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Gear / Technical Help => Ask The Tapers => Topic started by: battery on May 11, 2011, 11:13:40 AM
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After reading hundreds of posts about SBD source recording in Kickdown Central section,i find 99% folks here choose SBD -> 2-track flash memory potable recorder or 4-track recorder such as R-44
Q1: what's the advantage of mentioned above over the SBD -> soundcard -> laptop?
Q2:how does "SBD -> 2-track flash memory potable recorder" work? Is recorder equivalent to bucket？
Q3：I have owned one sony pcm m-10, what's my best solution for AUD and SBD source? buying another m10?
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It all comes down to ease of use and how you run a aud +_ sdb source because after you mix the two and if you use two decks you will have to sync the sources later. When I owned my r4 it was easy to mix w/ vegas even when there was delay because the sdb was back past 25' from the stage so I was able to mix pretty fast. Plus I did not like to edit my raw source to much mostly because I did not know how to make it sound better. Your second question about why SDB and aud? I found SDB to be sterile and boring and liked aud source better but some times the vocals or other parts don't come up in the PA so the SDB boost the recording ;D I always tried to go no more then 50%-50% but I prefer 75%-25% or so with aud out front at a good sounding venue.
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I prefer two different recorders because I often don't set my mics up anywhere near the soundboard. With a single 4-track recorder, you eliminate having to sync the sources later, but you're limited to running your mics near the SBD, else have a long run of cables getting one of the signals to your single recorder. I don't have any problem syncing the sources later (in my case, from an HD-P2 and an M10).
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Q1: what's the advantage of mentioned above over the SBD -> soundcard -> laptop?
The average PC soundcard doesn't have a great Analog to Digital converter (ADC). Most anything that's built for the purpose of recording will be better than a PC soundcard.
If you get a computer interface (Edirol UA-5 or one of hundreds of others), those are much better than the line in that's built into your laptop. If you get a four channel interface, that means you can have 4 channels on the same clock, which is great. But, have you ever noticed you don't see many laptop tapers? A lot of us tried that, and came to the conclusions that it's a hassle.
Q2:how does "SBD -> 2-track flash memory potable recorder" work? Is recorder equivalent to bucket？
Most people think of the recording chain as follows:
Microphones > preamp (perhaps more than 1) > ADC > storage device.
Sometimes preamp and ADC are built into one box (like a UA-5 or V3, etc). Sometimes the Preamp/ADC/Storage is built into one device, like on an M10.
We generally use the term "bit bucket" to mean a strictly digital input device that grabs the digital bits from the ADC and writes them to disk/card. It catches a bucket full of bits from a digital source. So most people would say when using SBD > M10, the M10 is not a bit bucket. It does more than that.
Q3：I have owned one sony pcm m-10, what's my best solution for AUD and SBD source? buying another m10?
When doing a matrix mix, there are 2 tasks when you get at home.
a) Aligning the sources left to right at one point in time, typically the beginning of the show.
b) Shrinking or stretching one source so it runs at the same speed as the other. 2 boxes will have different clocks, and there will be drift from one to the other. It might be something like 1/2 second over the course of an hour, but you have to deal with it.
Getting a 4 channel box like R4, R44 means (b) goes away, because they share the same clock. It also makes the (a) alignment easier because it will be a smaller, more known amount. Having 2 similar recorders frequently (but not always) means that the amount of drift will be very small, perhaps small enough that you can ignore it. Getting another M10 is a good option for this reason.
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good points Joe...
Q3 I have had limited success against drift running 2 recorders (even 2 M10s)
you really will prefer having both your clocks on the same recorder
after selling my 680 (read the thread) I went back to 2 machines and did care for doing that much work in post
I recently purchased the Tascam DR-2d to try running SBD/AUD on one recorder
and I have the advantage of having 2 outputs on my Littlebox to run both my recorders
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^^^ I have to agree, as someone who runs SBD/AUD pretty frequently, and who has also tried to sync in yet more sources (in this case, from another R-44), I've had limited success doing that sync in post, even after reading the thread.
If you're serious about doing a lot of 4-channel recording the DR-2D is the absolute cheapest way to go; the R-44 is the best price/quality mix IMHO, and above that you have some very expensive Sound Devices machines. I would avoid something like the DR-680 like the plague (darby and I had the same issue).
If you run the DR-2D, keep in mind that it won't have the sturdy inputs of the bigger decks, and also that you'll probably have to get a separate attenuator for the very hot SBD signal. Still, not a bad compromise, especially if you only record this way occasionally.
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^^^ I have to agree, as someone who runs SBD/AUD pretty frequently, and who has also tried to sync in yet more sources (in this case, from another R-44), I've had limited success doing that sync in post, even after reading the thread.
What thread are you referencing? Am I over-simplifying things? Here's my process for syncing the two sources:
1. Open MIC source in Cool Edit Pro and make sure the timeline is displaying in samples
2. Identify a distinct point near the beginning and zoom in enough to identify it down to the sample. That's point A.
3. Identify a distinct point near the end and identify the sample number. That's point B.
4. Subtract A from B. That's the length of the selection in samples.
5. Open SBD source in Cool Edit Pro 2
6. Identify the same points A and B, subtract their sample numbers, yielding the length of the same selection on the SBD source
7. Divide the two lengths, which will yield a ratio of either just over or under 1 depending on which number you divided by which (it doesn't matter)
8. Multiply that ratio by the sample rate to yield the sample rate that you need to adjust to.
9. Use r8brain to resample the shorter source by the ratio greater than 1, OR the longer source by the ratio less than 1
10. Line up the resampled source with the other source
An example (I keep notes of all of my work):
Last year at a show I acquired a KM140 > HD-P2 MIC source and a SBD > M10 SBD source. In CEP2, I determined that the length from points A to B of the MIC source was 295,825,752 samples. The same A-B on the SBD source was 295,818,010 samples. My HD-P2 source is always longer, and I always divide the bigger number by the smaller number. Doing so results in a ratio of 1.000026171496. Multiplying this by the sample rate of 48,000 gives 48001.2562318. That's the rate that the shorter (SBD) source needs adjusted to. After letting r8brain chew on that at the highest quality setting (takes just a few minutes), I opened the resampled file in CEP2 and re-measured A-B. The result: 295,825,754 samples. Exactly two samples difference from my HD-P2 source. And ready for lining up using any single point that's easily identifiable.
This whole process takes as little as 15 minutes and has never taken more than an hour. The worst I've ever done (on a first attempt) is to end up with the resampled source still differing by ~200 samples, which, at 48000 samples/second, is about 4 ms. Many wouldn't even notice this (myself included), but I give it another try and invariably end up with something matched to less than 50 samples (1 ms) without ever spending any more than an hour on it.
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We generally use the term "bit bucket" to mean a strictly digital input device that grabs the digital bits from the ADC and writes them to disk/card. It catches a bucket full of bits from a digital source. So most people would say when using SBD > M10, the M10 is not a bit bucket. It does more than that.
I still can't use English as native lauguage, especillay not familiar with some terms. >:( >:(
Based my knowledge :P，the signal from SBD's line out is digital source which has been processed by the preamp and ADC built in SBD or some connected to the SBD on the stage, which we can come to a conclusion that m10 is a bit bucket. ;D
Is there anything wrong with description above? ??? ???
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Q3：I have owned one sony pcm m-10, what's my best solution for AUD and SBD source? buying another m10?
………………2 boxes will have different clocks, and there will be drift from one to the other. It might be something like 1/2 second over the course of an hour, but you have to deal with it.
Is the clock mentioned here workclock generated from Crystal Oscillator Circuit?
But the 2 similar recorders such as two m10 should have the same workclock frequency,
if not, that means there is a flaw in the Crystal Oscillator Circuit of the device, right? ???
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If you run the DR-2D, keep in mind that it won't have the sturdy inputs of the bigger decks, and also that you'll probably have to get a separate attenuator for the very hot SBD signal.
1. why can not i have the sturdy inputs of the bigger decks?
2.Do i need a separate attenuator? if SBD->M10.
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Last year at a show I acquired a KM140 > HD-P2 MIC source and a SBD > M10 SBD source.
Thanks for ur warm and detail response, and valuable experience.
One more question: i don't know anything about the timecode of the HD-P2, will it give some help with post-matrix?
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It looks like differet points of view on this topic occur.
Some stand in the queue of "limited success", others stand in the queue of "no problem syncing"
Look forward to more discussion
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But the 2 similar recorders such as two m10 should have the same workclock frequency,
if not, that means there is a flaw in the Crystal Oscillator Circuit of the device, right? ???
Two different M10s would have similar, but not identical, clock frequency. It's not due to flaws in the devices, but rather sample variation within specified tolerances.
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^^^ I have to agree, as someone who runs SBD/AUD pretty frequently, and who has also tried to sync in yet more sources (in this case, from another R-44), I've had limited success doing that sync in post, even after reading the thread.
What thread are you referencing? Am I over-simplifying things? Here's my process for syncing the two sources:
1. Open MIC source in Cool Edit Pro and make sure the timeline is displaying in samples
2. Identify a distinct point near the beginning and zoom in enough to identify it down to the sample. That's point A.
3. Identify a distinct point near the end and identify the sample number. That's point B.
4. Subtract A from B. That's the length of the selection in samples.
5. Open SBD source in Cool Edit Pro 2
6. Identify the same points A and B, subtract their sample numbers, yielding the length of the same selection on the SBD source
7. Divide the two lengths, which will yield a ratio of either just over or under 1 depending on which number you divided by which (it doesn't matter)
8. Multiply that ratio by the sample rate to yield the sample rate that you need to adjust to.
9. Use r8brain to resample the shorter source by the ratio greater than 1, OR the longer source by the ratio less than 1
10. Line up the resampled source with the other source
An example (I keep notes of all of my work):
Last year at a show I acquired a KM140 > HD-P2 MIC source and a SBD > M10 SBD source. In CEP2, I determined that the length from points A to B of the MIC source was 295,825,752 samples. The same A-B on the SBD source was 295,818,010 samples. My HD-P2 source is always longer, and I always divide the bigger number by the smaller number. Doing so results in a ratio of 1.000026171496. Multiplying this by the sample rate of 48,000 gives 48001.2562318. That's the rate that the shorter (SBD) source needs adjusted to. After letting r8brain chew on that at the highest quality setting (takes just a few minutes), I opened the resampled file in CEP2 and re-measured A-B. The result: 295,825,754 samples. Exactly two samples difference from my HD-P2 source. And ready for lining up using any single point that's easily identifiable.
This whole process takes as little as 15 minutes and has never taken more than an hour. The worst I've ever done (on a first attempt) is to end up with the resampled source still differing by ~200 samples, which, at 48000 samples/second, is about 4 ms. Many wouldn't even notice this (myself included), but I give it another try and invariably end up with something matched to less than 50 samples (1 ms) without ever spending any more than an hour on it.
that's a very mathematical - and interesting - easy approach to sync'ing two sources - I like your explanation!
thanx
--Ian
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Yes, your thread. It's super-helpful and I know that many have had success with it. I just have not. I also don't find identifying an identical point in both sources down to the sample level all that easy. Audacity may also not be the greatest software for this as the zoom on it I find a bit uneven compared to, say, Audition.
^^^ I have to agree, as someone who runs SBD/AUD pretty frequently, and who has also tried to sync in yet more sources (in this case, from another R-44), I've had limited success doing that sync in post, even after reading the thread.
What thread are you referencing? Am I over-simplifying things? Here's my process for syncing the two sources:
1. Open MIC source in Cool Edit Pro and make sure the timeline is displaying in samples
2. Identify a distinct point near the beginning and zoom in enough to identify it down to the sample. That's point A.
3. Identify a distinct point near the end and identify the sample number. That's point B.
4. Subtract A from B. That's the length of the selection in samples.
5. Open SBD source in Cool Edit Pro 2
6. Identify the same points A and B, subtract their sample numbers, yielding the length of the same selection on the SBD source
7. Divide the two lengths, which will yield a ratio of either just over or under 1 depending on which number you divided by which (it doesn't matter)
8. Multiply that ratio by the sample rate to yield the sample rate that you need to adjust to.
9. Use r8brain to resample the shorter source by the ratio greater than 1, OR the longer source by the ratio less than 1
10. Line up the resampled source with the other source
An example (I keep notes of all of my work):
Last year at a show I acquired a KM140 > HD-P2 MIC source and a SBD > M10 SBD source. In CEP2, I determined that the length from points A to B of the MIC source was 295,825,752 samples. The same A-B on the SBD source was 295,818,010 samples. My HD-P2 source is always longer, and I always divide the bigger number by the smaller number. Doing so results in a ratio of 1.000026171496. Multiplying this by the sample rate of 48,000 gives 48001.2562318. That's the rate that the shorter (SBD) source needs adjusted to. After letting r8brain chew on that at the highest quality setting (takes just a few minutes), I opened the resampled file in CEP2 and re-measured A-B. The result: 295,825,754 samples. Exactly two samples difference from my HD-P2 source. And ready for lining up using any single point that's easily identifiable.
This whole process takes as little as 15 minutes and has never taken more than an hour. The worst I've ever done (on a first attempt) is to end up with the resampled source still differing by ~200 samples, which, at 48000 samples/second, is about 4 ms. Many wouldn't even notice this (myself included), but I give it another try and invariably end up with something matched to less than 50 samples (1 ms) without ever spending any more than an hour on it.
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I also don't find identifying an identical point in both sources down to the sample level all that easy. Audacity may also not be the greatest software for this as the zoom on it I find a bit uneven compared to, say, Audition.
It's not always easy, but the tough ones are the ones that take a few tries to get as close as ~50 samples difference (in my experience), which I consider to be plenty close enough. And the hour it might take me to accomplish that is nothing compared to the time I spend applying volume envelopes, etc., to adjust the mix throughout the recording.
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^^^ I have to agree, as someone who runs SBD/AUD pretty frequently, and who has also tried to sync in yet more sources (in this case, from another R-44), I've had limited success doing that sync in post, even after reading the thread.
What thread are you referencing? Am I over-simplifying things? Here's my process for syncing the two sources:
1. Open MIC source in Cool Edit Pro and make sure the timeline is displaying in samples
2. Identify a distinct point near the beginning and zoom in enough to identify it down to the sample. That's point A.
3. Identify a distinct point near the end and identify the sample number. That's point B.
4. Subtract A from B. That's the length of the selection in samples.
5. Open SBD source in Cool Edit Pro 2
6. Identify the same points A and B, subtract their sample numbers, yielding the length of the same selection on the SBD source
7. Divide the two lengths, which will yield a ratio of either just over or under 1 depending on which number you divided by which (it doesn't matter)
8. Multiply that ratio by the sample rate to yield the sample rate that you need to adjust to.
9. Use r8brain to resample the shorter source by the ratio greater than 1, OR the longer source by the ratio less than 1
10. Line up the resampled source with the other source
An example (I keep notes of all of my work):
Last year at a show I acquired a KM140 > HD-P2 MIC source and a SBD > M10 SBD source. In CEP2, I determined that the length from points A to B of the MIC source was 295,825,752 samples. The same A-B on the SBD source was 295,818,010 samples. My HD-P2 source is always longer, and I always divide the bigger number by the smaller number. Doing so results in a ratio of 1.000026171496. Multiplying this by the sample rate of 48,000 gives 48001.2562318. That's the rate that the shorter (SBD) source needs adjusted to. After letting r8brain chew on that at the highest quality setting (takes just a few minutes), I opened the resampled file in CEP2 and re-measured A-B. The result: 295,825,754 samples. Exactly two samples difference from my HD-P2 source. And ready for lining up using any single point that's easily identifiable.
This whole process takes as little as 15 minutes and has never taken more than an hour. The worst I've ever done (on a first attempt) is to end up with the resampled source still differing by ~200 samples, which, at 48000 samples/second, is about 4 ms. Many wouldn't even notice this (myself included), but I give it another try and invariably end up with something matched to less than 50 samples (1 ms) without ever spending any more than an hour on it.
That's pretty much how I do it too. For me, this kind of math makes perfect sense, but I have an engineering degree. Some people really struggle with this kind of math. Just like when someone who can play guitar fluently shows me a few chords and says "that's all there is to it, go play". For me, it's not that easy to play. We all have our different abilities.
But that's how it's done. If you can handle the math, great. If you can't, no shame, but you are probably better off to buy a 4 track recorder.
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Also, I just realized I did NOT use your method - I used the other one that involves stretching the time. I don't have r8brain, but maybe I should get it. Your method seems easier.
I also don't find identifying an identical point in both sources down to the sample level all that easy. Audacity may also not be the greatest software for this as the zoom on it I find a bit uneven compared to, say, Audition.
It's not always easy, but the tough ones are the ones that take a few tries to get as close as ~50 samples difference (in my experience), which I consider to be plenty close enough. And the hour it might take me to accomplish that is nothing compared to the time I spend applying volume envelopes, etc., to adjust the mix throughout the recording.
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I don't have r8brain, but maybe I should get it. Your method seems easier.
http://www.voxengo.com/product/r8brain/
The maximum file size for the free version is 2 GB, so you have to chop a larger file in half before resampling if you don't want to pay for the full version. Just plug your calculated new sample rate (I always go to 8 decimal places) in the drop-down-looking box. Your base/reference rate goes below it (48,000, or whatever). I always choose very high quality because it only takes a few minutes, and I've been using 32-bit float for the output with seemingly fine results. I've never really tried to understand what the whole 32-bit floating workspace is, but I know CEP2 utilizes it so I've been using this option in r8brain. Would love an explanation of this from somebody.
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Based my knowledge :P，the signal from SBD's line out is digital source which has been processed by the preamp and ADC built in SBD or some connected to the SBD on the stage, which we can come to a conclusion that m10 is a bit bucket. ;D
Is there anything wrong with description above? ??? ???
Yes. line out signal is NOT digital. Digital (SPDIF/AES, etc) is not often found as an output option unless it's a digital console (which is increasingly common, but not THAT common yet). It's usually analog, so the M10 would be doing the ADC and would not technically be a bit bucket.