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Gear / Technical Help => Recording Gear => Topic started by: Jema on November 19, 2011, 11:39:51 AM
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I recently bought a SD mixpre for work on film productions. My current recorder is a FR2-LE, and one obvious downside with this rig is not being able to use the mixpres balanced professional line output through XLR, as the LE can only take an unbalanced consumer level input (meaning tape out from the mixpre, 1/8 tele). This got me looking at recorders that were able to take a hot line input (such as the M10 or PMD661), but I've had some difficulty finding information about wether this actually has any affect on the noise floor. Apart from the line in comparisons done by Guysonic, the following post is the only one I've found that says something about it.
http://taperssection.com/index.php?topic=148612.msg1900251#msg1900251
I believe you are misinterpreting the specs. A pro mixer with +4 out(Nominal) would have max levels around +24
The specs for the DR-07 and like consumer recorders say max input line level +6, +4, -4.. doesn't make any difference.. these are all -10 nominal levels, and could use some kind of pad or -10 output from a pro mixer.
Has nothing to do with signal to noise.. just level matching.
Guysonics tests does show a difference, but what's the cause? Is it mainly the dynamic range/snr of the A/D? Also, does a professional line level signal give better snr than consumer level, or is it just a matter of level matching with no affect on the result?
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Also, does a professional line level signal give better snr than consumer level, or is it just a matter of level matching with no affect on the result?
In most cases it is just a matter of level matching without an audible change in SNR. The unbalanced line from the Mixpre to the FR2-LE will probably have a greater effect on the SNR.
So why not just try it out since you've got the gear? Connect the Mixpre to the FR2-LE, match the levels, and do some recording. You can use either the Mixpre's Tape Out or pins 1 & 2 of the Line Outs. If noise is a problem then you will have to find another solution. I suspect everything will be OK.
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Sorry for the time it took to respond.
I have of course tried using the mixpre's tape out into the LE (the main output is too strong), and I don't think I would have a problem with noise (maybe sometimes interference though, going unbalanced). I'm more curious to find out if there's any real differences out there, or if it's just splitting hairs.
I know most of todays recorders are very quiet compared to older equipment, and when compared to the noise in microphones and preamps it's often not something to worry about. But this is assuming that you record at somewhat high levels, which is often the case with music. I'm a film student, and I often record very quiet locations and need to have a lot of headroom to avoid unexpected peaks. It's not unusual to be at -20dBfs. The 95dB snr of the LE becomes in effect 75dB. This is probably still good enough, but is there a line?
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Youre good to go bro. If I were you, just to cut down on size, I'd sell that LE in a heartbeat and buy a Sony M10. I like mine so much I just bought a 2nd M10 for my other rig. And I went from an SD 722>M10, and havent looked back not even once ;)
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I went from an SD 722>M10, and havent looked back not even once
That's quite the downgrade! It was justified I'm shure, regarding price and size, and from what I've read you'd be hard pressed to tell the difference when using them as simply a bitbucket with a good mixer in front. Working with film though, I would go for the SD for the reliability.
I've been wanting to buy an M10 for quite some time, but it's almost double the price in sweden and I can't justify it. I bought my LE used for a lower price than the M10 new (I have never seen a used M10 here). Buying from the US is not worth it with shipping and taxes.
Regardless, I don't think I would want to let go of my LE even if I got an M10. Since the LE's preamps are pretty good, I can use it as an all in one and use my mixpre in front of an H2 or similar when I'm hard pressed for more than two channels. Still waiting for that good deal on a reliable multichannel recorder...
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Sorry for the time it took to respond.
I have of course tried using the mixpre's tape out into the LE (the main output is too strong), and I don't think I would have a problem with noise (maybe sometimes interference though, going unbalanced). I'm more curious to find out if there's any real differences out there, or if it's just splitting hairs.
I know most of todays recorders are very quiet compared to older equipment, and when compared to the noise in microphones and preamps it's often not something to worry about. But this is assuming that you record at somewhat high levels, which is often the case with music. I'm a film student, and I often record very quiet locations and need to have a lot of headroom to avoid unexpected peaks. It's not unusual to be at -20dBfs. The 95dB snr of the LE becomes in effect 75dB. This is probably still good enough, but is there a line?
I run the Mixpre tape out -> M10 all the time. I was initially worried about it being unbalanced, but I've never had a problem. Just use a high quality interconnect that is as short as is practical for your use. I think both Darktrain and Ted Gakidis will mail to Europe. I run Ted's cables exclusively now. IMHO good cables are one of the least expensive ways to improve your rig. Plus Ted's cables are just plain sexy!
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Fostex is the best of the small units. The clock is better.... it'll sync with film.
No problem running unbalanced.. has nothing to do with the quality of sound.
All these recorders are unbalanced internally anyway.
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Better clock you say? This is interesting news to me. I tried searching on it a bit but couldn't find anything. What are you basing it on?
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Fostex is the best of the small units. The clock is better.... it'll sync with film.
No problem running unbalanced.. has nothing to do with the quality of sound.
All these recorders are unbalanced internally anyway.
well it will for about 1 hour.
It's still a great light weight unit.
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well it will for about 1 hour.
What happens at the 1 hr mark?
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well it will for about 1 hour.
What happens at the 1 hr mark?
I have done a good bit of audio for video using both the FR-2le and the sd722. Syncing with video sources from GL-2's and panasonics and sony pro camera's.
When working in vegas syncing audio (fr-2le) and video sources I notice after about 70 minutes the waves start to loose sync. I am quite used to this when using lower quality recorders (they loose sync in about 30 minutes) So the quartz,crystal,oscillator what ever you wanna call it, in the FR-2LE is good in my opinion. My SD 722 does not loose sync with video at all (never worked with a sync job over 2 hour chunks).
Peace.
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When working in vegas syncing audio (fr-2le) and video sources I notice after about 70 minutes the waves start to loose sync. I am quite used to this when using lower quality recorders (they loose sync in about 30 minutes) So the quartz,crystal,oscillator what ever you wanna call it, in the FR-2LE is good in my opinion.
This is good news for those doing a lot of interviews and documentary footage on a budget! What other cheaper recorders are you comparing to?
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When working in vegas syncing audio (fr-2le) and video sources I notice after about 70 minutes the waves start to loose sync. I am quite used to this when using lower quality recorders (they loose sync in about 30 minutes) So the quartz,crystal,oscillator what ever you wanna call it, in the FR-2LE is good in my opinion.
This is good news for those doing a lot of interviews and documentary footage on a budget! What other cheaper recorders are you comparing to?
Well I also use a Oade modded PMD620, this unit is amazingly clean, however the oscillator is not the best, I find myself re-syncing in about 20 minutes.
Peace.
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Well I also use a Oade modded PMD620, this unit is amazingly clean, however the oscillator is not the best, I find myself re-syncing in about 20 minutes.
If you don't use time-code, and don't have to re-sync/stretch sources, consider yourself lucky. Any two A>D devices will have sightly different clock rates if they're not locked. Can't blame one or the other if they don't match up.
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^ Exactly. I think the earlier comments are misleading regarding whether a devices clock is good or bad. The only way to ensure synch is to use pro equipment and lock with time-code. Dissing one piece of gear over another because 'the clock isn't good' simply because it doesn't have a clock that's exactly parallel to another clock is arbitrary because no two clocks are gonna parallel forever.
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This is correct. It's the relative difference in any two clocks that determines how long it will take for the difference to become noticable.
Interstingly, if you playback two sources simultaneously independantly, using the same machines the recordings were made on, and manually sync the two at the begining, they will stay in that relative sync for a very, very long time. In that case the difference in timing between the two clocks is self-compensated for the most part since the same clock is being used for recording and playback, leaving only the absolute variation within each clock to cause things to drift, and that is miniscule.
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Relative difference - lesson learned. Though I suppose you could talk about the stability of clocks in comparison to a perfect one. If the two imperfect sources both have clocks that are close to this, then it should sync a bit longer.
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Interstingly, if you playback two sources simultaneously independantly, using the same machines the recordings were made on, and manually sync the two at the begining, they will stay in that relative sync for a very, very long time. In that case the difference in timing between the two clocks is self-compensated for the most part since the same clock is being used for recording and playback, leaving only the absolute variation within each clock to cause things to drift, and that is miniscule.
If only the PCM-M10's analog output didn't sound like butt!?! >:(
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This is correct. It's the relative difference in any two clocks that determines how long it will take for the difference to become noticable.
Interstingly, if you playback two sources simultaneously independantly, using the same machines the recordings were made on, and manually sync the two at the begining, they will stay in that relative sync for a very, very long time. In that case the difference in timing between the two clocks is self-compensated for the most part since the same clock is being used for recording and playback, leaving only the absolute variation within each clock to cause things to drift, and that is miniscule.
I have to ask - how does one achieve that feat?
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I have to ask - how does one achieve that feat?
It's really for playback listening only. I don't use it for mixing the sources, but stretch and sync on the computer for that.
Make sure you are playing back the files on the same recorders that were used to record them. Start by picking a region with spoken word near the begining of the recording. Count-offs, sparce music like drum stick hits can also work, but spoken word is by far the best as it is easier to hear subtle delay differences as well as much easier to hear gross absolute position errors. Pause one recorder at the start of that. Start the other playing and when that section rolls past un-pause the other. Determine which is slightly ahead in time. Use fast double clicks on the pause/play button of the machine farther ahead in time to delay that source by fraction of a second increments until the two are in close sync with no echo or smearing. You'll hear it and learn to get good at it. You may go a bit too far and have to switch to delaying the other, or rewind and re-try if you run out of speaking. You'll get a feel for what the minimal play-pause-play delay time length is and will begin to know if you are close enough to hit the sync or if you'll need to try again. If the two different sources are playing back on seperate speakers, it can help to get up and put your head between the two to help hear which is ahead and fine tune the sync. It may help to unplug the feeds to the other speakers. You may have one machine you can play-pause-play faster, with less incremental delay, so aim to begin with that one further ahead in time for better accuracy control.
Doing it requires concentration and a good ear, looks crazy and makes people smile to watch. I've sucessfully sync'd 3 seperate playback decks this way for multi-channel surround playback which is a PITA, but syncing just two machines is pretty easy once you get the hang of it. The trick is learning to hear minimal millisecond delays in the Hass time range that aren't long enough to be echos, and perfecting the very fast double jab on the play/pause button. Once sync'd that way the machines will remain in sinc until the file split, which usually seems to throw one deck off slightly even if playback across splits is supossedly 'seamless'.