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Using two different AES3 sources without word clock sync


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What kind of audible artifacts, if any, might appear if I wanted to record three separate ISO tracks on a recorder like Pix 240 using two different AES3 input pairs that share the same sampling depth and approximately same sampling frequency (48Khz/24bit), but are not synced to the same word clock?

As a practical example, let's say there was a boom mic connected to input 1 via a MixPre-D and a Sony dual channel DWX series digital wireless microphone receiver connected to input 2. The Sony receiver has a word clock input, but MixPre-D doesn't have one. Let's say there are no word clock sync cables connected between the devices at all.

It is difficult for me to understand which one of these devices will act as a "master clock" in such a set-up and how in practice will the recording device compensate for the slight drift in the sampling frequency between the two inputs. Will there be audible "jitter" on one of the tracks or is it more of a theoretical problem?

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From the SD website re: the Pix 240i (not sure if this is different from the Pix 240)

Digital Audio

· Sampling Rate / Bit Depth: 48 kHz, 24-bit

· AES/EBU: 4 channels in via XLR, 110 ohm, 2 V p-p.

· Accepts 32k, 44.1k, 48k, 96k, 192k sample rates with SRC at input.

· HDMI: 2 channels embedded HDMI input, 8 channels embedded HDMI output

· Accepts 32k, 44.1k, 48k, 96k, 192k sample rates.

· SDI: 8 channels embedded SDI input, 8 channels embedded SDI output, 48k sample rate

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Yes, Pix 240 does have SRC inputs. It was a bit unclear to me how they operate if the input sample frequencies are nominally the same as the recorded sample frequency. Is it correct to assume that SRC inputs continuously digitally interpolate the incoming bitstreams to sync the signal to the internal word clock of the recording device.. even when there is a word clock embedded in the AES3 input signal? Is that what is meant by "digital resampling"?

EDIT: I guess what I originally wanted to ask is: Are AES3 inputs with sample rate conversion aurally transparent when using multiple asynchronous 48Khz/24bit AES3 sources on the PIX240? Are the SRC interpolation artefacts audible with typical location sound material such as dialog?

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  • 2 weeks later...

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 how SD 664 works, but I'm not sure if that is the case with PIX. In 664 the analog limiters are usable also on AES inputs and the signal is indeed converted to analog and back to digital before the DSP.

I'm under the impression that the signal topology in PIX has more in common with the 7-series recorders than the SD field mixers though. There is a possibility that it is all digital when using AES inputs.

Well, it doesn't really matter to me as long as there are no practical implications in the recorded sound quality. I guess SRC inputs "just work" and I shouldn't worry too much about the whole thing.

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In 664 the analog limiters are usable also on AES inputs and the signal is indeed converted to analog and back to digital before the DSP.

I don't see why a limiter is needed with aes. If your aes is overloading the limiter will have no effect on the receiving end of aes and it means that te AD converter before is overloading.

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I don't see why a limiter is needed with aes. If your aes is overloading the limiter will have no effect on the receiving end of aes and it means that te AD converter before is overloading.

Yes, I agree. My point was that since the limiter in 664 is in the analog domain and it is available on the signal path, it means that there has to be a D/A + A/D round trip before the ISO track is recorded.

On the other hand, the limiters on PIX are available only on the analog inputs, but not in the AES inputs. That indicates that the topology of an AES input on the PIX series is perhaps all digital.

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