rofin

It's a combination of the very high meter input impedance and a miniscule capacitor leakage curent. As soon as you load either or both legs to pin 1 it will act as you expect. As a by the by, you don't need non polarised caps. Polarised with the + towards the phantom is fine. Rob

I was for 30 years, but moved on last Christmas. Rob Finch

Hi Gandy, have a look at these links, they should help. http://www.sqn.co.uk/old/e4s/4Sebook.html http://www.sqn.co.uk/old/e4s/4Sespec.html The IVe was available with a range of peak meters and also vu. The output clipping point is approx. +22dBu I think that the post fades were calibrated such that an input level achieving 0dBu balanced out with the fader at 12 o'clock gave -10dBu unbalanced out at the post fades. Rob

The SQN5 has 5 iso outs (selectable as either pre or post fader), unbalanced L/R outs which can be mono'd plus 2 stereo balanced outs each switchable to mic or line level. Additionally, inputs 3/4/5 can be removed from the mix but remain at the iso out. Monitoring ... 2 x switch selectable auxilliary stereo pairs (individual level control) and internal stereo. Rob

I think you're right Eric. It's too close not to have come from the same stable ... Cameron-Pace http://www.cameronpace.com/v2/index.php/whatwedo/equipment/243

<url>http://www.productionrecording.com/Tools/AlaCarte.html</url> various accessory prices

Matt, I may have misunderstood you but the 3:1 rule isn't applicable to mics on the same source. It's specific purpose is to limit the pickup of say mic B of source A and mic A of source B, and works on the principle that a mic 3x the distance from a source will experience approx. 10dB lower sound pressure than the closer one. This in turn reduces inevitable comb-filtering by limiting the impact of one signal on the other. Basically it is an amplitude tweak rather than time based. So basically, any two mics on the same source can be at any separation as long as their relative signal levels are at least 10dB apart when ultimately mixed. Phasey and time-delay issues are another bag and for all practical purposes are as difficult to manage as herding cats. As an engineer it pains me to agree with the recording bods, but as they say, 'if it sounds good, it is good'. Rob

It's parallel. (If like numbered xlr pins are linked)

Hi Jeff, I've been experiencing this for the last 3 or 4 days as well. Not a single other instance, only jwsound, and maybe 8 out of 10 attempts result in a redirection. Usually displays something like ad.leadbbolt, then googleanalytics and ultimately something about prettyunicorn.net Rob

Not all! SQN mixers have a greater input range than Nomad, even with NeverClip. So when NeverClip allows the input to 'exceed 0dBFS' at the input, how do you prevent any of the outputs from clipping if not by reducing the trim, the fader or applying one of the multiple 'needless' limiters? SQNs have no input limiters. SQNs have no input or output transformers. I bet. Rob Finch SQN

Hi Matt, if you set the channel input to Line level, this will place a 100 or 200 ohm resistor (dependent on model) across the mic amp input and will simulate a mic plugged in but without the additional mic generated noise. Rob

It looks like ... Hirose SR30-10PF-6P(71) Plug Hirose SR30-10R-6S(71) Socket Rob

The manual doesn't say that anyway. "Inputs 2 through 5 can only be routed to X1 and X2 pre-fade, while Inputs 1 and 6 can be routed to X1 and X2 pre- or postfade."

From the website ... CPAntenna Uses Circular Polarity for Strong Reception The IWxCPA antenna is not like your traditional paddle or rod antenna commonly used with wireless microphone receivers and pro-audio antenna distribution systems. The antenna incorporates directional, Circular Polarization (CP) technology for both receiving and transmitting pro-audio wireless applications. We guarantee* that this antenna will reduce interference, reduce drop outs, help eliminate ‘swishing’ noise artifacts, improve RF signal to noise, and enhance reception of signals propagated through and around objects. http://www.kaltmancreationsllc.com/rf-test-equipment-html/invisiblewaves-html/cpa-antenna-html/

I'm pretty sure (but don't take as gospel) that the PIX uses sample rate converters at the AES ins and effectively resamples and records at 48KHz/24 bit. So the PIX is the master.

This is pretty much how I see it as well. For bag setups. My concern is that, what are after all portable sound kits, are becoming increasingly specious by design at the expense of core function ergonomics and are basically unnecessarily expensive and power hungry due to the inclusion of rarely if ever used I/Os and complex processors performing dubious at best dsp routines to feed them. e.g. 10 inputs in a bag? really? On a multitrack recorder what is the purpose of having ISO output connectors? On a 6 channel mixer/recorder, do all 6 inputs really need phantom power and low noise pres or are at least 2 or 3 likely to be radio ins? Digital Outs? Vincent, I presume you recorded 4 to the PIX240 because the SD552 couldn't, and it follows that you had to go in digital because the PIX240 can't input 4 analogue. But would you have recorded to the PIX anyway if you had a multitrack recorder at your disposal? Digital ins/ Digital mics? ... Common? To implement a proper (to spec) AES42 input (let alone 2) is a big ask for battery powered gear. 10V/250mA per channel! etc.

Same IE prob here. firefox ok.

For the EFP scenario, is it ever likely that all 6ch are wired or are at least 2 always radio? So what about digital ins and outs. Does anybody actually use these? Direct outs (pre/post) on a multi-track recorder. What's that for?

A long debated topic, but just to get a current perspective! So how many inputs does a recordist 'really' need in a bag, and of what type? e.g. what proportion of wired to wireless? And what about digital? And what about monitor? What about outputs? Iso, camera, bus, phones, digital, analogue? The saying 'No one size fits all' may actually not be true. The penalties of providing potentially unnecessay full spec input and output channels such as XLR panel 'real estate' and associated electronics / power consumption / menu options etc impact heavily on the overall design of a product which would surely be better served if tailored properly to the 'modern' workflow.

Only if their radii from the sound source is different.

Dave, based on that the mic works when AA powered but not from either of two machines and that the crackling is 'worsened when the mic cable is moved even slightly', I'm suspicious of the screen connections in the cable. Being phantom powered, the screen is the return for the current. Try more localised, gentle tweaking of the cable to see if you can narrow it down to a particular spot or connector. I understand that it is a rental kit so you might not want to 'disassemble' it, but if poss, unscrew the xlrs and have a look for broken screen or indeed frayed screen with strands touching where they shouldn't. Rob

This is rather misleading if I may say. The main advantage to gain reduction in an amplifier vs input level reduction is that the amplifier itself has an inescapable input noise associated with it whether there is any signal present or not. A 'mic' level signal amplified to 'line' level by this amp will swamp the noise at the output and hopefully result in a good signal to noise ratio (SNR). As Glenn said, digital attenuation, by which I presume he means feeding a lower digital value to the DAC and into the amp, will indeed worsen the SNR as the output noise is still maximum. Active output drivers, which again I presume means actively reducing the gain of the amplifier is indeed a much better approach as it also reduces the output noise level and therefore maintains the same SNR, as long as the system base noise isn't approached. Output resistive attenuation on the other hand is a perfectly acceptable alternative because as well as attenuating the signal, the noise is also attenuated by exactly the same degree and hence the SNR remains constant. With the same system base noise caveat. As with most things though, there are ways to skew (screw up) these techniques. e.g. If the output amplifier has a worsening noise performance at reducing gain, or if the resistive attenuator uses high value resistors for the centre element (thermal noise).

I'm not sure what you mean. If you took the above setup and fed the 0dBu mixer out and calibrated it at -20dBFS in to a digital recorder, then you could increase the input to the mixer by 42dB i.e. +22dB above what would have equalled 0dBFS without clipping the front end of the mixer, and using the fader to prevent the input of the digital recorder clipping. No fruit salad. Anyway, I don't want to hijack this thread with comparisons, it was just an observation in response to J1mbo's observation. Rob

Just tried one on the bench. With the input gain set to '20', fader set mid way, and an input level such that the mixer output is at 0dBu (-20dBFS), you can increase the input level by 42dB (+22dBFS) whilst reducing the fader to 9 o'clock to prevent the mixer output exceeding +20dBu (0dBFS). The mixer's output won't actually clip until +26dBu. So, with the SQN input handling capacity apparently similar (can't be sure until Glenn reveals all ), it possibly is the headphone feature or the limiter settings. Rob