Topic: Need Help w/ Workflow

[cabbie11]

Hello Everybody

I've been away from the taping scene for some time. Due to recent health issues I find myself with time on my hands and decided to archive my live recordings to my Mac.
I got Amadeus Pro, but with my health I'm finding it difficult to search out and assimilate all this info. This is not a sob story, but a request for a helping hand with my workflow.
If anyone could provide a step by step basic workflow (doesn't have to be Amadeus Pro, just basic steps) I'd really really appreciate it.

Also, tips for cleaning up tape hiss, loud clapping, room echo, etc. would be equally appreciated!

Thanks for all your help

[morst]

that's a very good question and really requires a huge answer, which I'm not prepared to exude in its entirety at this time.

First, I hope you plan to continue to preserve your masters, in case you later find that you could have done better, and just for general safety backups.

So, speaking of backups, I will suggest that you plan to make redundant copies. A minimum of two copies is essential to sleep at night. When (not really if) a hard drive fails, you will be very happy you had a third copy so you don't have to do the "one-drive panic" if you go from two down to one. Besides, drives are cheap, and you probably won't fill the first one the first month.

other than that, my first-day suggestions would include making a file structure where you plan to keep the files, and give your folders recognizable and consistent names. You can break things into years or alphabetical by bands, or just by dates, but it's a good idea to figure out your hierarchy before you get too deep into it. I prefer dates to be like 2012-03-20 and keep my current masters in files named with the date, then a very brief description of the place.  I keep those in folders named with the year.

But enough about me - what are you doing, and what do you plan to do? You have a stack of DAT's? Cassettes? Master CD's from FM broadcasts? Give us some details to chew and we can suggest more.

[Todd R]

Amadeus Pro is a good choice for Mac.  In addition to that, pick up Scott Brown's xACT, which is a great all-round utility.  Those 2 software packages are all you need.

Basic workflow:

- load the raw wav file in Amadeus
- cut out the beginning junk and end junk of the file you don't want
-  Perform any edits you want to the file

This is a big area, and too big to really get into in much detail at a general level.  Many folks prefer that no sound editing be done, but editing might include light compression, EQ, multi-band compression, High Pass Filtering (bass rolloff), etc.

- (Optional, my preference) Balance the overall output between channels.  Select All on the file, do an Analysis>Waveform statistics to get the stats on your music file.  Compare the average RMS power between channels and note which is louder and which softer and by how much (note that the RMS value is negative, so the softer channel will have a larger negative value).  Select all of the channel that is softest and boost its gain (Effects>Amplify) by the amount that its Avg RMS value is less than the louder channel (be sure not to create any clipping by boosting the channel by this much, otherwise lower the louder channel to make both channels match).

- Add fade-in and fade-out on the beginning and end of the file
- Normalize the file to increase the overall volume as much as possible. Select All on the file, Effects> Normalize:  Normalize to say 99.7%, don't check the "treat channels independently" box, reference to max sample value, not max RMS power.

- Add tracking/track markers: Selection>Mark Selection to insert a track marker.  You'll need to start by inserting a track marker at the very beginning of the file, then position the cursor at every other place you want a marker (beginning of each song) and insert another marker.  You'll want to be sure that markers and resulting song lengths occur at sector boundaries, otherwise you get a noise during playback of songs burned to CDR. To make sure this occurs on the last song, put a track marker 1/4 second or so before the end of your file. Once you track out, throw out this garbage track, which will insure the last song also fits within sector boundaries.

-Amadeus does not default to cutting tracks on sector boundaries. Once you have all of your track markers inserted, insure you track on sector boundaries by doing Selection>Modify Markers... and modify marker locations Align Marker Location at 75 Hz.

-Save your track markers for later use by using the File>Export Markers command

-If your file is 16/44.1k, then export individual song tracks using Sound>Split According to Markers command.  This will save individual song files, I generally save as FLAC here, but you can save as Wav files if you like.

-If you aren't at 16/44.1k (and assuming that is how you want to save and share), then you need to get there first.  If 24 bits, dither down from 24>16.  Amadeus comes with (or prompts you to download) a bunch of free VST and Audio Unit plugins.  Use the Effects>VST Effects> mda dither plugin in dither from 24>16 (Select All on your file first).  You could also use other dither plugins to do this. 

-Then you need to change the file characteristics to reflect your now 16bit dithered file from 24bit to 16bit:  Use the Sound>Characteristics command to change your file characteristics to 16bit.  Under the pop-up menu, select 44.1k sampling (assuming this is your original sampling, or 48K if that's what you used, or whatever), and change the bit depth to 16bits, check the Don't Re-sample check box (since you already dithered), and change the sound characteristics

-If you weren't at 44.1k to begin with and want to get there, Select All again, and again go to Sound>Characteristics command, select 16bits again, and then select 44.1k sampling.  This time, leave unchecked the Don't Re-Sample choice box since this time you do want to force a re-sampling from 48K>44.1K

- Go ahead and export the song tracks individually as above using Sound>Split According to Markers now that you are at 16/44.1k

Now you have your individual song files, and are done with Amadeus and can go to xACT (which I'll describe in another post).

[Todd R]

xACT workflow:

- Amadeus probably saved your song files as Marker01.flac, Marker02.flac, etc (or Song01, Song02).  Use the xACT utilities tab to change these.  Load all of the song files onto the xACT>Utilities tab, select Batch Rename and at the pop-up put in something like 'phish2012-03-21-AKG483-s1t" and hit ok.  This will rename your files sequentially phish2012-03-21-AKG483-s1t01, phish2012-03-21-AKG483-s1t02, etc.

- Not a bad idea to make sure you don't have any Sector Boundary Errors (SBEs), so load all of the now re-named files onto the Fix SBE tab.  This will check and fix any SBEs you have on your tracks.  If you use the Amadeus workflow above and be sure to modify markers on 75Hz and create a final junk track at the end that you threw away, you shouldn't have any SBEs anyway.

- Load all of the song files onto xACT's Tags tab to tag all of the files.  Load the files, select them all, then add in artist, album, date, etc into the tag fields. xACT can also auto-name the individual song titles, so use the Auto-name option and load on a list of all the song titles (titles only, no track numbers).  You can only do this if you saved all the individual songs as FLAC files, not Wav files (which is my workflow).  If you haven't, before you can Tag, you'll need to load the Wav files onto the Encode tab and save the files as FLAC files

-Now you've got now got tagged FLAC files of all your songs and are pretty much done.  Note:  I never burn to CDR, so this is all I need, but if you do like CDR, you'll need Wav files.  Either by saving that way to begin with from Amadeus, or by using the Decode tab of xACT to decode your FLAC files to Wav files.

-If you want to share via bit torrent, you can use the Checksum tab of xACT to make the needed FFPs or MD5s.

- If you like, you can also load all of the FLAC files (or Wav files) onto the Lossy tab of xACT and make a set of mp3 files for ipod listening.

[cabbie11]

thanks for the reply. Sorry if I was too vague. I'm used to editing photographs and have a workflow I follow (a series of steps). I just figured it would be the same for digital music files. So I was asking what steps people follow. Ex: place marker, EQ, then normalize etc But, after transferring a few DAT's I've decided to hold off on editing for now. I have a ton of cassette and DAT masters I want to archive and preserve. So getting them all onto hard drive will be my top priority for the next few months(hopefully).

thanks for the tips on naming and back-ups. I think your advice on naming files one of the best pieces of advice I've read on the forum yet! So many people never think about it until
it's out of control. So thanks for that. I was planning on using the naming structure I use for my photography, might as well be consistent! As for back-ups my plan is for two onsite and one offsite.

About me, I have a rather large pile of DAT and cassette masters dating back to the early 90's when I started taping shows. I have an even bigger pile of DAT's and cassette's that I got in trades over the years. But, the masters are my main concern right now. Being your typical anal taper type, I just want to make I get the best possible transfers to hard drive. Once they're on hard drive and I learn Amadeus Pro I can spend countless hours tweaking them to perfection!!

Thanks again, morst, for your advice,

Jarrod

[cabbie11]

WOW Todd! Thanks man, really nailed my question. I've come to realize I have many many hours of transferring before I get to editing. But, eventually I'll want to do some CDR's
and I would like to share some shows that maybe aren't in circulation right now.

I do have a couple more questions though. About 99% of my DAT's are 48K. I'm guessing it is best to transfer at 48K and save as a WAV for archiving. Then make a copy of that file for editing to CDR or whatever I use I want? Also, when transferring a Cassette what would be the sampling rate to record at?

Again, thanks for taking the time to help me. I really appreciate it.
 

[Todd R]

WOW Todd! Thanks man, really nailed my question. I've come to realize I have many many hours of transferring before I get to editing. But, eventually I'll want to do some CDR's
and I would like to share some shows that maybe aren't in circulation right now.

I do have a couple more questions though. About 99% of my DAT's are 48K. I'm guessing it is best to transfer at 48K and save as a WAV for archiving. Then make a copy of that file for editing to CDR or whatever I use I want? Also, when transferring a Cassette what would be the sampling rate to record at?

Again, thanks for taking the time to help me. I really appreciate it.

No problem cabbie, glad it helped.

If your DATs are at 48k (and 16bits by definition), you'd transfer them at 48k and save as a 16/48 Wav file.  You won't need the 24bit>16bit dither step above, but Select All on the file in Amadeus, change Sound Characteristics to 44.1k and choose the resample box to force Amadeus to resample the 16/48 file to 16/44.1k

Cassettes are analog, so you can do digital sampling at whatever you want:  44.1k, 48k, 96k.  Lots of debate if you search around here on what is the best sampling rate to use.

Given the frequency response of cassettes, I'd probably sample at 44.1k to make my life easier if I wanted to get it to 16/44.1k for use on CDRs.  Then again, I'd probably choose to have a bit depth of 24bits on transfer not 16bits, so what do I know?  (Really, if you are going pretty much straight from cassette to CDR, 16/44 is probably fine.  Bit depths of 24bits on the transfer would be helpful though if you start doing EQ/Compression/whatever sound editing.)