[timhumParticipant]

When I bought my QU16 some years ago I got a SanDisk ultra USB stick and it has worked fine. A job came up where I needed to give an editor the recording and had to source extra memory sticks. The ones I bought a few weeks ago did not work, neither did an Extreme version. Others on the list which I could source easily did not work or recorded errors when recording multitrack. I bought 6 of them altogether so wasted a lot of time and money trying to get the system to work.
The card which worked perfectly (using firmware v1.95) is a CFast2.0 Integral Ultima Pro 32GB(it is a version of a normal CF card which does not fit the normal CF card socket).
It has to be used with a Startech CFASTRWU3 reader/writer. It is a system used for video cameras and has a whopping 520MB/S write speed!
It is not a cheap option but it works!!! and in fairness, a very cheap alternative if compared with a separate multitrack recorder.

[timhumParticipant]

The definition of Zero Level is indeed 775mV into a 600 Ohm Load. The issue is that things have moved on since that specification was introduced and although it does remain a “standard” it used to refer to 8dB below peak level, now it refers to 18dB below peak in order to contain short peaks which would distort in digital systems. When the PPM system was introduced back in the 1930’s, system noise was a big issue and keeping programme material above the hiss was important. Now microphones and audio circuitary are commendably quiet, the issue is more with the bubbling mud type of peak distortion you get with digital systems, hence the “standard” shifting to -18dB from the now historic -8dB. It does not help the situation when some cameras used for broadcast use do not have calibrated meters and editors complaining they have to turn up their speakers.
As regards the QU 16 I have and just looked at, if you ignore the red peak LED, the next one down is +12. If you consider that to be the peak, then your Zero Level calibration tone should be sent at -6db (i.e. 18dB below +12) and helpfully there is a green LED with -6 next to it!!! So with only a decibel or so error you should keep in the green with only a very occasional bottom yellow LED lighting up which is as good as you need to get on the hoof with un-rehearsed live action happening on your OBs.
Thank you for your question, I have been working with the QU16 mainly as a PA mixer so it was good to check how it would integrate into a broadcast standard setup. It also occurs to me that the post production sound guys who say “Yellow is the new Red” know their stuff and have a handy catchphrase to pass on their knowledge!

[timhumParticipant]

Zero Level as defined as a reference level, for example if you are sending zero level tone as a reference is 18 dB below peak level and is what we line up to in TV land, there is sometimes, on professional equipment, an indication at that level on a meter. It corresponds to PPM 4 on an old fashioned analogue meter.
To be pedantic, it should be -18dB in an EBU system and -20dB in a SMPTE one but what is 2 dB between friends!

Using this level gives 10 extra dB of useable level before nasty digital overload distortion kicks in if you compare it to the old analogue PPM meter. The 10 dB safety margin is useful because it makes any very short spike which does not show on the meter, pass safely through undistorted.

Post production may gripe that they have to turn up their speaker volume, iI have had that and on that show I put a limiter in the last 6dB and gave them a hotter mix which they seem happier with.
On other shows where the sound is remixed in the dub from ISO tracks, the same post production bods like to say that “yellow is the new red”. That stops any possibility of overloading and they can add noiseless gain of course. There, the professional standard of peaking at -10 dB with a “Zero Level” reference at -18dB would apply.

I found this on Wikipedia, it says what I say above with references.
Because quasi-peak PPMs indicate neither loudness nor true peaks but something between the two, it is important to allow sufficient headroom when using them in the control of digital audio levels. The EBU convention (R68) provides for this by defining Alignment Level as −18 dBFS.[22] Thus a peak to the Permitted Maximum Level as indicated on a quasi-PPM corresponds to −9 or −10 dBFS. This 9-10 dB margin allows for operator error, the true peak typically being several dB higher than the PPM indication, and that subsequent signal processing (e.g., sample rate conversion) may increase the amplitude.

SMPTE RP 0155 recommends a different alignment level, corresponding to 0 VU, of −20 dBFS.[23] The two conventions result in line-up tone levels that differ by 2 dB, but in practice the level of programme modulation tends to be similar.

The SMPTE and the EBU agree that regardless of whether −18 or −20 dBFS is used as the Alignment Level, that level should be declared and that in both cases programme should peak to a Permitted Maximum Level of −9 dBFS when measured on an IEC 60268-10 quasi-PPM with an integration time of 10 milliseconds.[24]

So your question is a good one which has at least two answers depending on your customer. It might be best to talk to your post sound people to see what they would like or to tell them what they are going to get so everyone knows where they stand. I last sent zero level tone to a camera about 10 years ago, even professional ones don’t always have a calibrated meter on them but if you don’t mod over -10 dB you will keep your job!

[timhumParticipant]

Those two frames difference in 90 minutes would be unacceptable in many workflows. The error might be down to the camera drifting or having a one off glitch which can be caused by changing the camera battery for example. I work on a TV show where the timecode is synchronised between the sound recorder and cameras just twice during the day, first thing and lunchtime, the timecode stays in sync except for those occasional problems which to be honest amount to operator error.

Your idea of using a constant feed of timecode onto a spare channel is fine and what I did with good results. I used the ST1 input but a spare channel at line level should work too, just check for the distinctive timecode crosstalk in adjacent channels.

Contact your editor and ask if you can supply timecode as an audio track, most systems allow this but some editors don’t know they do.

USB, ABS and S/PDIF are just means to an end, as long as the timecode is recorded in the same audio file as the audio, all will be good.
Modern synchronising workflows do not always use timecode, a clapper board is a great reference whatever system may bein use and some editors are perfectly happy with that nearly 100 year old system! Putting an offset into the audio or video when synching up is routine and not an issue. Look up pluraleyes, it is used as an automatic synching system which looks at the audio waveforms on the audio and camera (feed the camera with audio or use its owm microphone).
I did a shoot with 10 cameras recently, Go Pros, DSLRs and all manner of mini cameras, only two had timecode capability and when I offered to sync those cameras with my audio, I got a blank look from the Director/Editor! Idon’t think he knew what it was but runs a production company making top quality corporate videos.

[timhumParticipant]

Thanks for the corection GCumbee,
You are right of course. The centre cardioid goes to both left and right and the fig8 as I described it.
I was amiss in omitting that detail.
I recorded a concert last year with a fig 8 centre mic along with the fig8 “S” mic. The advantage was supposed to be that you could rig the mic a little closer to the orchestra and cheekily do another MS decode (with the centre mic phase reversed and the “S” mic phase reverse going to the Left instead of Right)) to add a controlled amount of hall ambience. Easy enough to do in post production and was quite effective but it only works if you have a nice acoustic in the hall of course.

[timhumParticipant]

I record MS stereo a lot. Not on the QU mixer though. It is a simple matter to do the MS to LR matrixing in the mix down process after the recording session using any of the plugins available but also you can DIY by duplicating the M and S tracks and inverting one S track and calling it the R channel. I monitor on headphones on location and am happy just to make sure the M and S are recorded at a sensible level and not distorting etc.The beauty of the MS technique is that you do all the adjustments in post production, not during the recording.

If I wanted to monitor in LR stereo during the recording with a QU mixerI would use the Aux sends and send the feeds to a headphone amp with a MS matrix or use a cheap 4 line input mixer to turn MS into LR as above.

[timhumParticipant]

I use magnetic tape. There are there venues I use which have a standard set up each which I keep stored in the QU16 and there is a strip for each of them. Very handy.

[timhumParticipant]

As above, just turn up the vocal mic and have enough grunt in your PA to overcome the backline amps. I do the PA for a big band in the function room of a pub. The band is massively too loud for the vocalist who told me all she could hear from her wedge was the band. Only her mic was feeding the wedge so that was the scale of the problem. I work a fraction below the point where the vocal mic rings prior to howling and it is frankly not enough when the band are blowing hard.
Which brings me to another solution, maybe the room you are in is too small for that volume from the guitars. If you can, ask the players to hold back a bit on their volume in line with what you can do for the singer. It is more normal to adjust the mix to fit in with what the drummer sounds like. In several bars in Brighton, groups are told to bring a set of electric drums so the overall volume is right for the venue.

[timhumParticipant]

That’s amazing! Were you there? Spot on, many thanks. I think the routing crept in with a software update and for some reason I had the ST1 fader up with the output on it. I am most grateful, a beer is yours should we ever meet!

[timhumParticipant]

I have been a location sound mixer for TV for 40+ years. A friend introduced me to a Big Band playing once a month in West London and after a while I started operating their PA. The band’s PA was inadequate so I bought the QU16, the first digital desk I have owned, and some active speakers.
Elsewhere on this Forum you may find that I am having a bit of trouble with the room, the loudness of the band (103dB on my noise meter) and a vocalist who struggles to hear herself and the audience only hears if they are sitting near a FOH speaker.

My Son plays in a couple of bands and composes, Wife and Daughter both teach and are active musicians also. I play with the faders and try to produce a musical result.

In spite of the above I enjoy working with musicians and conveying the result to an audience. I appreciate I have much to learn and find that is stimulating. The show must go on – even when someone spills beer down the mixer and horrible noises and smoke emerge from the speakers! A previous mixer thank goodness, I spotted the QU16 when shopping for a direct replacement to the Mackie I was using on the fateful night and could not resist taking it home.

[timhumParticipant]

Thanks for the clarification cornelius78, I think that next time I am in that awful room I will use the PEQ to notch out the primary offending howls and use the GEQ to deal with the rest of the offending frequencies. I think I took out about 6 or 8, rolled off the bass and ever so carefully edged up the level during the performance to establish the max loudness possible. It was nearly enough, pushing the level even further, the room became very lively like I had added some sort of reverb and I backed off from that a tad. My first PA job and I nearly wished I had not volunteered! Perhaps it is a good case for the automatic howl suppression kit.
I was somewhat vague in my previous post about the iPad app. It is called FFT Plot and gives a graphic showing the entire frequency analysis of what is going on in the room including the value of the loudest frequency, ie. the howl but it does not have to be loud enough to offend the ear as you can see the ringing develop before the full blown howl develops.

[timhumParticipant]

The settings depend on the room shape, size, position of the FOH speakers and the level of the vocal mic may depend on the amount of sound absorbing audience attending.
In rehearsal, once your basic mix is established or before if you get to set up before the band arrive, hit the RTA tab on the Home screen and the fade up the vocal mic until you get the your first howl, notice the frequency on the RTA and notch the graphic EQ down a few dB at that frequency, repeat these steps until you achieve audio happiness.
The skill comes in, and here I will have to defer to other more experienced practitioners, where you inevitably find yourself looking at a graphic EQ setting resembling a profile view of the Alps with serious notches in the vocal range. This may have the effect of making your vocalist sound like a duck and that is before they ask you for a “bit more middle”, “more presence” or reverb. All of which will compromise your anti-feedback settings. There is a rather nice and cheap iPad App called FFT Plot which is useful to use when doing this.
You could but a “Feedback Destroyer” but then you are in the hands of an unthinking algorithm which will save your speakers but may have unwanted audible artefacts.
I am interested in following this myself because I have a regular booking in a truly ghastly room with a Big Band too loud for the room and a vocalist who can not hear herself. I have tried squeaking the room with some success but am waiting for the day someone spots what I have done to achieve the nearly OK sound!

[timhumParticipant]

Hi AB7, I use Audacity and there are no issues. I have to load the tracks individually and they all line up. I did a video shoot and mixed down using the desk but it was a long recording and I would not do that again. Far better on the computer in my view.

[timhumParticipant]

Like Bob above, I record 14 tracks of material (and 4 empty tracks) regularly on a SanDisk extreme. The gig starts at 9PM and grinds to a halt at around 11:30 PM. I don’t normally bother to stop recording at the break around halfway through the evening.
The manual for the QU 16 does not include the SanDisk Extreme memory sticks as suitable because they were not around at the time it was written, they work perfectly. The only warning I should give is that there are plenty of fake memory sticks out there so make sure you buy from a reputable dealer. The memory sticks are so much more reliable than the bog standard hard disks which can experience problems with vibration at high acoustic sound levels experienced on stage!
One more word of caution, it is possible to snag the memory stick with your headphone lead, I route mine around the back and bring it up on the left side of the mixer to avoid messing up the recording.

[timhumParticipant]

Although it is a waste of disc space when recording, the unused channels can be disposed of when importing into your computer for mixing down. Even using the QU for remixing, the same memory stick can be used, then used again once the project is over. Memory is relatively cheap.
If you archive all your multi tracks, I agree, it is a waste of memory and a firmware update to record fewer tracks would be useful.
It occurs to me that uprating the firmware a couple of times a year is a very good way to increase the usefulness of a product and keep it at the cutting edge using users suggestions without having to design a new product every few years from the ground up.
There are workarounds for just about any scenario but an elegant software solution is much more attractive. Until the CPU is gasping for breath and reliability is a problem.

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