in that it is how the compression strength takes affect once it starts compressing.  Is that right?  I don't really understand what this does sound-wise, but it's an interesting concept and I'd love to actually hear examples of what the knee does exactly.

Normalizing is when the volume of the overall track is raised/lowered without changing the dynamics of the waveform, correct?  So, normalizing to -3dB just means taking the whole waveform and raising the output volume of that, right?  However, gain is attenuating the highest part of a waveform at a certain threshold, thus changing the dynamic range of the sound?  So, do you normalize first?  Then compress afterwards?  At which point can you EQ?  and is EQ essentially compressing, but specific frequencies?  I ask that because I do EQ out some boxy/muddy sound out of my recordings since I record in a small space.  Not a lot but a little and isn't that doing the same thing as compressing but only for certain frequencies?

Yes, the patch points is what I was originally trying to understand!  Because I had seen somewhere where someone was saying that you could run your tracks through a patch point instead of using a bus and I was thinking maybe that's what I should have set up for the "Voice" part of my setup instead of putting that on another bus.  (Is it bus or buss? lol, I keep seeing it spelled differently)  I wasn't sure what the benefits were for that!  So, based on what you wrote - If I flow my tracks through a Voice patch point one at a time, can I see the new waveform that is created after all of the effects have taken place?

Thanks, again, for your help!

On 8/29/2023 at 10:58 AM, JohnnyV said:

This is why I recommend the paid version of the You Lean meter. It is the only meter that allows instant results by drag and drop of audio (and video)  files to the GUI. There is a free version that works in real time as well but the upgrade is only $50 Can. If your time is worth anything to you this is a very good investment.  https://youlean.co/youlean-loudness-meter/

 

I think I already explained clearly my workflow so I won't elaborate. Key point is that if each track has been optimized and adjusted then there would be no need for buses or a separate export per track. It would all be done in way less time. Good workflows are ones that might take more prep work but save time in the long run. That prep work includes a proper recoding environment and set up and the right mike and equipment. If you tried a SM7B you might not need to do any editing. I used to own an AT 2020. It does not work at all for me for the reasons you just mentioned. Noisy. 

Thank you, I will keep the youlean loudness meter in mind.  I do have the free version.  I will also keep the workflow that you presented in mind.  I have been reading and re-reading this entire discussion post to learn from.  I have not tried the SM7B microphone.  Honestly, I don't think I am at the point to being able to try it because when I looked it up, it is quite above my budget.  For now, I will have to find a way to make the AT2020 work.  Thanks, though, and I will keep it in mind later when I have more money that I can spend on equipment.

 

=========================================

For those of you, who have been kind enough to help, would you mind if I posted a 45 second sample of the piece I have been recording?  I have two samples; one is raw and the other processed.  I am just wanting a couple opinions on quality and to know if my final piece sounds okay or if I should change something to make it better.  I'm not worried about performance because that is something I am also still working on and right now my concern is more quality of sound, first.  I'll post it if you guys don't mind.  Thanks!

=========================================

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Is all good, we all have to learn at some point, so asking (and more often doing) are the best ways to learn new things. Let me summarize again for context with what you asked above.

The "knee" is where a compressor kicks in. Below it, the ratio is 1:1, and above that is where the compressor kicks in and uses whatever ratio you have set. It is simple the point where the compressor kicks in and looks like a knee on that graph.

You are correct with normalization. It takes the loudest part of a wave form and raises it to the level you specify while keeping the dynamic range. I do most EQ prior to compression (you can have multiple EQs in a chain) for the simple reason that if you reduce the dynamic range (compress) on a signal, it makes EQ more difficult, and if you have something you didn't want in the first place (like room noise), it is harder to remove after going through a compressor. The normalization is so that the compressor kicks in consistently for everything you are feeding it (without it, the entire signal may pass below the threshold you have set, so the compressor is doing nothing). The reason is more so that FX chains you set up will be easier to tailor when already "close" in the future.

Again, there is no right or wrong to many things, so is more consistency in a workflow that works for you and makes sense. Patch points are just a feature to keep in mind, but they allow you to record a signal at a processing point that you choose.

 

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