Tascam 388 question

Painful100 said:
Through experimenting I was able to answer some of my own questions. Sorry some of my questions were so unclear. Regarding the first question, I assigned the tracks using the pan/l-r knobs, and kept the red assign button down so I could eq the mix being sent to tracks 7/8. Everting went fine and I probably should have left it at that, but was wondering what the difference would be if the red assign l/r was depressed on tracks 7/8. Maybe that would allow more eq effect to print on those tracks. When I track a band, the red buttons are up and I mix the headphones with the monitor section. My friend records with the red buttons depressed. Maybe it’s the same means to an end?

Regarding the second question, I think I discovered my answer. Thanks for explaining the next parts.

I have another question that hopefully I can word well. So I have a m-1b line mixer. I read a recent post in which you commented, but there’s another poster who suggested using that line mixer to expand effect capabilities. Follow the instructions on that posting, I was able to hook up the line mixer with separate spring reverb unit. What I’m trying to figure out is can I have 3 or 4 global effect units hooked up to the tascam 388 ready for use without having to fuss too much behind the unit. Obviously the machine has 2 options, effect and Aux... but my limited understanding is that can be expanded with the line mixer. Is this achievable and if so, how would I would I wire things up.

And... can I hook up a second pair of external monitors to the M – 1B? I definitely will be experimenting with all of this as well.

Thank you very much
Click to expand...

So many times I wish I could just talk to people face-to-face wherever their gear is at to answer questions...

Regarding the L-R assign switches and your first paragraph, what I need to understand that you aren't answering is this: do the channels that have the L-R assign switches latched also have one or more PGM group assign switches latched, or are you saying you are able to record to tape tracks with only the L-R assign switches latched? If it is the latter, there is something wrong with your machine. The L-R assign switches assign signal to the L-R main STEREO buss. There is no pathway from the STEREO buss to any of the tape tracks, unless you physically patch the STEREO OUT jacks to PGM input jacks. Now, the L-R assign switch ALSO acts as a solo-in-place function IF there is also one or more PGM group assign switches latched on a channel that ALSO has the L-R assign switch latched. So if you have PGM group assign switches latched on channels 1, 2 & 3 and ALSO have the L-R assign switches latched on channels 1, 2, & 3, then you will have channels 1, 2 & 3 routed to both their respective PGM groups AND the L-R main STEREO buss. Under those circumstances any other channels with ONLY L-R assign switches latched will be muted in the main mix. This is one way to monitor your signals being recorded during recording, but that's not really the way Tascam intended you do it...they intended for you to use the monitor mixer to monitor your PGM groups during tracking, overdubs and punches.

Regarding your second question about the M-1B, if I understand you correctly, what you are wanting to do is use the limited number of AUX busses (the AUX buss and the EFF buss) to simultaneously feed a greater number of send type effect processors and then sum the outputs of those processors into a mono or stereo signal...is that correct? Or to have the whole mess hooked up all the time and be able to pick and choose what processor you want to use without having to mess with connections on the 388 backplane?

That would be addressed by using a patchbay. And then hooking up the outputs of the effect processors to the M-1B. Or if you want to feed them all simultaneously there are two ways to accomplish that:

1. Connect all the processors in parallel...use 'Y' cables to divide the AUX and/or EFF master send into multiple feeds and connect those to the inputs of your effect processors. As long as the inputs of those processors are high impedance there shouldn't be a problem driving them with the AUX and/or EFF send outputs. Also keep in mind you'll only have singular control of the output level of those auxiliary busses, so your effect processors will need to have input trim controls to fine tune the input levels at each processor. Then you could connect the outputs of the processors to inputs on the M-1B, and use the M-1B to sum those multiple signals in to a mono or stereo submix and bring that back into the 388 either using EFF RTN 1 or EFF RTN 2 or, like we talked about in an earlier post you could also bring that sum back using one or more of the 388 mixer channels and have control over assigning that sum to the main mix and/or tape tracks.

2. Daisy-chain your effect processors; serial connection. Connect the AUX or EFF output to the input of one processor, the output of that processor connects to the input of the next and so on. Yes this will bring a different result than feeding all processors in parallel as in #1, but this is another way to do it. And yes you need to experiment and think about the order in which the processors are chained. But, again, this is not a weird concept to execute. With this method you don't need the M-1B to sum the effects because they were never split into multiple channels.

I hope I'm understanding something about what you are asking.

Can you share a link to this post your talking about where there were instructions for how to use the M-1B to expand effect capabilities?

Also, IIRC there are application examples for the M-1B in the 388 manual...maybe...maybe I'm thinking of the M-512/M-520 manual...can't look right now. You have the manual right?

And hooking up a second pair of monitor speakers to the M-1B outputs...to monitor what? If you're using the M-1B to sum your send effects, that's what you'll be listening to on the second set of monitors. And you could do that instead by using the 388 mixer to monitor wherever you are bringing the effect sum back into the 388...if its via a couple channels on the mixer section you can use the L-R assign switches in their dual purpose role to solo-in-place any of the 388's 8 channels. What do you want to monitor? And are you wanting the second set of monitors just so you have a couple different types of speakers to use in the course of mixing? Or a "studio feed" for playback monitoring for the talent in the next room? Having multiple monitor speaker sets hooked up is a really common and smart technique to test your mix as you go, but you usually have the second set connected to the same outputs as the first set, and have some way to switch between the two in real time. Is there some reason you are wanting to monitor something different than whats being monitored on your main set of monitors?
 
“do the channels that have the L-R assign switches latched also have one or more PGM group assign switches latched, or are you saying you are able to record to tape tracks with only the L-R assign switches latched?“

It’s the former.

“ Or to have the whole mess hooked up all the time and be able to pick and choose what processor you want to use without having to mess with connections on the 388 backplane?”

Yes I’m basically trying to figure out how to hook it all up so that I don’t constantly have to shift around behind the unit. And I have a ph-32h patch bay I’ve haven’t used yet. I’d like to have multiple effects units hooked up and be able to switch between them easily, listen to them on global effects aux/eff sends, and print them to individual channels as needed.

“ And hooking up a second pair of monitor speakers to the M-1B outputs...to monitor what?”

Yes I want to hook up a separate pair of speaker monitors for listening back. But I got a hosa switcher now so I can plug another set of speakers in. But I’m currently using the rca stereo outs for my monitors, and the monitor outs for a tascam mh-40 headphone amp. So that leaves the xlr stereo outs (can you use these at the same time?) which I wanted to hook up to a laptop. So there are no more ways to come out of the tascam 388.

And yes I have the Manual. Which seems limited on some info. But also i just need to learn the concepts. Here is the link to the thread I mentioned. AUX IN/OUT QUERY re: TASCAM 388
 
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You are dealing with a technology that is already pushing the limits and to have some person set things up wrong because he does not have the right test tape is a big mistake. You can not randomly change things in these machines and think everything is going to be alright- that does not work that way. If you have a dBx noise reduction capability why is that person trying to squeeze 2dB more out of he deck. It all sound to me like he does not have the right test tape and is taking short cuts.

What has to be understood here is that you are putting audio on cassette width tracks and with that comes all kinds of things that are to be considered about pushing levels. If you send in high levels and get negative results- well? You are much netter off using a Tascam 38 to do higher levels as the track width is double. The 388 was a gimmick they came up with and it is not made for doing production like some people think. it is after all what is called a porta studio until you go to lift one. No Technician at Teac in Chicago liked working on that thing. They treated it like the A5300 or A5500 reel decks.
 
So i finally uploaded samples of the bass tracks (link below).

Sweetbeats, I am still actually having the "pshh" issue when recording bass. Even with line levels low, and Vu well below 0. I have 2 different samples for you. The first (track 4 - harmony bass) is recorded with a 69 Harmony bass, with space echo effect, through a aguilar tonehammer DI box. The second track (track 6 - P bass) is recorded with a P Bass via an orange amp head DI. The orange amp head DI is so hot when it comes in that I have to have the gain on 1. Neither track has any EQ on it. And If i take out around 4k, i can get the "pssh" sound to nearly disappear.

Thanks for your advice, and I responded to your previous questions above.

Skywave, i appreciate your opinion. Should I ask if he has the right test tape? I understand that this machine has limitations. Im really using it as a learning tool since I got a screaming deal on it. So far, i have been learning a lot. Give a listen to those bass tracks and let me know what you think could be wrong. Thanks

LINK: bass test tracks by Jon Payne 100 | Free Listening on SoundCloud
 
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... ...

I think that mixer channels with white PGM ASSIGN and red ASSIGN L/R buttons depressed simultaneously will "auto-mute" in strange ways to prevent an internal feedback loop from developing, also referred to by sweetbeats as "Solo-in-Place".

It's nature of the beast with an all-in-one unit like the 388, where internal signal paths are shared by different subsytems.
 
Painful100 said:
...I am still actually having the "pshh" issue when recording bass. Even with line levels low, and Vu well below 0. I have 2 different samples for you. The first (track 4 - harmony bass) is recorded with a 69 Harmony bass, with space echo effect, through a aguilar tonehammer DI box. The second track (track 6 - P bass) is recorded with a P Bass via an orange amp head DI. The orange amp head DI is so hot when it comes in that I have to have the gain on 1. Neither track has any EQ on it. And If i take out around 4k, i can get the "pssh" sound to nearly disappear.
Click to expand...

Per the manual, the TRIM knobs do not apply to signals connected to the 1/4" unbalanced LINE INPUT jacks. The TRIM knobs only apply to the mic amp and mic level signals conneted to the XLR MIC INPUT jacks. So it doesn't matter where you set the TRIM knob for your channel connected to the Orange amp DI output. So you have to adjust it at the amp. Is there an output level control on the Orange amp DI output? If there is, are you saying that's turned all the way down and the input signal at the 388 is still too hot? And what is telling you the signal is too hot? Does it sound distorted? Is the overload "OL" LED triggering? if neither is true, then what are you worried about? If it IS distorted sounding at the input or the OL indicator is lighting, and there is no way at the Orange amp to further attenuate the output, then you'll need to put some sort of an attenuator inline between the amp DI out and the 388 input. But I'd be *really* surprised if there wasn't a way to adjust the signal level at the source...or maybe there really isn't a problem here and the signal seems loud to you, but you control that with the channel fader on the 388. If its hot, its hot. So what. If its hot and overloading the line amp or and/or sounds distorted then, yes, that's a problem. What model Orange amp are we talking about here?

The sounds samples are really quiet so its difficult to hear what you are talking about...I *think* I hear it, and honestly what I think I'm hearing is maybe related to bias...I would suggest as an experiment to increase the bias level by 10mV on the track you are using that has the "pshhh" results, try it again, and see if there's any difference. If no difference, step it up by another 10mV and test again. Make notes each time of what you are hearing both with any distortion (that's what I think the "pshhh" is), as well as HF response, though that will be difficult to hear with a bass track. But if we assume the bias level may be low, then as you increase the bias level distortion will decrease until it reaches a minimum, and then at the "knee" point you'll begin to attenuate the HF response. That's what setting the bias does and there's a sweet spot where distortion is at a minimum and HF response is not yet impacted. This may not be your issue at all, but given the information you've shared, if it was me, that's what I'd be monkeying with. This isn't meant in any way to indicate I think your tech did anything wrong. If your tech is honest he'll tell you setting the bias on a 388 kind of sucks because its a two-head machine so there's no real-time monitoring of the outcome of changes in bias level (on a three-head machine you can put a track into record with incoming tone, and tweak the bias trimmer while monitoring the repro head and watching the VU meter for that knee point), and to make matters worse because of the format (8-tracks on 1/4" tape) it is difficult to use certain bias level setting methods like the IM distortion (i.e. "bias rocks") method, and at the very least its no fun recording tone, playing back, noting the VU meter level, tweak bias level a little bit, record tone, play back, note the VU meter level, teak bias level blahblahblah until you think you've hit the knee point, and now do that for 7 more tracks. So that's why Teac put the method for setting bias in the manual of just measuring the bias level straight off the bias amps...mmmmm...how 'bout 150mV? Yeah that sounds good. But remember different tape types require different bias levels for optimum performance, and even two tapes of the EXACT same make and type can have different bias level optimums from batch to batch, and it is KNOWN that there is a SIGNIFICANT bias level requirement difference between what Teac originally spec'ed for the machine (456), and something like LPR35. Years ago Jimmy in analog support at Teac in Montebello, CA alerted me to that, and the manual doesn't tell you that or tell you what to do instead to properly set the bias. So that's why I spent hours some years ago doing the best I could to identify the optimum bias level using the "bias rocks" method, and that's how I came up with 110mV-ish for LPR35. And don't forget, you have to measure the bias amp level with a meter that can actually read frequencies accurately at the range of the bias signal, which on the 388 is 100kHz. This means you need specialized equipment to check/set the bias level on a 388. Most meters cannot accurately read even beyond 1kHz. But the "pshhh" sound reminds me of the sound I would hear when I was doing that bias level experiment with LPR35. The "bias rocks" method involves recording a subsonic LF tone and listening for what sounds like rocks knocking together. I know. Sounds weird. Its more like the sound of large cobblestones tumbling. So you adjust the bias level until that sound is minimized. As I minimized that sound with my adjustment of the bias level I could then hear a "pshh-pshh-pshh" sound consistent with the subsonic tone (7Hz is what I think was used?), so I kept tweaking until that was minimal, but then there was a HF "hahhhhhhhhhh" sound that started to diminish, and when I tested with program material I found that if that "hahhhhh" sound was too diminished as a result of trying to make the "pshh-pshh-pshh" sound totally go away, I was impacting my HF performance...too much bias. So I found a compromise with some "hahhhhh" and some "pshh-pshh-pshh"...no cobblestones tumbling though. Again, IIRC I used 7Hz as the test tone so I wouldn't expect to be getting the "pshhh" at low frequency program material within the frequency range of the machine...I'm just saying what I can hear on your sample track reminds me of what I heard during my experiment, so I wonder if there is not quite enough bias. Its tough to get all this stuff to work on something like a 388. Its a limitation of the format, period. People love their 388s and I'm not gonna steer anybody away from that. They are really fun to use, and can sound nice, and I do like them, but its this kind of stuff that makes them a pain in the arse to work with. You might find even bumping the bias level by 5mV gets you the results you need if I'm right and its contributing to your problem. And that's why I'm making it clear I'm not knocking your tech. I suspect your bias level is very close to what it should be, and if it is a little low the trade-off is you will have better HF performance and I seem to recall you mentioning your tech mentioned really taking some time to get the machine in spec response wise...again, this is a challenge on a 388. And you are challenging its limitations with a bold bass line. BUT...here's something else to consider:

You ARE working with tape. You ARE working with a less professional format of multitrack tape. You may be getting the best you can out of it for a bass track...AND...man, I can barely hear that on your test track...that "pshh" is going to get COMPLETELY buried in the mix. Have you ever listened to raw multitrack master tracks from actual songs that have been released and are famous that were tracked on analog tape through analog mixers? There is typically all SORTS of noise and grit and shit in there that you can hear if you turn it up. And that's happening not on just one track but accumulating across, 8, 16, 24 tracks? Can you hear it in the final release mix? I can't...that's why I'm surprised to hear all the crap on the multitrack masters. But it is part of the analog beast. The digital age has made us too averse and sensitive to "noise". There's a track on Jeff Buckley's "Grace" album...I can't remember which one it is, maybe Mojo Pin...but at the beginning there's a a solo instrument track or tracks and then right before the rest of the tracks come in you can hear the "hsssss" as the engineer pushes the faders up on the rest of the tracks...and, I dunno...I *like* that. It feels real and intimate, like I'm in the club or on a couch in the control room. I'm not saying go out and find some noise, but that whole album sounds incredible from a mix and engineering standpoint on top of the brilliant performances and writing, and I have to assume that "incredible" is in part (in addition to pure talent) a by-product of whatever gear they used to track and mix, and so is the noise. So, I love it.

Lastly, what happens if you track the bass with dbx off? The resultant noise floor might bury the "pshhh" so you can't tell anymore, but I'm wondering if it goes away without dbx. That would answer Dave's suggestion it is related to the dbx processing.
 
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