Audacity 32bit 96k to 16b 44k?

[clashradio]

Hello,

I'm using Audacity to capture my vinyl records at 32 bit float, 96K that uses it's own .aup file format, which then needs to be converted or "exported" to use with other editing programs. I want to export (convert) the file as a WAV at 16b 44k, but there seems to be no option to adjust the frequency or "Project" Rate Hz as it's called in Audacity. The drop-down option for WAV is worded as "WAV signed 16 bit PCM". I've gone in the Options mode, but there's a message that says "if you need more control over the export format please use the "Other uncompressed files" format, but that doesn't lead anywhere. So when I drop the converted WAV file in Sony Forge to edit, it's at 16b 96K. Basically I'm trying to save the multiple conversion. Why convert in Audacity, and then have to convert it again in Forge in order to burn it to a cd-r.

Or do I just capture the vinyl @16b 44k and call it good?

Thanks in advance for any help,

Jason

[poisondeathray]

You have access to other export options if you install the ffmpeg plugin for audacity, using "custom ffmpeg export"

[Brazil]

FYI resamplers comparaison: http://src.infinitewave.ca

[jagabo]

Audacity: Tracks -> Resample.

[redwudz]

Instructions on how to add ffmpeg library to Audacity:

http://manual.audacityteam.org/index.php?title=FAQ:Installation_and_Plug-Ins#How_do_I_...ort_Library.3F

[clashradio]

I was able to download the plugin, and now I do have access to the Export options. I'm a bit overwhelmed by all of the options. I selected what I thought was the most logical (Show All Formats = "wav" Show All Codecs = "wmav1") I entered 16 in the Bit Rate column, and 44100 in the Sample Rate column. There are of coarse many other codecs..pcm_s16le, pcm_u8 etc. When I export, it is just an empty file. Am I missing a step?

Jagab: yes, I could resample. I did try to resample at 16b 44k, exported, dropped it in Forge, and the file was still at 16b 96k.

Isn't it best to capture audio at the highest Bit and Sample rate, and then compress to 16b 44k, vs sampling at that rate?

[poisondeathray]

At the bottom left hand corner , thre is drop down menu for the project sampling rate. Adjust that after you resample , before you export

You're not really doing "multiple conversions" ; ie. if you export 96kHz and let SoundForge convert it to 16bit 44.1Khz, it's not an "extra conversion" compared to if you let audacity do the conversion then imported into SoundForge

What are you doing in SoundForge that audacity cannot do ?

[jman98]

Isn't it best to capture audio at the highest Bit and Sample rate, and then compress to 16b 44k, vs sampling at that rate?

In theory, yes, but if you are playing back at CD bit and sample requirements, it's debatable whether most or even any people could tell any difference. I know it sounds like the difference between recording at those high sample rates and bit rates vs. recording at CD rates so you don't have to convert will be massive and obvious, but really most people can't tell the difference. There are always some people who say that they can tell the difference and maybe they really can or just think so, but I'm not convinced that most people can tell any difference. However, there is an entire industry dedicated to the belief that everybody can indeed tell the difference that sells very costly audio electronics, $30 and up LP pressings on vinyl, and so on to the people who think they can tell the difference.

[Cornucopia]

LOTS depends on the SampleRate conversion algorithm. Back over a decade ago, and again about 5 years ago, I did an exhaustive SRC comparison/shootout. It was very clear that there were winners and losers, and you wouldn't want to do SRC with one of those losers - it would be noticeable and disturbing unless you're deaf or tone-deaf.
As long as you went with a good algorithm that uses "polyphase" FIR filtering, you'll probably do fine and not be able to notice.

What I suggest that you SHOULD be doing is, if you don't find the filesizes being larger, and your computer & A/D interface has no trouble with the higher samplerate/bitdepth files, it does make sense to capture at those improved specs, then do your processing, then FINALLY at the end, downsample & downrez to your regular, dynamic-range-optimized playback master (after keeping a safety copy of the improved, finished master).

You can do the editing and/or processing in whatever tool works best for you: Audacity, SoundForge, Audition, ProTools, Nuendo, Cubase, Samplitude, Reaper, etc. They just have to be able to support 96/24 (or similar, whatever you'ved used) NATIVELY as input. Sounds to me like you want to do some stuff in Audacity, then move it over to SF for some other stuff (processing w/ particular plugins, etc) and then finish & distribute. As long as you are going LPCM (or losslessly-compressed) WAV files (or flac, etc) and they're supported, it really doesn't matter how many times you bring it out and bring it back in.
Note, of course, that there are slight minor changes when you do this with processing (including the adding of dither), so you don't want to build it through TOO many generations, however, if your intent is to PROCESS the file, you already know the file is going to change somewhat.

Now, if you are trying to bring an HD audiofile into SF and are having trouble, you need to troubleshoot why. I have an older version on this PC (v8) and v9 on my laptop, and both of those have NO trouble natively using original 96/24/2ch files that I recording using my H4N, so I know it can do that. In fact, I tested as much while I was typing this.
My guess is the 32bit FLOAT subformat, which IMO is totally unnecessary as stored in an audio file. 24bit (integer) already stores 144dB worth of dynamic range (which you will never even get to make full use of as almost ALL A/D and D/A converters actually top out at ~21bits' worth = 120dB). 32bit made sense not to audio engineers, but to computer programmers, because it is an even 4Bytes (as opposed to 3Bytes of 24bit).
Now, 32float, 48int or float, or 64float makes lots of sense in the processing engine in order to avoid non-dithered rounding error distortion, but IMO to save a file as 32bit float is overdoing it as well as making it less compatible.
If it is the case that Audacity can ONLY support 32bit float (and not 24bit int) and SF can ONLY support 24bit int (and not 32bit float), then you have to change your workflow, either by using an intermediary conversion app, or by not using one of those 2 apps. If you aren't really sure, use MediaInfo to tell you what kind of file is being saved.

Scott

<edit>Just did another check, and looks like my SF8 can write to 32bit, so I'm guessing it can read it too. Maybe there's a non-standard implementation being saved by Audacity...or am I just way off base of what you've been trying to say?</edit>

[clashradio]

Scott,

Thank you for your info! What I'm trying to do is capture at a better rate of 16b 44k, even though in the end, it ends up at those specs. Audacity seems to be the only program I have that captures from an external source. Sony Vegas, Sony Forge, and even the program that came with my capture device (USB Phono Plus) do not capture. Once captured, I run SoundForge Pro-10, so I can eliminate low rumble hum via Wav Z Noize, and then I de-click. I'm sure you can do both in Audacity, but I like Forge. It's easy to make tracks markers, plus I stretch the wave form out so far, it really makes it un-noticeable when I cut out minute sections of the song where there was a big "click" etc.

So, from my turntable via directional RCA cables, to the capture device (USB Phono Plus), to my computer via USB into Audacity at 32 float/96k. Since Audacity has it's own file system (.aup) you have to convert as you export. I figured I should convert directly to 16b 44k. I'm not Hell bent on 32 float. I could use 24bit if it makes no difference? My version of Forge does recognize 32 float, 96k. So, if I export from Audacity using WAV signed 16 bit PCM, drop it in Forge @16bit 96k, I can then resample it down to 16b 44k for cd-r. Like you said, you have to be careful about too many generations.

I do plan on saving a data-dvd of the raw 24 or 32 bit/96k captures. And using a gold archive cd-r for the final, edited playback cd.

Which leaves me wondering what to do?... capture 16b 44k?, capture 24b 96k, then export at 16k 96k, then resample to 16b 44k.

If I could export at 32b 96k, then I'd probably save the edited WAV file at 32b 96k on a data dvd, then burn a 16b 44k cd-r.

I guess I need your expertise on which method to go. And then maybe how to export at my desired rates.

Jason

[Cornucopia]

If you could use SF to capture, it would make your life easier, right? I will check into that because you ought to be able to.

Scott

[pandy]

Use SoX (unless no further processing is planned - i.e. this is LAST step before writing CD)

@sox --buffer 131072 -S -D -V %1 %1-p16d.wav rate -v -s -I 44100.0 dither -f high-shibata -p 16

Such wav can be burn directly on CD (i.e. it is full compatible with CD Audio).

[Cornucopia]

OK, now I'm confused:

1. I see all my interfaces in both apps.
2. Audacity can be set to record in 2496 by default.
3. SF has NO trouble directly reading its (Audacity's) output, whether 1644, 2496, etc.
4. SF has NO trouble directly reading 32bit float files either.
5. The options to use those for WAV are available in Audacity (you do: Save as type "Other compressed files", Options: Header - "WAV (Microsoft)", Encoding - "Signed 24bit PCM", "Signed 32bit PCM", "32bit float", etc)

So unless you have a really strange device with non-standard drivers, you ought to also be able to use it in both apps. What's your audio device's model #?
Even assuming there's a problem there, you should have plenty of options without having to "downsample/downconvert" etc.

Scott

[clashradio]

"SF has NO trouble directly reading its (Audacity's) output, whether 1644, 2496, etc"

That can't be right? Audacity uses it's own file format, which then need to be converted to mp3, wav etc. In order for other programs to reconize the file, it has to be converted. When I was using Creative SoundBlaster, you could record as a WAV. Is there a way with Audacity?

My audio device (external sound card) is a Focusrite Scarlett 18i6, but I'm running the audio into the Phono Plus USB, not the Focusrite.

[Cornucopia]

I'm not talking about Audacity's "aup" format, which isn't just a media storage container anyway - it includes session project data (tracks/channels layout, edits made, filter chain history, etc) combined into one file. Personally, I think that theirs is a BAD way to save project data. All major DAW players use a separate "sidecar" project/session metafile to save to. This makes save time very quick, doesn't affect the media file(s), and is more easily parseable and translatable. And XML session metadata is where Parametric Editing has been going for all media.
So, I say you should stop thinking of what you are doing in an export as being a "conversion". It isn't, at least concerning the base media data, which is the important stuff here. If your project setting has set up the default recording/storage format as being 2496 and you export to 2496 WAV files, there is no "conversion" taking place, only Rendering of the existing state of the Session and re-wrapping into the WAV container. NO LOSS, except the loss of the project metadata.

If stuff is going through your USB Phono Plus box, you should set your Control Panel/Sound properties to "USB Audio Codec", just like it says in the manual (which I downloaded & checked). It looks VERY straightforward, you don't even need to install additional drivers, as it uses Windows stock USB drivers. I see above where you said, "even the program that came with my capture device (USB Phono Plus) do not capture". This tells me that you likely haven't gone through the all steps in the manual to correctly set up the device. Yes, that means that TEMPORARILY your Sound Mapper/Sound Manager chooses a different device than normal, but you can always set it right back again after you've done the ingest/capturing. Heck, I have 3 devices on this XP box I'm at right now that I swap regularly depending on the task (Mobo sound, Creative Audigy2 and occasionally Avid M-Box). It's not going to break things if you do that. I have a feeling this is what's at the heart of your trouble.

Scott

[clashradio]

The program that came with the USB Phono Plus is called Soundsaver Express, and I couldn't get it to work. I did some searching, even called the company and they said they didn't support Soundsaver anymore and said to install Audacity lol, which is what other people who couldn't get that program to work also did. You say you were about to download and get it to work? I'll try it again right now.

Ya, I admit I'm stuck on the export/conversion part. Just a little confused about it and trying to understand the best my little brain can comprehend.

***UPDATE*** I broke out the manual after my last post, then I remembered something about being limited to only 16b 48k anyways...so I did a few more searches on the sample rare. Below is an excerpt of a review on the ART USB Phono Plus from SOS.

"With a frequency response of 10Hz to 50kHz within half a dB either way, a THD of 0.01 percent typical at 1kHz and a noise floor 90dB below clipping, the USB Phono Plus is capable of doing a worthy job of transferring vinyl to computer, though the sample rate is limited to 44.1kHz or 48kHz and the resolution is just 16-bit. This isn't a problem if you record as hot as possible without clipping, but it's not quite as forgiving as 24-bit encoding. The USB 1.1 interface is compatible with Windows 98SE or later, Mac OS 9.1 and OS X, and will work with the audio device drivers built into Windows 98SE, Me, 2000 and XP and with any Apple computers that have USB support. No additional drivers are necessary, though ART do include a CD-ROM containing the audio freeware program Audacity, which is a simple stereo wave editor with some effect plug-in capability"

[Cornucopia]

So, other than that review getting the interface spec wrong (it's 2.0 instead of 1.1), that means you are limited by your source device ANYWAY. So the best you can START with is 16bit 44.1/48kHz. Still worthy, just not spectacular. If you plan on doing much processing, it still makes sense to convert it from 16/44-48 to 24/88-96 for the intermediate work.

I wasn't asking you to use the Soundsaver app (I probably wouldn't push ANYONE to that app), but it should be clear that the interface uses simple Windows stock drivers, which should make it available to ANY audio app, including SF. You just have to change prefs in the control panel.

If you have to continue with the workflow you are currently doing, it's not the end of the world anyway. I, however, don't like the capture "interface" in Audacity, SF and other apps make more sense, particularly if you come from an analog audio engineer background.

I wasn't about to DL Soundsaver: already have about 18 audio apps on this PC alone and don't need any more that don't have specialty features (NR, forensic analysis, etc). As I said, all my interfaces can be seen by ALL my apps (with the exception of the old v7.4 of ProTools that only works with the AVID device on purpose, but I knew that).

So your choices ought to be:
1. Figure out how to set the default input/recording device to your USB Phono Plus, and use SF only.
2. Continue doing things as you have done.
3. Use Audacity only.
4. Use some completely different set of apps.
5. Give up.

While I quite like Audacity, I'm hoping you'll try #1. SF Pro 10 is quite capable of doing lots of neat stuff, and ought to have NO problem with this.

Do you need step-by-step instructions that aren't in your manual?

Scott

[clashradio]

I feel pretty stupid... two reasons. 1) I think the 16b 48k limit was with the Soundsaver program, not the actual ART device itself, obliviously because I captured @ 32/96.

2) I figured out how to export to WAV @ 32/96, so then I can edit in SF. Like you said, I needed to get unstuck on the export/conversion process.

But I still can't capture directly with SF though.

Thanks for all of your help. I really appreciate it. You're a Prince man!!

Jason

[Cornucopia]

Let's be clear about this: if your DEVICE cannot do better than 16/48, do NOT kid yourself - you are only getting, at best, 16/48's worth of quality. The remainder of those 32/96 files are wasted on ZEROs or, more likely, NOISE.

I know that is the device's limit, as that is listed in the equipment's specs in the manual. Normally, the driver tells the app what it can and cannot do, and then the app sticks to choosing from within that available subset. Maybe Audacity ignores/overrides that and fits the limit within whatever it picks, and maybe this is part of the reason it is known to be so easily adaptable to the various devices. Just understand that it cannot create something out of nothing, nor undo a hardware's limitations.

Do me a favor, and in both Windows' Sound Control Panel and in SF's Options-Preferences-Audio tab, (making sure to take note of all the existing settings) try and play with and ALL the options & dropdowns. What do you see?

Scott

[hello_hello]

The drop-down option for WAV is worded as "WAV signed 16 bit PCM". I've gone in the Options mode, but there's a message that says "if you need more control over the export format please use the "Other uncompressed files" format, but that doesn't lead anywhere.

Haven't read through the whole thread, but was your original question answered?

Exporting while choosing "other uncompressed files" and then selecting "Options" should give you lots of choices.

[2Bdecided]

If you plan on doing much processing, it still makes sense to convert it from 16/44-48 to 24/88-96 for the intermediate work.

It's vinyl. It makes almost no sense to work at a higher bitdepth. 16-bits gives you more than 90dB SNR. Vinyl has about 70dB. You would have to do a heck of a lot of processing to create any measurable benefit to working above 16-bits, never mind an audible one.

It makes no sense to work at a higher sample rate either. Some processes benefit from higher sample rates, e.g. dynamic range compression (though most will use a higher sample rate internally, so you don't need to worry about it), but why would you apply that to a commercially produced recording that you've captured from vinyl? Most other processes don't really care.

Worry about cleaning the vinyl. Worrying about the stylus, cartridge and turntable. Worry about the pre-amp and A>D if you really must. But 24/96 when the source and destination are both 16/44.1?! Don't worry about that.

Cheers,
David.

[Cornucopia]

I was speaking generally WRT processing, not necessarily his current workflow (which may change). Also, I know your stance on this, and I both agree and disagree. Analog has, as you know, inherent medium/transmission noise, but while in one sense it has limited dynamic range & bandwidth, in another sense it has infinite bandwidth and dynamic range ("gracefully receding"). With noise reduction processing, it IS possible to make use of that residual resolution.
Also, higher samplerate largely has a tighter clock = less jitter = better sound.
And as I said before, it may not matter if 16/44 is limited on ingest, but going 24/96 or similar is OFTEN a good idea if one is doing much processing, in order to avoid cumulative noise/distortion buildup. Surely, you would agree with that?

I think you're going at it with a "it doesn't help & might hurt" attitude, while I'm using a "it doesn't hurt & might help" one. I would understand this a decade ago when processing power & storage space were at a higher premium, but now the additional burden for doing so is incremental.

I do VERY MUCH agree with you regarding care of the source, as that is the weakest link in the chain.

Scott

[clashradio]

Ya, I'm an anal bastard. I have read both cases, 16/44 vs 32/96 etc. I'm of the old school mind...quality, no mp3 . No transferring records just to put on I-pods. So I didn't mind the extra space needed for higher rez. Didn't know vinyl only has 70db. I knew that's why there's only about 30 minutes per side....to get the hottest sound.

I do agree about cleaning prior to recording. I don't go ballz to the wall & have them vacuumed though. Gruv Glide seems to work pretty good. It got rid of one nasty skip I had.

Wow, I feel educated....and saddened that it really doesn't matter about the higher rez.

[2Bdecided]

Analog has, as you know, inherent medium/transmission noise, but while in one sense it has limited dynamic range & bandwidth, in another sense it has infinite bandwidth and dynamic range ("gracefully receding").

In that same sense, digital also has infinite dynamic range as long as dither is enabled.

With noise reduction processing, it IS possible to make use of that residual resolution.

Fair enough, but with any analogue source, the artefacts of digital noise reduction (even the tens of thousands of dollars/pounds units) are much much higher than 16-bit quantisation noise. If you try to get the background noise that quiet, you're almost certainly over-using the noise reduction.

Also, higher samplerate largely has a tighter clock = less jitter = better sound.

I don't know how true that is. Some converters only work at one or two internal clock speeds: a very high multiple of 44.1kHz and/or a very high multiple of 48kHz. The 48, 96, 192etc kHz that you get into your PC is all generated from the same original high sample rate signal using down-conversion after A>D, and all converted to the same high sample rate before D>A. Hence the jitter is the same.

And as I said before, it may not matter if 16/44 is limited on ingest, but going 24/96 or similar is OFTEN a good idea if one is doing much processing, in order to avoid cumulative noise/distortion buildup. Surely, you would agree with that?

A higher bitdepth obviously reduces cumulative noise buildup when processing, but I think it's often overstated for what people do at home. It might be worthwhile for the most pristine digital home recordings, but it's irrelevant for most vinyl and cassette sources. A higher processing sample rate only helps certain specific types of processing. The kind of things you do in this context would only be affected if they were atrociously badly implemented in the software. Even then, for some other tools (especially some declickers), they work better at 44.1kHz and (again, due to being badly implemented) worse at higher sample rates.

I think you're going at it with a "it doesn't help & might hurt" attitude, while I'm using a "it doesn't hurt & might help" one. I would understand this a decade ago when processing power & storage space were at a higher premium, but now the additional burden for doing so is incremental.

If you have a 24/96 source, I agree there's no harm and some possible benefit working at 24/96. I certainly would. Whatever your source, there's no harm (and genuine benefit) working in 32-bit floating point. I usually do. But if you have a 44.1kHz source, it's generally a bad and pointless approach to upsample it, process it, and then downsample it again. Resampling should be benign, but sometimes isn't. If you're the type of person who wants to worry about barely-relevant things, then you should worry about resampling twice unnecessarily.

I do VERY MUCH agree with you regarding care of the source, as that is the weakest link in the chain.

Exactly. People worry about digital parameters and things in their PCs because they can change them easily, but the real problems are elsewhere.

btw, while more isn't essential, a device sold today that only does 44.1/16 may not be targeted at people wanting to do a great job, and may not sound as good as a better device (even at 44.1/16). I don't know. I have no experience with what the OP is using.

Cheers,
David.

[2Bdecided]

If it helps to gauge how well you're doing, here are some vinyl samples

These are the best vinyl samples I have ever heard:
http://www.mediafire.com/?59m192uz8mq09
from this thread:
http://www.hydrogenaudio.org/forums/index.php?showtopic=71960&st=50&p=683458&#entry683458
(note the extreme transfer method - really quite amazing)

48/24 44.1/16 comparison:
http://www.hydrogenaudio.org/forums/index.php?showtopic=93477

44.1/16 transfers restored:
http://www.delback.co.uk/general/vinyl_examples/vinyl_examples.html
test record:
http://delback.co.uk/turntable_tests/

good (except the Beatles one):
http://www.hydrogenaudio.org/forums/index.php?showtopic=78189

this is about as bad as it gets:
http://www.hydrogenaudio.org/forums/index.php?showtopic=65947&st=50&p=633478&#entry633478


Cheers,
David.

[clashradio]

Hey thanks David for the comparison link. Pretty intersing. I'm listing to the 24/96 clips thinking yeah!! this is it! then I play the 16/44....I admit I couldn't hear the difference.

And thanks to Scott for all of your time and info.

If the point is mute at 24/96, at least it makes for good conversation right?

[Cornucopia]

Found a good link that describes nicely the pitfalls of digital audio (and how to avoid them).
http://forum.benchmarkmedia.com/discuss/feedback/newsletter/2010/08/1/unique-evils-dig...ow-defeat-them
Particularly like the idea of 16/44 being "fragile", compared to 24/96.

Scott

[Cornucopia]

I wish I still had precision analog equipment with which to transfer stuff like that. What I personally have now is very MOR consumer, but it works (though I don't do my personal stuff anymore, it's all digital now).

What I thought would/should have done with those workflows, however, is use a better A/D. The Emu USB device is OK, but not even close to top of the line stuff from Antelope, Lynx, Presonus, or even Alesis. Same for the PreAmp (none was mentioned?).

Also, I'm mildly surprised at David for presenting them (though I know it's for comparison's sake) as examples, when they are in aac/mp4/m4a format. (I would have thought he would avoid lossy-compressed audio). If I really wanted to WOW somebody and elicit balanced examples of A/B comparisons, I'd want to have 24/96 LPCM WAV clips to offer.
But that's nitpicky and pretensious - those are good comparisons nonetheless.

I do understand all the hurdles involved with this kind of thing because I have also done some LP restoration. One example - my setup at the time (c. ~2002) was: Technics DD turntable (don't remember the model now), Shure V15type3 pickup, custom-built outboard phono Preamp, Alesis 24/96/2 A-D, into a ProTools 24/96 system, where I denoised (click, pop, hum, wideband, etc.), eq'd, adjusted dyn range and then downsampled to 16/44 for output to CD. The source was a locally produced EP which had had a bad pressing: EVERY pressed copy had a scratch/crackle throughout the beginning of the "hit single". In 1978/79, when it was originally produced, the problem was just ignored. That would be unacceptable 20-odd years later when I did the transfer & restoration. I only have the 16/44 copy now (production company retained my restored 24/96 master). Here is an AAC 16/44 copy for your enjoyment (BTW, it does still have a very small amount of that click - I didn't want to be too aggressive with the NR)...

Scott

[clashradio]

So this .m4a is a downsampled from your 16b 44k? Still sounds pretty good.

You say you eq'd...did you save a raw file without eq? I only ask because when you eq, you take the chance of changing the overall sound. I think you did a good job. I'd be nervous as my tastes my be different than someone else's. I like the tradition V style pattern.

Do you still have a copy of the original ep?

Oh, that's for that benchmarkmedia link. I'll have to re-read that article. Lots of info there!

[Cornucopia]

I just converted the m4a to give as an example. Still have the 16/44 wav CD masters.

No, the eq was basically done to un-eq the coloration of the LP (EP), just minor corrective stuff. Not even what I'd call full remastering, and still always under the creator's artistic direction.

Vinyl is not a straight 1:1 medium, you have to apply a number of dynamic range & pre-emphasis/de-emphasis eq techniques to make sure that the groove stays within certain modulation parameters. So all those things need to be un-done (hopefully, correctly matching what was done - which in this case was a crap shoot), to get back to some equivalent of the original master tapes. I'm sure on this project, I erred on the side of doing too little, rather than too much.

I do have the EP, though it is in long-term cold storage out of state. Hopefully, I'll be able to retrieve it all by this time next year.

I have a similar set of restorations that I did - from cassettes, no less - that, while they probably wouldn't win any quality awards on their own, I am quite proud of. Not least of which is that the artist involved was an old acquaintance of mine.
I contacted him to ask for permission/rights to use a copy of one of his children's songs in an educational video I was involved in. His response was: "SURE! I didn't know anyone still had good copies of that." I get all this info about master tapes being lost in a fire, etc. So I paid him for his sync rights licensing and then, gratis, found the 2 cassettes that he & his band had done 20 years earlier.

That one I DID need to do a lot of eq and "remastering" of. But I'm happy to say, they turned out well. Those were GREAT kids songs, too. Equally as good as anything you'd hear from Raffi or SL&B, etc. Not the kind of stuff you want to get lost to history.

************************

So, back On Topic, have you got the transfer/conversion stuff all worked out?

Scott