Sync question for you.

[A. J. deLange]

Originally Posted by Dave Anderson

Just as 2 clocks are never the same time, off by fractions of a second, 2 time-clocks in recording devices are never the same.

The fact that a pair of clocks read a fraction of a second apart at any particular time is half the story i.e. the offset. The other half is that the rates are slightly different. In the time one ticks off 100 seconds the other may tick off 101 (or 99 or some other number). It is the role of wordclock, blackburst or video sync to make two entities' (cameras', recorders') clocks run at the same rate. It is the role of time code to mark recordings at the same instant of time with the same "address" (time stamp) though many pieces of equipment are capable of using time code for the rate as well. There are 80 bits of timecode information in each of 30 time code frames (not necessarily the same as video frames) per second for 2400 bps which when multiplied by 20 gives 48,000 (or by 40 gives 96,000) sps wordclock for audio. When the frame rate is 30 that's 1600 audio samples per frame. When it's 29.97... its 1601.6 and it gets tricker. And so on for other frame rates.

Having things run at the same rate is more important than time stamping because you can usually figure out the offset in post but time code is handy. The trick is to make sure you have synchronized blackburst for the camera(s) and wordclock for the recorders. Both are available from the aforementioned Lockit and Clockit boxes. Synchronizers (e.g. MOTU Time Piece) can derive word clock from time code as can some recorders so a camera which puts out LTC (Linear Time Code), which many do over their LANC ports, can often be sync'ed to a recorder. The combinations of ways in which this can be done are numerous.

Always keep in mind that a LTC signal is an audio signal which, if recorded on a spare track, will ultimately always be available to tell you what time it is (so long as you have a way to read it).

[Steve House]

Originally Posted by Alastair Brown

My mini-desc is the same. Never out, never a problem. So whats difference? They are still recording zeros and ones?

Link sorted.

The difference is due to the way the recording is tranferred into the NLE with a minidisk versus a file-based recorder. While they are both recording ones and zeros, the file-based recorder records as a bwf (wav) file which is simply copied over onto the editing computer -it's a straight file copy. The minidisc, otoh, effectively "plays" the recording to transfer it into the editing workstation, sort of like hooking the digital output of one device into the digital input of another. It's still a digital transfer but it's not a just a direct file copy. Why should this matter? Consider two recorders, one file-based and the other a minidisc. For simplicity lets say they are both recording at 48kHz samples rate and both of them have sample clocks that run 10% too fast. They're recording a sound that lasts exactly 1 second. In each case the recording is supposed to contain 48000 samples but because the clock is ticking too fast, it actually has 52800 samples (48000 plus 4800). With the file based recorder, we copy its file into an editing computer whose clock is running exactly on-spec at 4800kHz and play it back. After 1 second of playback time it will have played 48000 samples. But our file still has another 4800 samples to go, which will take the workstation another 0.1 second to get through. In other words, the editing computer will take 1.1 seconds to playback the number of samples that were originally recorded in 1 second. The sound is running slow. But consider the same scenario with the minidisc. Even though its clock is also ticking 10% too fast and it records 52800 samples in one second instead of 48000, when it is being transferred into the computer it is effectively being "played" on the recorder itself, the playback rate governed not by the workstations more accurate clock but instead by the recorder's 10% fast clock. The result is that the 52800 samples play in 1 second, NOT 1.1 second, and the event time isn't distorted.

It's a shame the whole digital rights managment broohaha led Sony to effectively cripple the minidisc technology. It had some distinct advantages that gave it a lot of promise, including the aformentioned behavior. But I'm afraid it's now a technology that's as anachronistic as DAT or analogue audio tape, or 2 inch quad, 1 inch helical, or VHS and Beta videotapes.

[Bill Davis]

I think this is getting WAY too complicated to solve his actual problem.

You DO NOT need extra hardware to solve this.

Look the problem is fundamental.
The standard for DV (plain old digital video) is a 25Mpbs video stream and two 48Khz audio streams.
If you bring in something other than that, it needs to be re-calculated. That's all.

Fixing this is largely trivial.

Just export your audio to a file and RESAMPLE that file into 48khz audio. Period. End of problem.

I work with Quicktime and Final Cut Pro - not Vegas - but back in the early versions of FCP before they coded in "on the fly" audio resampling - I had to do this all the time.

We'd get CD audio that was 44.1khz and if it simply got stuck on the timeline - the audio would drift.

The solution was always to simply export the audio from the timeline as an AIFF File, Open it in Quicktime Pro - change the sample rate to 48Khz And replace the original timeline audio with the newly sample file.

Presto - NO MORE DRIFT.

I'm confident you have equally simple and easy to operate tools in the PC/Vegas world.

This is just MATH - nothing crazy - and we all know that computers are REALLY good at math.

Just re-sample the audio and get on with building your video.

Good luck.

[A. J. deLange]

Originally Posted by Bill Davis

Just export your audio to a file and RESAMPLE that file into 48khz audio. Period. End of problem.

This will work if and only if the other (recorder) clock is synchronous (in terms of rate) with the camera clock. Interpolation to 48kHz from 44.1kHz only means that the resampler computes 480 evenly spaced output samples from every 441 input samples. Were the recorder clock off 1% its sample rate would be 44.541 kHz and the resampled file would be sampled at an effective 48.480 kHz which would drift relative to camera sound and video. So if you are lucky (as some have said they are) or you can tolerate small amounts of relative drift this method is OK. Otherwise, you must do something additional. Solutions in software are possible if you have in and out sync points on recording and video. Two clapper strokes 10 sec apart would be 445,410 samples apart on the recorder with 1% clock error. In many NLEs and DAWs one can put cursors on such audio to stretch or shrink it to match the video in which case the software would interpolate 480,000 samples from the 445,410 input samples (arbitrary sample rate conversion is possible - it's just math though not trivial math) thus effectively locking recorder and camera clocks.

Were the quote true there would be no market for synchronizers, black burst generators, cameras with "Jack Packs", Lockits, Clockits etc. This gear is expensive, sometimes tricky to master and makes for more complicated setups so if synced recording is an occasional requirement and if approximate sync is sufficient by all means use resampling (with stretching if necessary) and manual time alignment. For professional work electronic syncing is a must.

[Bill Davis]

AJ,

Don't mean to be a pest about this but I haven't seen a temporal offset DUE TO SAMPLE RATE problems that was as hard to solve as you allege since engineers went to crystal controlled clocks a LONG time ago.

(The closest I remember "in the modern age" is back in the late 90's when Canon released a camcorder (the ORIGINAL XL-1?) that had a slightly wonky sample rate. So the guys at Apple had to put a check box into Final Cut Pro 1.5 for guys who were shooting that particular camcorder. Even THEN checking the box let the Mac re-calculate and - POOF - no drift - everything was fine.)

The kind of delay that the tools you mention in your post are designed to deal with is TRANSMISSION delay. Like sending audio or video through a few hundred feet of co-ax plus a few routers and switches for good measure in a studio facility. Yeah, those kind of timing delays may ABSOLUTELY need hardware to address.

But blackburst and genlock were developed to keep SEPARATE machines synced in REAL TIME. The OP is NOT dealing with separate machines. Just reading the same file on two separate devices.

And I haven't seen anyone design a camcorder circuit that drifts as badly as you're talking about in DECADES. Plus it's writing it's data to a digital stream right AT THE ENCODER.

Then taking those ones and zeros and digitally transferring them (with checksum and "keep the numbers accurate" math going on) into yet another system (likely IEEE1394 buss fed or similar) where there is AGAIN essentially NO opportunity for the kind of transmission delay you note.

What the OP is facing is NOT complicated. It's just MATH. Pretty simple math for any reasonable computer processor at that.

I'm saying this is a solution because I've been DOING it successfully for a decade now.

In fact, it's a stupid debate.

OP - just TRY what I suggested. Take the audio file and re-sample it right in your computer. At the MOST, you'll need a free downloadable audio program that can re-sample WAV or AIFF files. So my suggestion means you don't have to go drag down the 100 pound Allied catalog or spend the weekend on-line ordering stuff then waiting for UPS. It costs you NOTHING.

Just try it.

And I bet you report back that everything on your timeline comes out JUST FINE.

Sorry guys, but I got to tell you that there's a HUGE tendency in this business to try and "over think" solutions. (not saying that there aren't things that need serious effort and expensive gear to solve, just that reading a DV tape into a computer and keeping the audio and video in sync isn't one of them) Not after twenty years of constant NLE refinement.

If your signals are drifting that badly, you're doing something WRONG. Perhaps simply recording at a rate you're timeline isn't properly set up to handle. No need for anguish or expense. Just tell the computer to FIX it and move on.

Good luck.

[Ty Ford]

Yes, but what's interesting is that the humble consumer minidisc works, but the H4 doesn't.

Regards,

Ty Ford

[Michael Liebergot]

Originally Posted by Ty Ford

Yes, but what's interesting is that the humble consumer minidisc works, but the H4 doesn't.

Regards,

Ty Ford

That's because the Zoom recorders use reeeeeeeeal cheap crystals from China.
I believe Sony, Marantz, Edirol use crystals from Japan.

Funny it used to be a running joke about things being made in Japan.
Now China is the joke, and Japan means quality.

Go figure.

[A. J. deLange]

Whether the simple technique Bill advocates works or not depends on 3 things: the luck of the draw, the practices of the manufacturers of both pieces of equipment and what your tolerance for relative drift is. A decent crystal oscillator is typically specced to about plus or minus 25 ppm. If you are lucky both pieces of equipment will have the same error or nearly the same error. If you are unlucky one piece may be off plus 25 and the other off -25 for a difference of 50. Thats 50 seconds time base error in a million or 1 frame every 11 minutes. Even so for some applications that's acceptable while for others it is not. It is not, of course, likely that one crystal would be at the maximum allowable drift and the other at the other. More likely one would be at say plus 12 and the other at say plus 17 for a difference (and it is definitely difference in running rates of oscilators that we are speaking of and not cable or switcher delay) for something like 1/5 the error in the max case which would amount to 1 frame in 111 minutes. Better, but in some applications still intolerable.

To the crystal's basic tolerance we must add temperature instability, voltage instability and ageing which taken together simply mean that the drift between units can be greater than the example numbers I've thrown out. All of these, including tolerance, can be compensated for - voltage instability by good voltage regulation, temperature stability by ovenizing, basic tolerance by careful selection and ageing by recalibration. The Lockit boxes solve the sync problem using high quality crystal oscilators. Ambient gives careful consideration in their design to all these factors. The manufacturer of a consumer camera (which costs less than a Lockit) does not nor, in general, do the manufacturers of prosumer or even pro cameras to the same extent. If they did, Ambient would not have a market. Even with the Lockit approach (in which each crystal is locked to a master at the beginning of the shoot/day/scene) there will be some drift though it is hundredths of a frame (10's of audio samples at 48K per day). In some cases this is not tolerable and in those hard wiring or radio linked sync is required.

I hope that this helps to make it clear why some people can get by without sync and others can't. One thing that has not been mentioned which may help when running without is to try to have all pieces of gear at the same temperature.

Again I emphasize that I am not talking about transmission delays which result in clock phase error but rather clock rate (phase rate) error. Transmission delay must be accounted for if all signals resulting from an event are to arrive at a given point at the same epoch. Time code is there to help with this but cameras, for example, often have controls which allow pre or post triggering adjustment. With sound it is not the cables or radio links (which contribute delays in the 10's to hundreds of nanoseconds) but propagation of soundwaves in the air. A camera 30 feet away from a podium will record sound 30 ms (1440 samples at 48K) later than the speaker's mic. But it is a fixed offset. Certain that camera and speaker's recorder clocks are running at the same RATE (seconds per second - the clocks can be different) one can confidently make the PHASE adjustment once and rely on it even if the guy goes on for an hour or more.

Since people are throwing out personal experiences I'll give mine. Two XL series cameras drift enough that the differences are plain to the casual observer after about 15 minutes. Now these are my 2 cameras and your 2 may be fine for an hour (or for 5 minutes).

[Dave Anderson]

Wow, this thread went way beyond me. But from what I've read, I have a question. Why can't we just hook our recorders to the line-in of our computers, play the file back on the original recorder and capture it on the pc just like we capture the video from our cams? Wouldn't this make the sampling problem not a problem?

If I'm understanding what I've read, then capturing everything from all the different devices to the same device should result in everything being in sync. Yes it would take more time to capture the audio in real time, but it would be worth it for me to get great audio.

[Steve House]

Originally Posted by Dave Anderson

Wow, this thread went way beyond me. But from what I've read, I have a question. Why can't we just hook our recorders to the line-in of our computers, play the file back on the original recorder and capture it on the pc just like we capture the video from our cams? Wouldn't this make the sampling problem not a problem?

If I'm understanding what I've read, then capturing everything from all the different devices to the same device should result in everything being in sync. Yes it would take more time to capture the audio in real time, but it would be worth it for me to get great audio.

You could do that and probably reduce the drift but you wouold be going between the analog and digital domains with the subsequent risk of generational losses. Like copying an old VHS tape to another tape or an audio cassette to another audio cassette. Probably could get away with it for a few generations with pro grear but quality loss would be a risk

[Dave Anderson]

Originally Posted by Steve House

You could do that and probably reduce the drift but you wouold be going going between the analog and digital domains with the subsequent risk of generational losses. Like copying an old VHS tape to another tape or an audio cassette to another audio cassette.

Ah, didn't think about that. Thanks.

[A. J. deLange]

Originally Posted by Dave Anderson

Why can't we just hook our recorders to the line-in of our computers, play the file back on the original recorder and capture it on the pc just like we capture the video from our cams? Wouldn't this make the sampling problem not a problem?

When you do what you propose the player produces signal at every instance of time at the correct rate (assuming the playback and record clocks are the same which they should be for the most part except for temperature and voltage induced drift) and the "recorder" samples this continuous waveform at the correct times. This is essentially the same as what happens when you import a file from a recorder and tell your DAW/NLE the start and end points of the file (or some part of it) relative to reference start and end points (slate or other short duration sounds)in the camera audio or video. The software computes the values (usually between recorder samples) at the times it needs samples to be "locked" to the video. There is no dual (D/A plus A/D) degradation but the filters which do the interpolation must be properly designed to maintain NPR (noise power ratio - a type of signal to noise ratio).

[Jeff Kellam]

Originally Posted by Why can't we just hook our recorders to the line-in of our computers, play the file back on the original recorder and capture it on the pc just like we capture the video from our cams? Wouldn't this make the sampling problem not a problem? Dave Anderson

Ah, didn't think about that. Thanks.

Most recorders and soundcards have optical (digital) inputs/outputs nowdays

[Richard Gooderick]

Originally Posted by Michael Liebergot

That's because the Zoom recorders use reeeeeeeeal cheap crystals from China.
I believe Sony, Marantz, Edirol use crystals from Japan.

Funny it used to be a running joke about things being made in Japan.
Now China is the joke, and Japan means quality.

Go figure.

My Fostex FR2 LE is made in China and it works fine thank you.

[Steve House]

Originally Posted by A. J. deLange

.... There is no dual (D/A plus A/D) degradation but the filters which do the interpolation must be properly designed to maintain NPR (noise power ratio - a type of signal to noise ratio).

True, but while the signal is in the analog domain we have to be careful how we handle it or else we run the risk of degrading it. Being "careful" with analog signal handling is why a Nagra IV was slightly more expensive that a consumer cassette recorder of similar vintage <grin>.