Darren

Correct me if I'm wrong, but, I believe that there is a very popular automated software solution that uses editorial's EDL's and the metadata from the sound files to re-conform the sound to picture.  It's Titan.  I know that this has been used with  success on at at least the last 4 or 5 projects that I have worked on. It is definitely in the sound post house's best interest to have an automated conformation process.  Conforming can come out of their time and budget.  They would rather get the conforming done in a couple of minutes instead of a couple of days so that they can get on with the real work and meet their ever tightening deadlines. Most of the sound houses that I have dealt with are well aware of this. I know that Titan reads the iXML info, and it may well read the Bext info as well. If I'm wrong about this, please accept my apologies.

It might be very interesting to hear from as many posters on this board as possible on how they feel about still using Sound "Rolls".

Courtney, Thank you for the explanation. Personally, I use a "ROLL-SCENE-TAKE" file naming scheme. I suppose it's a shame that, while some of them can, not all of the recorders out there are able to encode the Roll Number into their metadata. I find that, at the very least,with the possible exception of the Sound Devices recorders, rolls can be quite easily organized on any media using folders that are named after that particular roll. That being said, I certainly do not have experience with all of the NLR's out there, so this may be an issue with other recorders as well.Thanks again Courtney, Darren

Thank you. I can see how this would be fine for editorial, but I can also see how it would be a problem for telecine.

So If one completely eliminated roll numbers, has it been discussed how one knows what sound report goes with what disc/Cf card, folder, or what have you??? Why is there a movement underway to eliminate sound rolls? I feel that they still have relevance, at least the way I work. I find them to be more of a convenience than an inconvenience. I'd be very appreciative if someone would please fill me in on the history of this. Thanks, Darren

So I was fine using the 01V96 to feed 4 simple tracks of AES/EBU to the 744T. The trouble began when I started trying to sync the different digital devices, lightpipe, firewire, etc. ***No, the trouble really began when you decided to try and clock from the 744. This is NOT your fault as your line of thought was a correct and you were doing it for all of the right reasons. This is a design flaw of the 744, but in all fairness to SD, this is a fairly new issue that is now becoming more common as more of us use digital mixers. The PD4 worked the exact same way.*** What you are saying makes some since although I'd be lying if I said I completely understand why the 744T wants to kick into slave mode with the incoming AES/EBU signal. *** See above*** I did cruise the word clock thread too, but not until after I got into my own deep muck all alone. OTOH, troubleshooting this web of cables and menus (or trying to rather) has helped me in other ways relating to familiarity of the general interfacing between the components, and writing it all down in my earlier post also helped to gel the specifics in my fading mind. It may not be rocket science, but it is starting to seem like it and I guess it IS computer science when it all goes digital. I only hope I'm a little smarter when this is all done and working... ***NONE of this is as complicated as it currently seems, you are relatively new to this and it will come to you quickly. Be fair to yourself. You can't expect to just know all of these things, give yourself a chance to acquire the necessary knowledge.*** My testing of Boomrecorder plays back well. At least it sounds good for the up to 5 minute runs I've been doing. Of course if I have drift involved it could get ugly over time. The disconcerting part is I have no way to check for this. ***At the very least, you should listen for pops and clicks and reduced transparency of the upper and lower frequencies. If clocking is particularly bad, you will get distortion or momentary (or longer) dropouts. Record some tones and listen to them. Pops and clicks that are often very easy to hear with tone can be very difficult to hear with dialog. This would give you a false sense of security. I'm not sure about boom recorder, but Metacorder allows you to monitor AFTER the signal has returned from the computer (not the hard drive, the computer), if you always monitor in this fashion, any clocking issues should be immediately apparent. *** So If I can't master word clock from the 744T due to the AES/EBU inputs, would it be better to master TC alone from it and WC from the MOTU instead? I know there is a MOTU clock/preamp upgrade I can have done, and I plan to look into it one of these days, but I'm hoping to get this all working for the interim without kicking down more $. ***Yes. I would experiment. Try using the MOTU as a masterclock first, then if you are not satisfied, use the Yamaha. You will have success. I have recently switched to the Metric Halo 2882 as I found it to have better clocking. BTW, i highly recommend taking an ANALOG feed out of your recorders for the tape returns back to your Yamaha. Then if you hear any clocking issues you will KNOW that it is on the recording side of things, not on the tape return side. I do recommend that you TRY the following: If it works, it will be the best solution for you and will guarantee you sample accurate T/C. Feed the t/c from the 744 directly into the Motu. Set your MOTU's clock source to SMPTE; it will then clock itself off of the highly accurate timecode provided by the 744. Then clock the Yamaha from the MOTU. The 744 will automatically then clock itself from the Yamaha. Ensure that you have set the Yamaha's clocking menu to clock from incoming word clock as the Yamaha has LOTS of clocking options. I am fairly certain that this 'should' work. However, it 'may' not as it indirectly still relies on the 744. I do feel that this will work, but if it doesn't, go back to the above mentioned approaches. *** The main reason I have tried to do it as I have is so i can maximize the use of the Ambient Clock in the 744T. I saw you commented on the same idea in the other word clock thread. I know it's the best clock i've got in the system by far. ***Please lobby with SD on this; I have.*** Boy, this sho is thirsty work. Too bad I drank all the ales while playing with this last night! So what would you suggest as a plan of attack? ***More Ale*** -Darren

If I am correctly visualizing your system, on the surface it looks like you have NO master word clock at all! I believe that you are trying to use the 744 as a word clock master via BNC, while all the while the 744 is trying to slave off of the 01V96 via the digital audio being sent from your Yamaha's AES card. The 744 CANNOT be used as a master word clock while being fed digital audio. As soon as it sees digital audio at an input, it defaults to slave mode. SO, in my opinion, you are currently suffering from highly unstable system clocking as you have no master. If I am correct, I can offer you a couple of solutions, just let me know if you'd like some more assistance. Cheers, Darren

Actually, the Traveler definitely is a digital mixer. It has 8 inputs and 8 outputs and multiple mixes that allow its outputs to be used as aux sends, or as feeds to comteks, or what have you. It, and other similar interfaces, can be also used with a control surface to provide you with the necessary fader control, etc. It even has a dedicated talkback function. What most of these devices are missing is DSP. Without it, there is no eq or dynamics control. -Darren

Curious about the state of recording and what percentage of us now use a computer and a firewire interface, for either recording or playback? Poll results thus far will be shown to you immediately after you vote. Cheers, Darren

Yes, I use the Mac version everyday. -Darren

yes

That is really great news! With the addition of iXML and real firewire throughput speeds, a 7 series recorder would make a handy addition to any boom recorder/metacorder package. One will now be able to use it for over the shoulder or insert car work and then just drag the sound files into your current metacorder folder, and the files will even show up on the sound report! I must confess, I had felt that without file renaming capabilities, the 744 really wasn't useable as much more than a backup recorder. You could re-name your files on your computer but the metadata would still have been wrong. -Darren

I have two throat mics that I use with my FRS radios for communications between my sister and I when we are travelling on our motorcycles. They work GREAT. Good Signal/noise in loud environments and not susceptible to wind noise. However, not particularly HIFI either. http://www.iasus-concepts.com/nt/nt-modular.htm#

This is NOT possible. You cannot use the 744 as a Master sync clock when you are feeding it digital audio. As soon as the 744 sees a digital input signal it puts itself into slave mode. As a result, if you try to clock everything from the 744 you effectively have NO master clock at all. It sounds real ugly. Best, Darren

I shouldn't expect that there will be any clocking issues with this setup. You will need your Sonosax to be the clock master. The 744 will automatically slave to it. You will need to instruct your MOTU to slave to its SPDIF input. I'm not certain if Boom Recorder has a monitoring section that allows you to send the return from the computer back to tape returns on your console like Metacorder does. If there is a method to monitor the return from the computer, put some tone up and listen carefully for 30 secs or so while watching your meters. Any clocking issues will show up as pops or clicks and also possibly as a slight level instability on your meters. -Darren

I thought that the Sound Devices recorders might be ideal for use as both a WordClock and timecode master. These little recorders are not much bigger than a Denecke GR-1 but have the added advantage that they can also generate WordClock and can quickly be removed from the cart for use as a second recorder for over the shoulder shots, or the insert car if you feel that your laptop is too unwieldy. Unfortunately it turns out that, when using a digital mixer etc, the 7 series recorders cannot be used in this way. They cannot be used as a MasterClock whilst audio is being fed to them via their digital inputs. As soon as a 7 series recorder is fed a digital signal it defaults to slave mode and ceases to generate its own clock. Of course the timecode will still function normally. I view this as a liability, and I would suggest that anyone who uses one of these machines lobby Sound Devices to add a software switch for the user to manually select whether the 7 series will be the master or a slave with regards to WordClock. I have spoken with them on this topic and they do agree with my viewpoint. However, Sound Devices seem to play their cards quite close to their chest with regards to where things are on their priority list or whether they will even implement a given suggestion at all. Best, Darren Brisker

Hi again Noah, Without evidence to support the contrary, I would have to say that you are right in that it is not a proven fact that the machines that we commonly use that do output both wordclock and timecode do indeed have both of these clocks tied together in one way or another. That being said, I'd be both shocked and extremely disappointed in the manufacturers if they are not. This especially holds true for devices such as the new Ambient 501. I'm not certain what the point of this device would be at all if its wordclock and t/c are not locked together. All of that being said, perhaps the next stop for you may be the manufacturers themselves. Best! Darren

Noah, There are probably many factors involved in what makes an A/D "more accurate". For the purposes of this thread let's assume that by more accurate we are only referring to a more accurate clock. I'm not sure that the idea that the timecode generator should be the master clock first and foremost is a completely accurate way of describing this scenario. There are two separate theories at play here. The 1st is that it is not a bad idea to have your timecode clock and your system sync clock tied together for the purposes of Sample accurate timecode. The 2nd is that, yes, you should use your best clock for your system sync. Most probably you will find that it just so happens that your timecode generator IS your most accurate clock. How much difference will this actually make to quality of your sound? Hard to say. I suppose the improvement will be inversely proportional to the quality (ppm/jitter) of the wordclock in your current system. I don't believe that you will find any A/D's with clocks that are more accurate than those that are in timecode generators. As previously mentioned, you will probably not find ANY A/D's with .1ppm TXCO clocks. Some may not consider any of this to be absolutely critical. I suppose one could just consider this the optimal way of doing things. In any case, this may not be possible with all combinations of equipment. As mentioned, it does require either a t/c generator (whether built into a hardware recorder or not) that can also output some sort of clocking signal that the rest of your digital gear can accept. Also in the case of hardware recorders such as the 7 series, Cantar, Fostex, Deva, etc, this is probably al ready occuring internally anyway. These machines are often the only digital device on one's cart and therefore are already being used as both the A/D and the Master T/C generator. -Darren

I am suggesting that you use the "machine" that generates your t/c as your Master Clock with regards to wordclock. If this machine happens to also have a wordclock output, you may use that, Digital equipment can also be clocked via the actual digital audio signal as well. So, if you are feeding your second recorder via AES, SPDIF, or ADAT Optical, it will be able to lock to these as well; it is not necessary to use the actual wordclock BNC connection. In fact, locking to the ADAT or AES signal can often be the preferred method. It all depends on your exact setup with regards to signal flow. If you are feeding both of your hardware (744, cantar, etc) recorders from an analog board and both of your recorders have built in timecode generators, it is probably not necessary at all to lock them together via anything other than timecode as they will probably already have their timecode and wordclocks tied together. You could also lock them via sending wordclock from the unit generating the t/c to the b/u recorder. If you are also using a digital mixer, everything changes as that now also has to have its wordclock synced as well. There is no simple answer as it completely depends on your unique combination of equipment and signal flow. If you tell us what you have, we may be able to offer suggestions as to how to clock. Thus far *my* clocking arrangement has been: GR-1 supplies timecode. Timecode is fed into an analog input of MOTU Traveler. Traveler is told to clock to the incoming t/c via software that's on the Traveler (If you look at the photo of my cart, I believe you will see that the front panel of the Traveler shows that it is using SMPTE as a clock source). 01V96 console is then slaved to the Traveler via clocking signal contained within ADAT optical signal that I use for "Tape" returns from the MOTU going into the 01V96. Any backup recorder, if required, will have its clock slaved via the sync information that is contained within the AES or SPDIF digital audio that will be sent to it from the mixing console or the Traveler. You are correct in your assumption that you can not simply feed SMPTE t/c into a wordclock input. -Darren

This can be done using a 744T, or the new Ambient 501, as both of these devices emit *both* wordclock and t/c. Or any other device that emits both t/c AND wordclock. There are probably other recorders capable of this as well. Or, as Take mentioned, it can be a device such as the MOTU Traveler that is capable of *syncing it's word clock to incoming timecode*. The Traveler would then go onto be used as the Master Word Clock for the rest of the devices in the chain. -Darren

Hi All, I'll revisit this and try to explain it as best I can. First of all, in the digital world, the better your clock (lowest PPM and least jitter), the better your audio quality will be. Poor quality clocking results in reduced fidelity at best and a totally unusable signal at worse. Somewhere in the middle are pops and clicks. Very poor clocking will result in distortion, or a totally unuseable signal. So it is very important to use the best clock available to you as your Master Clock. In the music (read commonly available interfaces) world, most decent clocks spec out at around 50-100ppm. Many are much less than that. Really high end interfaces may spec out at 5-10ppm. As we all know, we have now become fortunate enough that our t/c clocks are now spec'ed as low as 1ppm (Denecke) or 0.1ppm (Ambient). As well, most timecode clocks are TXCO rated whereas Wordclocks generally are not. In summary, our timecode clocks are generally FAR more accurate and temperature stable than most word clock Master Clocks. For this reason, if you have a method of slaving your Master Word Clock to your timecode, you should do it. You will have a much more stable word clock for doing so. The second reason for slaving your word clock to your timecode does indeed have to do with sync. As we know, in the non-linear world linear timecode does not exist. Timecode is derived from a stamp at the beginning of the file and then *samples* are counted from there to extrapolate what the timecode should be at any given point in the timeline of the audio file. So, it stands to reason that, if your timecode clock (0.1ppm?) and your word clock (100ppm?) are not *exactly* in sync because they have drifted apart, the *extrapolated* timecode for a given position in an audio file's timeline and the *actual* timecode for that same position will not be the same. If my math is correct, even a spec of 100ppm could allow for a slip of up to 1 frame every 6 mins. This will cause problems with the editing and especially the re-conforming process. Locking your word clock to your timecode clock will 100% ensure that the extrapolated timecode and the linear timecode (such as what the slates are seeing, etc) will *always* remain locked together. This is referred to as "Sample Accurate Timecode" or "Sample Accurate Sync". I hope that this will help shed a little more light on this subject. -Darren

I forgot to mention... I have an additional Remote for Metacorder as well. It is fairly small, lightweight, cost effective, and extremely versatile. And it will run for months on rechargeable AA batteries. It will allow you roll, stop, play back, change your metadata, arm tracks, and much, much more! It's a bluetooth keyboard! I have two. One is from Apple, I think it was around $60. The other one was around $100 but it is VERY compact. It is a Think Outside "Stowaway". http://www.thinkoutside.com/stowawaybt_product.html Pictures of the Apple one on my cart can be found in the pictures section of this web board. Cheers! Darren

I remote roll Metacorder using MMC (midi machine control) from my Yamaha 01V96 mixer. This is my standard way of rolling/stopping with Metacorder. -Darren

Personally, I would think twice about this. The North American mods might be more acurately reffered to as the Trew Audio Mods, as that is where the idea to implement them came from. I completely disagree with the idea that the channel mutes will NOT mute the direct outs. I absolutely insist that my channel mutes DO mute the direct outs. There is nothing worse than being on one of those poor communication/hectic shows and starting a take and it is not until after the camera rolls that you discover that one of the actors that you were told is in the shot, is actually not in the shot and, worse, saying or doing something embarassing, and you are going to record him pre-fader for the entire take. It is great to know that you can mute the direct outs at will. In my DA-98 days, I would have simply unarmed that actor's track. But remember, for the most part in the non-linear world, you can NOT unarm a track once you have started recording that take. -Darren

I spent an hour fooling around with this Sonosax recorder. User interface was VERY intuitive and very easy to use. Shortcomings of this recorder (at the time that I tested the demo version) were: 1. Monophonic file format only. NO polyphonic option. 2. Can only playback files that are recorded on itself due to propietary system of using a text file generated with each audio file that indexes or tells the machine where to find the files. 3. No timecode outputs. 4. I also seem to recall that I believe that only 4 tracks can be recorded through analog inputs. -Darren