Jordan Wolf
Forum Replies Created
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Did you advance the show with the event production personnel? That’s something I wish companies would do more because it makes life easier and keeps everyone on the same page.
I operate similar shows to the one you described all the time – it’s what I do for a living. If the client tells me that there’s going to be someone filming, I’ll do my best to follow-up with that person.
I come from a broadcasting background of sorts, so I have certain expectations regarding audio feeds. I know how I want it if I’m going into a venue to shoot something, so I try to give that to whoever comes into where I work.
Here’s what I try to provide:
1. Personally verified hum- and buzz-free mono audio from and independent mix bus (Aux send, matrix out, etc.).
2. XLR connection(s) with the option for Mic or Line level and, possibly, a pad.
3. 1kHz tone referenced to +4dBu or -20dBFS that’s piped through the feed until a few minutes before the event starts.
4. A wye cable for splitting the audio to a safe channel (for screaming, etc.).
Wolf
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Alex,
As long as each interviewee eats the mic and the music isn’t blasting, I don’t think the change in background music will be too distracting. Now, if you have a really quiet talker or choose to shoot the interview right in front of a speaker stack, etc., you’ve got problems…
One way to keep things consistent is to keep yourself stationary and have the people come to you. Also, adding a room mic or two and getting a feed from the audio mixer may help with transitions.
Wolf
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Richard,
From routine use on various consoles, I’ll vouch for needing to crank the gain with the venerable SM58. The PR40, I’ve never mixed with (would like to), but the PR35 and PR22 I own/use don’t require THAT much gain.
Wolf
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Mark,
What interface are you using to go from the A&H mixer to your computer? Is it one of the ZED series boards?
You’ve got some nice gear there, and I’d like to assume that your gain structure is set up correctly, but assumptions are where troubleshooting should begin – so:
Would you mind sharing how you set the gain for each device in your signal chain?
Wolf
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Hey All,
Any device with one or more conversions between the analog and digital domain(s) will have some amount of latency due to the processing involved to convert the voltage from an analog waveform into 1’s and 0’s that the computer can understand and manipulate. There are various ways to reduce latency, but it will always be a part of digital processing. Most people are comfortable with 10ms or less; most interfaces are capable of 3ms or less nowadays.
The main thing for what most of the readers here need to consider is that, unless you are hearing both the original signal AND the affected signal at the same time, you don’t have to worry about the latency.
Another term is “propagation delay”, which more specifically refers to the combined A/D and D/A latency times to arrive at the total delay from input to output. I deal with this in my Sound Reinforcement work when using a DSP to processing loudspeakers or when interfacing digital equipment with analog signal chains. In my line of work, it’s definitely audible sometimes…which is never a good thing.
Wolf
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Absolutely go XLR out of the breakout box. If you need to unbalance it, I recommend using an interfacing box that will properly match the impedance and voltage characteristics involved in going from a professional line level output to a consumer-grade line level input.
You would go XLR into the unit and then go 1/4″ TS (not TRS) to dual RCA or mini jack (1/8″/3.55mm TRS) to get into the Genelec 5040.
Wolf
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Chris,
Take a look at this PDF, Section 3.4. It should help you understand the DialNorm process a bit better as well as show you a visual representation of what Dolby compression actually does.
Happy Reading!
Wolf
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Steve,
If it’s wired as most others are, it should be:
Pin 1 = Shield
Pin 2 = +/Non-inverting (left channel)
Pin3 = -/Inverting (left channel)
Pin 4 = +/Non-inverting (right channel)
Pin 5 = -/Inverting (right channel)HOWEVER: I think it would be best to make/buy a breakout cable similar to this. That will allow you to use either channel, both channels, or even feed them separately for a “normal” channel and what I call a “scream” channel (gain set to not distort during louder-than-usual passages). To make your own, you would just split Pin 1 to both XLR 3-pin connectors and use Pins 2 & 3 on the Left/Channel 1 connector and Pins 4 & 5 on the Right/Channel 2 connector.
Hope that helps!
Wolf
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Hi Chris,
First, a question: when viewing program material (not commercials), is there still a disparity in levels between the two tv sets? If so, then you’re got a few more places to check.
Here are some options to explore:
1.Most set top boxes allow the user to select fixed volume (used when there’s a central receiver unit) or variable volume control (when the TV is already set). Take a look into the settings to see which is selected for each box.
2. All the “modern” televisions I know of allow for some adjustment of the volume and rudimentary EQ adjustment. Make sure these aren’t set to one extreme or the other. Also, realize that sometimes these settings are source-dependent and are independent of each other.
3. Try hooking up the CRT TV’s cable box to the “modern” TV and then vice-versa for that TV’s box (but only test them one at a time). If the issue transfers with the box, you know where the problem is. If it doesn’t, then you know it’s the TV.
One other point to consider is what connections are being used and how the signal gets into the TV, is processed, and gets to the speakers.
A number of things, I know, but I figured I would try to put them all on the table for you. Anyone else think of anything I missed?
Wolf
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“My question is this: Do these devices (recording PCM 16/24bit 48mhz) record sounds higher than human hearing?”
Andy,
According to the Nyquist Theorem, the sample rate should be at least 2x the highest frequency you wish to record. Therefore, anything recorded at a 48kHz sample rate should have a high frequency cutoff around 24kHz…how true this is in reality, I couldn’t tell you because I haven’t measured it.
Your idea is used all the time by radio-frequency microphones/in-ear monitor units as well as to keep TC. It’s called a carrier or pilot tone (19kHz is fairly common).
See here for an explanation from Wikipedia – from my knowledge, it’s a pretty good overview.
Wolf
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