Topic: TASCAM DR-680 portable 8-track #6

[thunderbolt]

There's a lot of information in these threads, but there's also a lot of random chatter. Finding the key information within dozens of pages of posts isn't always easy.

I gathered up some comments and advice from the various 680 threads and put them in a Word document to help me figure things out.  I don't know if this will help or not.  I didn't plan on posting it, and not all the threads are attributed (apologies), but perhaps it might help newbies to the machine like me.  Here goes:

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UPDATE - I tried this: In Menu under REC > ST REC, I selected 'DIN'. Under I/O > INPUT SEL, I selected ANALOG (this refers to channels 5/6, even though it doesn't say so on the display) . Plugged in the S/PDIF out from an outputting DA-P1 , both set to 44.1, and the 680 indicates 'DIN LOCK'. So this looks good so far. But I'm not seeing any indication of input level on the 680 or hearing anything from the headphone monitor.

UPDATE II - OK, I figured it out: Under HOME/FUNC you have to select DIN MONITOR (i.e. un-select MIX MONITOR) to hear the S/PDIF input. Yahoo! 7/8 have mic inputs!

If I read it correctly, go to HOME/FUNC > I/O > INPUT SEL, the 'ANALOG/DIGITAL' option there is for channels 5 and 6. Channels 7/8 of course don't have an analog option and can only be either the 'DIN' stereo track or the stereo 'MIX' track (selectable in 'MENU' under REC > ST REC > DIN/MIX).

If you select 'ANALOG' there, you merely retain the 1/4" analog inputs to channels 5 and 6, but if you select DIGITAL, the digital input then goes to 5/6 instead of 7/8, and (I presume) channels 7/8 default to 'MIX'.

The setting in the DR-680's I/O menu for choosing digital or analog input on 5/6 is bypassed if you select digital for 7/8 by setting ST REC in the REC menu to DIN. If you've got DIN in 7/8, 5/6 will automatically default to analog input.

And while you can route digital to 5/6 and see/hear incoming audio on those channels without having to change the Functions/Monitor setting to DIN, you DO have to change that Functions/Monitor setting in order to see digital audio on 7/8. Squirrelly!

The person reporting that has already recanted.  1.12 does not remove spdif > track 7/8 recording capability.  You have to change the monitor settings to see the levels from digi-in because they are reset every time you power it off.

To record on 7/8, you need to go the the recording menu off the main menu, and set "ST REC" to "DIN", meaning digital input (as opposed to "MIX", for recording a mix of channels 1-6 on the 7/8 channels).

This is all you need to record digitally on 7/8. According to the manual, if set to "DIN" on "ST REC", on the I/O menu, it will not matter if you select "ANALOG" on INPUT SEL" or if you select "DIGITAL", either way, the 680 will record the analog channels on 5/6, since you have set it up in the REC menu to record the digital channels on 7/8.

I don't think the DIN MON settings or D OUT MODE have anything to do with whether you can record digitally.  They will affect whether you can monitor your digi input channels on the headphone, or output the digi input to the spdif coax output (for a patcher, etc), but I don't think they affect whether you can actually record off the digital input.

For recording the digital input on channels 5/6, the "ST REC" setting on the recording menu must be set to "MIX" (sending the 1-6 mixed signal to 7/8), and then also set on the I/O menu the "INPUT SEL" to "DIGITAL".

Thanks guys! Teddy actually filled me in on how to do this. To record digi-in I needed to set the 680 to Digital and Mix on the I/O instead of DIN, Stereo - Mix on the Main Menu shutting off 1,2,3,4, channels, and Mix Mon on the function screen. I now get the digi-in reading on the 5 & 6 channel, and I can monitor the levels there on 5&6, and the levels display in the mix side of the screen also.   


Yes, upon further testing I find that if the Digital Output setting of the SD702 is "Professional" (which their manual also refers to as AES), then the DR-680 can receive and record digital data at either 44.1 or 48k. (If the rate on the 702 and 680 don't match then I see "DIN Unlock" on the 680, which is good practice.)

If the 702 is set to "Consumer", and the 680 and 702 are both set to 48K, signal passes and recording happens OK.

But if the 702 is set to "Consumer", and the 680 and 702 are both set to 44.1, the 680 shows DIN Unlock and won't pass audio.

What actually appears to be happening is that the 702 is outputting, or the 680 is misinterpreting/enforcing, a SPDIF subcode bit that triggers the old "no digital copying at 44.1" copy prohibit protocol.


As for the other things in my previous post: yes, the 680 will record an incoming 44.1 stream from the 702 (with the 702 set to "consumer") when the 680 is set to 48k. But no sample rate conversion is going on. And no indication is heard in the audio that anything is amiss during recording.

But if I then play that file back, the data that streamed in at 44.1k is played back clocked at 48k, so it's higher in pitch and faster in tempo than the original audio. This is bad design practice. If the rates don't match, then DIN Unlock should show up to announce a problem.


Another point to note is that I initially found the monitoring on the DR-680 to be quite confusing, probably because I didn't read the manual closely enough for quite a few outings. My mistake was that I thought the Solo function was what I needed to use to monitor specific stereo pairs, but all that function does is output a single channel to both left and right headphone channels. Instead, you need to use the front panel "Mix Level" button and lower the mix level to zero of any channel you don't want to hear. That's quite a tedious process because if you only want to hear channels 1&2, you've got to manually turn down 6 other channels (or 4 or 2 depending on how many total tracks you've got going). I much prefer the monitoring on the R-44 where a simple button press rotates between 1&2, 3&4, or all four mixed together.

Quote from: achalsey on April 18, 2013, 04:16:59 PM
What signal does the digital out send?  Just the digital input, or can it send a mix?  Is there any way to send individual channels (analoge out set to "direct") then send two specific channels through the digital out?

The digital out sends whatever monitor mix you set as a panned and level adjusted mix of the 6 inputs.  If you set the machine for 'direct' output of each individual channel to it's analog RCA out, set the monitor mix levels to only output the two specific channels you want, one hardpanned left and the other right, with the other channel levels set to zero, it will do what you describe.
Quote
Also, if I set analoge out to "mix", and have a board feed into channel 5/6, then set all the other channels in the mix at 0, I would be have 3 separate outs giving the board feed, correct?

Assuming you don't have very old firmware, you would actually have board feed to all outputs.  If you don't pan anything you'd get the identical mono mix of ch5 & 6 on all outputs.  If you pan ch5 hardleft and ch6 hardright in the monitor mix, ch 5 would be routed to all the odd numbered RCA outs and ch6 to all the even ones.  The digital out would consist of ch5 left / ch6 right.

If you set the analog out to 'direct' you'd get the ch5 input on the ch5 output, the ch6 input on the ch6 output (regardless of monitor panning or levels), and 5 & 6 on the digital out, panned however you set that in the monitor mix.
« Last Edit: April 18, 2013, 05:56:06 PM by Gutbucket »


You can only monitor ch1-6 off the machine. You can only play back either ch1-6 or the stereo channel, but not all eight channels simultaneously.  It's really a six track machine with an 'extra' stereo channel and that's where that limitation shows up.

[kirk97132]

Service manual can be found in the user manual archive section of this site, posted in multiple sections:

http://taperssection.com/index.php?topic=147560.0
http://taperssection.com/index.php?topic=147559.0

I have a copy the full manual in a single PDF (~2MB) which I can email you if you like.

Switching between channels to adjust input gain with the pot is fast and easy using the channel selection buttons located under the meters.  It's more difficult in the dark by feel, but still doable.  Ganging adjacent channels together to adjust them all simultaneously with one knob while retaining there relative differernce settings is well worth having to select using the buttons IME, but I'm not a video guy.

The limiters (and low-cut) are implemented post ADC, so presumably they will not prevent clipping the analog input stage or converter unless there is a feed-back loop to the analog stage which isn't indicated.   There are circuit diagrams showing that in the service manual as well as a useful diagram showing levels though the system.

I use the Tascam snug cover which prevents accidental movement of the switches on top, but I do always make an effor to double check them before recording.  The top switches can be set to be locked out seperately from the front swtiches if desired, but I don't use that feature.

The short story on external battery powering seems to be not to use a supply that self-senses voltage output needs, but is either single output voltage only or hard-switched.

I know there are at least two version of the top board, maybe three.  I found that out the hard way ordering one.  I "think" it is based on manufacture date and I can't remember what the difference's were.  I have to dig back through my Emails to Tacam's tech dept.  And sadly I have a ton of them.  I want to say it has something to do with the power input design, but I am not positive after all these years.

[connloyalist]

I apologize if I missed it, but I don't remember reading in any of the DS-680 topics how exactly the gain on the DR-680 works. I was just wondering how the math works on the DR-680. Since you can go from -31 to +31 and low/high I am a bit confused how much gain it has.

I also own a DR-100mkII and seem to remember seeing somewhere that the Low-Medium-High button on the back adds 25dB at each step and the dial on the side adds 30dB at maximum level 10. So at "High" and "10" that would give 25+25+30 = 80dB of gain (does that sound plausible?). I usually run my SDC's (MJE-384K) at Medium-6. So that would be about 25+18 = 43 dB (dial at maximum is 10, which apparently is 30 dB. So each whole number is 3dB. For level 6 that would make 3x6=18dB).

Basic testing with my voice seems to suggest that the DR-680 at Low-6 gives about the same as the DR-100 at Medium-6, but I would like to have a ballpark idea of how it scales.

Regards, Christine

[lukpac]

I can't comment on specific numbers, but as I've noted before, if you're capturing low level performances (in my case, acoustic with dynamic mics), it's better to set the switch to High and turn the gain down than to set it to Low and turn the gain up. This is exactly the opposite of what the manual indicates, but the results will be significantly noisier with the switch set to Low and the gain turned up.

[connloyalist]

 

I can't comment on specific numbers, but as I've noted before, if you're capturing low level performances (in my case, acoustic with dynamic mics), it's better to set the switch to High and turn the gain down than to set it to Low and turn the gain up. This is exactly the opposite of what the manual indicates, but the results will be significantly noisier with the switch set to Low and the gain turned up.

So far I have only recorded acoustic instruments (either a 40-person community band or a big band). I don't own any dynamic mics (yet, but turning things over in my mind), but your answer suggests that the DR-680 does have enough gain to deal with these.

I do indeed seem to remember reading somewhere here about anything above Low/+20 it being better to switch to high and that you will find about the same amount of gain something like 15 points lower? So Low/+20 would be roughly equivalent to High/+5?

If I remember this correctly it suggests that the low and high scales have about 3/4 overlap between them. Assuming that each number is 1 dB, then -31 to +31 on low scale = 63dB + additional 15db on high scale = 78dB max gain. Which is pretty close to the presumed 80dB of the DR-100.

If my math on my DR-100 and basic testing between the DR-100 and DR-680 is correct, that comes pretty close. DR-100 medium 6 = ± 43 dB (apparently). DR-680 at low/+6 = -31 to +6 = 38dB. Plus perhaps some kind of offset. With my rough test that is probably close enough to be in the ballpark.

Or am overlooking something?

Regards, Christine

[lukpac]

I just checked using one mic and a splitter. High @ +23 = Low @ 0.

[achalsey]

I know virtually nothing about the technical side of things, so forgive my ignorance...but wouldn't 'unity' just be at zero gain?

[hi and lo]

More like giant can of worms...

To answer the question directly, I don't know. The DR-680 gain range is +/- 32 with zero being default. If zero was unity, then negative gain values would be attenuation while above zero is applying gain. I don't think this is the case and you should not be afraid to run below zero if required.

The concept of unity gain has been grossly misunderstood on this forum for years, myself included. It's a false assumption that 'equal in / equal out' sounds best because once you start understanding how opamps work and the basics of preamp design, the notion is immediately dispelled. Typically, opamps perform "best" (lowest noise and distortion) at higher gain ranges and their performance can be much worse at their minimum gain setting. In no way are they a straight wire at low/no gain and could actually be adding significant small amounts of noise.

The concept of Unity Gain took off [on this forum] because of a few recorders with poorly implemented mic/line inputs. Examples include portable DAT recorders like the D7/D8/M1/D100 or the Edirol R-09. I think the R-09 was really when this notion of unity gain went off the tracks. These recorders were simply not built to handle high-sensitivity microphones and/or loud SPL environments and their inputs easily overloaded. For the DAT recorders, the idea was not to run below 3-4 of 10 on the dial and for the R-09 it was not to run below 8 of 100 (or thereabouts) otherwise digital attenuation would occur. When a mic/line stage is poorly implemented like it was on these units, the best solution is to simply apply attenuation prior to the Mic/Line input.

These anecdotes are useful because they give the average user a frame of reference for operating their equipment, but do not imply that unity gain is the setting that produces the best possible sound. The concept of unity gain has been grossly misinterpreted by the entire community, to be honest. There's still the notion that a mic input stage colors the sound whereas a line input stage is completely transparent (i.e. a straight wire) and that that the best sound is produced at unity, which is simply wrong. Both mic and line inputs pass the signal through opamps prior to the A/D and will always affect the sound in some way. It's actually worse running at 'unity' once you understand that many/most opamps perform better at higher gain ranges, have at least some minimum gain (i.e 10 dB), and that in order to get 'equal in/out' you might actually be applying unnecessary attenuation.

The better question to ask is 'what gain value sounds best' rather than 'what value is unity' and the most likely answer to this question is the gain value that gets as close to zero dB without clipping.

[achalsey]

Thanks for the comprehensive answer.  I do remember DSatz posting an informative comment similar to yours and was considering mentioning the possible misnomer of 'unity gain' but don't know nearly enough about it all to be of any help.

[aaronji]

^^^ http://taperssection.com/index.php?topic=152494.0

Not sure if this is the thread you mean, but it has good explanations of both minimum safe gain setting and the "straight-wire" myth...

[Gutbucket]

Here's the DR-680 level diagram extracted from the service manual (page 19)-

[Carlos E. Martinez]

Perhaps I would risk using two words to what I think might simplify the 680 level settings: overload and noise. Please do not forget this is an oversimplification I'm doing.

Overload usually happens at the first stage, due to wrong gain settings. In the 680 the only gain settings are High and Low. Dynamic mics usually need High gain. Care should be taken with condenser mics, as some need High gain and others Low gain.

Noise usually happens when levels set by volume pot are mishandled. Shure had to go deep into this with their portable mixers, and explain quite well how to set levels, because users opened their level pots too much, and made a fame of Shure mixers being noisy. Later on they improved on this, by using a pot that set gain and levels together. But people still misused the Master pot. Peak leds helped warn you of overloads, and the 680 also has clip warnings.

I don't think distortion levels are increased if the gain levels are set too low. What does increase is noise levels, which might be considered some kind of distortion too, in extreme cases.

One good thing about the Tascam is that the level setting (Trim) is after just one chip stage, so at least it doesn't add up the noise of another stage. The other good thing is that there is a way, explained in the manual, that shows when distortion is happening at the input or further on in the chain. So it's a question of setting the gain level and see what happens below on the meter, and then setting the volume pot and see that the meter levels don't surpass the upper line.

But..., and on this there might be people who disagree, I believe eventual overload peaks for the first stage won't always be something bad and should help keep noise low.

[kirk97132]

I can't comment on specific numbers, but as I've noted before, if you're capturing low level performances (in my case, acoustic with dynamic mics), it's better to set the switch to High and turn the gain down than to set it to Low and turn the gain up. This is exactly the opposite of what the manual indicates, but the results will be significantly noisier with the switch set to Low and the gain turned up.

Just bear in mind that the possibility to overload the inputs exists like this.  A mistake I learned the hard way.  I had been running things exactly like you stated, and in the midst of a day long event and multiple performers, I had a group and after seeing the levels being too high turned them down.  The first clue should have been that I was down to the 1 and 2 range.  Since I didn't monitor with headphones it "looked" ok but when I played it back later, there was significant distortion.  As you get used to the 680's levels you'll find this easy to avoid.  And while the 680 has less than stellar readout for levels, that line around -14 is well placed and works great for a rough mark especially in the 24 bit realm.  Nowadays, I almost always have my input levels between -4ish and plus 5ish.  If I see that my levels are too high then I back down the sensitivity switch on the top.  Using this method I have never had an overloaded input.   As for the statement "eventual overload peaks for the first stage won't always be something bad".  My expereinces with Tascam recording devices, like the 680 & HD-P2 (which are newer recorders relatively speaking) is that they do not respond well to "overs".  And when these recorders do clip they sound ugly and are unforgiving.  I'd say they hove no "headroom" but since it's the digital world that is an outdated and highly inaccurate term.  The best recordings I have pulled with the tascam decks is to run them at higher input levels, be it sensitivity or trim pots, but not to push then past that.
And the "unity gain" talk...I agree with the posts that say it's misunderstood and too much value is placed on it.  I believe this topic became a hot button when there were many more people running external preamps and the goal was for the recording deck to not"add" anything to the signal it was receiving.  hi and lo nailed it with 'what gain value sounds best' instead of what is unity on my deck or deck xyz. 

[lukpac]

Just bear in mind that the possibility to overload the inputs exists like this.  A mistake I learned the hard way.

I should probably run tests at various levels sometime to compare, but it's worth noting that at lower levels the difference in noise isn't subtle. There's a *definite* advantage to recording using the "High" setting, at least within a particular range.

I had been running things exactly like you stated, and in the midst of a day long event and multiple performers, I had a group and after seeing the levels being too high turned them down.  The first clue should have been that I was down to the 1 and 2 range.  Since I didn't monitor with headphones it "looked" ok but when I played it back later, there was significant distortion.  As you get used to the 680's levels you'll find this easy to avoid.  And while the 680 has less than stellar readout for levels, that line around -14 is well placed and works great for a rough mark especially in the 24 bit realm.  Nowadays, I almost always have my input levels between -4ish and plus 5ish.  If I see that my levels are too high then I back down the sensitivity switch on the top.  Using this method I have never had an overloaded input.   As for the statement "eventual overload peaks for the first stage won't always be something bad".  My expereinces with Tascam recording devices, like the 680 & HD-P2 (which are newer recorders relatively speaking) is that they do not respond well to "overs".  And when these recorders do clip they sound ugly and are unforgiving.  I'd say they hove no "headroom" but since it's the digital world that is an outdated and highly inaccurate term.  The best recordings I have pulled with the tascam decks is to run them at higher input levels, be it sensitivity or trim pots, but not to push then past that.

Are you saying your recording was clipped? Or that peaks were below 0dBfs but were still distorted? Because overload in the input stage wouldn't have anything to do with digital clipping (unless of course the trim was set high enough to cause clipping). Are you saying you had the gain set to "High" and the trim set to +1 or +2? And the meters were well below 0dBfs? Or am I not understanding your scenario?

At any rate, digital "overs" and overloading the input stage are two different things.

I just checked using one mic and a splitter. High @ +23 = Low @ 0.

Oops. I guess that should be Low @ +23 = High @ 0.

[hi and lo]

Interesting. I don't have my 680 anymore, but seem to recall that the "High" gain setting was considerably noisier throughout the entire gain range and would run on "Low" with the gain cranked up. Maybe I'm just not remembering things correctly.