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To improve a song, I did another mixdown and master. The result was a song that was too loud and in the red consistently. Has that clipping become a permanent part of the audio? Should I start over from the protected original or try to fix with remaster/remix?


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I would go back to the source and redo. That being said, there are audio editors that can help remove the clipping, but it's better to go back to the source.


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You won't be able to edit wav files saved in the clipped format. You'll need to go back to whatever level you were before boosting the signals. We would need more info on what software was in the chain as you went from your final mix to mastering, etc.




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Thank you for that. I had earlier tried a remaster. The results were not good. This are BiaB WAV renders mixed down in Reaper and rendered to two tracks. So far, so good. I transfer them then in real time to a Tascam digital recorder which has mixdown and master modes. Usually, it works out all right; but, this time, the level got out of control. (i.e. I lost it.)
Thanks for the advice. I'll go back to the original files and transfer them anew.

Last edited by edshaw; 10/10/19 01:10 PM.

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edshaw,

One way to think of clipping is to imagine each wave file sample as a bucket. When there is clipping each clipped bucket fills to overflowing and spillage occurs. What's spilled is lost and can not be recovered.

Restoration software looks at adjoining samples and calculates from the good samples what the lost information might be but it is just an approximation.


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Yes, do it over. It’s time wasted to attempt to work with a file with clipping.


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Thank you for that. Interesting illustration, Jim. I had intuited that the file had been maimed. Yes, Matt, as you know, I've spent energy on building a standardized file system. Times like this, glad I did. (Not that I'm not capable of creating the random birds nest here and again.)


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Whenever I render a BIAB song I do so to individual tracks. Then when I pull the tracks into Reaper (or Cakewalk) I normalise all tracks to about -12 dB. That gives me a starting point. From there I get each track to how I want it then put the entire mix through a limiter set with a ceiling of -1.3 dB (I usually put mine to MP3). I use the limiter to get me to about -14 or so LUFS. This avoids any clipping and delivers a consistent level.

There are many good Youtube videos on Gain Staging. These are worth a look.

My thoughts
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Last edited by Teunis; 10/10/19 01:55 PM.

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Thanks, Tony. I import my BB WAVs into Reaper, too. So, your remarks give me something to go on. If I follow your directions, it could mean fewer problems in the Tascam mastering process, which involves working with eight little bars on a tiny screen.


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I should mention I set up the action Xenakios/SWS Normalise selected takes to dB value... I set this to Shift N. So all I do is pull in all the tracks - make sure they are all selected - press <shift N> and a box appears where I enter a value eg -12. Very simple, very quick.

Just make sure you have the SWS extensions installed in Reaper. The SWS extensions give Reaper a lot more capabilities.

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Your DAW doesn't indicate clipping on each track and the stereo buss?

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Bud,
Sounds like his DAW is a hardware device.
Quote:
.. are BiaB WAV renders mixed down in Reaper and rendered to two tracks. So far, so good. I transfer them then in real time to a Tascam digital recorder which has mixdown and master modes. Usually, it works out all right; but, this time ..


Your question is still valid, but just pointing out what he is left to work with in the hardware unit may not be the familiar tools/features/flow we are used to having in a DAW.


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Thanks. I need to read a little more carefully! smile

Having said that I thought Reaper was a DAW?? Why not throw a limiter on the mix and output it via one of nowadays industry standards, e.g., CD, streaming, the LUFS thing, etc.

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Originally Posted By: edshaw
. . . If I follow your directions, it could mean fewer problems in the Tascam mastering process, which involves working with eight little bars on a tiny screen.


Which Tascam Digital recorder do you use?


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I use the 8 track DP-03 SD, Charlie. The main thing keeping me from upgrading to the DP-24 is the size. I use it as my combination mixer/bg track player when I go out into the field. Everything is right there. The more I go along, the more I find those little bars are all you need -- fat but not peaking, tracks easily adjusted in the final mixdown.
It has bounce, mixdown, and master; cut, copy, and paste. Mixdown makes bounce virtually redundant, but still is handy to have around. Take the vocals, for example. They can be mixed and mastered, then imported back into the project. I have basically three elements: voice, guitar, BB track. Once the vocals have been mastered, it is hard to go back into the track, so you have to keep a protected original, just in case.
Lately, I've been using the ol' movie editor's trick, A and B rolls, especially useful for obtaining the same results as a punch in. A C roll can be used as a scratch track, same idea as a cue track.
Is this information helpful to understanding my situation? Thank you and the others for taking the time to help me with this. I know most here rarely use the old style recorder.

Bud: ("Your DAW doesn't indicate clipping on each track and the stereo buss?")
Yes, I should have caught it, but probably thought I could correct it later by turning down the volume. Not.

Last edited by edshaw; 10/12/19 02:52 AM.

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The 8 track DP-03SD is a very good unit and competes well with the DP24/32 in my opinion and unless you do live recording or have a dedicated home studio, it beats the socks off the Model 16/24. I am one that truly enjoys stand alone digital recorders and Tascam seems to be my favorite brand. I don't currently have a DP-03 but do have a DP-004 and DP-24. I had several cassette Portastudios and also the Neo2488MKII. Never had a crash with any of them.

I agree with the others to restart from scratch rather than attempt a repair. They've given you good advice and the only thing I'd advise you regarding mixdown and mastering on the hardware is the reminder that during these actions, the channel meters display the recorded audio on the track and not the fader position as it does during track recording. The OL lights indicate input and not track levels so there may not be clipping to any of the individual recorded tracks. The effects you use during mixing and mastering effects all add gain to the mix. EQ, compression, reverb are all cumulative gain. Leave plenty of headroom on your tracks before you do your mix.


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Thanks for that, Charlie, and thanks to everyone who chipped in to help. I did a song today, after rebooting to the BB WAV render. The process flowed way smoother and the results were much improved. While not many of us still use the digital recording hardware, the fundamentals are consistent and each comment added to my understanding. With experience, I'll get that headroom doped. That's really when the trouble started -- the compression phase. You know, the bars temporarily disappear throughout, ironically, when you need them most. The ears remain open, though.
Tuenis, I'm into that YouTube instruction area. Thank you, had no idea such an intense community existed!


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Yep start over from the original tracks...or the unadulterated master. Assuming it is not clipped from compression.

You can't unbake a cake. So don't use the clipped file and try to fix it.

Clipping comes in my mixes from running the compression too high or pushing the mix with my Boost FX. I call them "overs" where the signal exceeded zero dB... hitting the top of the digital limits. In extreme cases, you end up with bricked files. In other cases, the overs can cause nasty sounding digital distortion. The way I mix, I have to go back to my original project and turn my compression down. It's amazing what a few dB reduction in compression can do to open the mix and let it breathe.


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Thanks for that, Herb. With the help of this forum and a lot of patience, I've been able to see some progress in the final results.
EQ and COMP still have grey areas for me. As a hands-on learner, the only real course is to keep on keeping on. All information from they who been there is vital and appreciated.

Last edited by edshaw; 11/04/19 03:37 AM.

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Originally Posted By: edshaw
Thanks for that, Herb. With the help of this forum and a lot of patience, I've been able to see some progress in the final results.
EQ and COMP still have grey areas for me. As a hands-on learner, the only real course is to keep on keeping on. All information from they who been there is vital and appreciated.


compression and loudness boosters will put you over the line quicker than anything else. The important thing is to keep the dynamics in the music. I always use a wave editor so I can look at the finished wave. If it looks like a flat top wave, where everything is extremely level, I will go back to the original project in Sonar and look for my mastering plugs and see which compressor I have turned up a wee tad too much. I like to see hills and valleys in the wave. That lets me know I don't have the dynamics squished out of the music. I really only like to see my peaks touching 0dB in one or two places and nothing that is an "over".

Any time I have a mix where I have to put a booster (like Boost11) into my vocal signal path, I know one truth. I have everything else too loud. So it's time to turn down the band rather than trying to bring the vox up. Trying to boost the vox will result in overs and a brick looking wave on export.

A quick, handy, DAW trick, is... rather than going back through all the instrument tracks and lowering them....especially if they have volume envelopes... as that can be a tedious task.... I will simply pop in another BUSS and call it BAND.... and send all the instruments to it. All the volume changes are kept intact, and now, I can lower the overall band volume while leaving the vox at a good level.

Last edited by Guitarhacker; 11/05/19 05:44 AM.

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