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Author Topic: DPA d:VICE just announced  (Read 48903 times)

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Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #375 on: April 23, 2019, 05:49:44 PM »
Cool can you calc out the 0.5% levels for us? or show us how to? Im not sure how to do this, and it would be good to have apples to apples numbers.
« Last Edit: April 24, 2019, 10:40:23 AM by jerryfreak »
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Offline aaronji

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Re: DPA d:VICE just announced
« Reply #376 on: April 24, 2019, 11:46:11 AM »
It’s linear, so if you double (half) one parameter, the other also doubles (halves). There is a page on the Sengpiel site about this. I’ll try to find it when I am in front of the computer again.

< 1% @ 123 dB SPL ~ < 0.5% @ 117 dB SPL. Still hard to compare because of that “less than” sign...

Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #377 on: April 24, 2019, 11:49:49 AM »
i guess we'll have to treat "less than" as "equal" in this case between the various mfs, its all we got :)

i wasnt aware that distortion was linear

DPA's own promo stuff doesnt show distortion as linear under 1% (even though its a hypothetical graph for demonstration purposes. might as well call it a cartoon as either axis could be linear or logarithmic, hard to tell)



« Last Edit: April 24, 2019, 11:52:48 AM by jerryfreak »
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Offline aaronji

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Re: DPA d:VICE just announced
« Reply #378 on: April 24, 2019, 12:15:05 PM »
Well, I could be mistaken. It wouldn’t be the first time! I’ll try to find the reference...

P.S. Time for a new thread.

[EDIT TO ADD:]

See this post on the Microphone University section of DPA’s site.

Quote from: DPA Microphone University
In general, the distortion of a circular diaphragm will double with a 6 dB increase of the input level, so you can calculate other levels of THD by using this factor.

With the wrinkle that the 406x mics don’t have circular diaphragms...
« Last Edit: April 24, 2019, 12:24:29 PM by aaronji »

Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #379 on: April 25, 2019, 04:20:28 PM »
interesting

they also spec "At DPA, we specify the SPL for 1% THD, measured at 1 kHz. This figure is important to know as it forms the basis for calculating the dynamic range of a microphone. Dynamic range is expressed as the difference between the level at which a THD of 1% occur (at 1 kHz), and the noise floor (self-noise of the microphone, RMS A-weighted).

again the massive low-frequency energy that the 406xs see due to their response will probably make them distort before a directional mic of same sensitivity spec at 1KHz. note that the larger omnis like 4006 and 4007 are 20 db higher in max SPL

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Offline Gutbucket

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Re: DPA d:VICE just announced
« Reply #380 on: April 25, 2019, 06:07:23 PM »
^ Interesting point!  Might explain the 4060 distorting thing when SPLs don't seem to be high enough to reach the 1% distortion spec threshold.

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Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #381 on: April 25, 2019, 06:19:13 PM »
looks like measurement at 1kHz is according to IEC standard

https://schoeps.de/en/knowledge/knowledge-base/technische-grundlagen/maximum-sound-pressure-level.html

the same standard probably suggests that the level of measured distortion must be stated (as xxx dB-SPL at 0.5% or at 1%, etc)
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Offline detroit lightning

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Re: DPA d:VICE just announced
« Reply #382 on: May 05, 2019, 02:07:53 PM »
using my dvice again after a while waiting for my new 6061s, and just wanted to confirm app settings. I'm using the shure motiv app to record - am I using the DPA app for any of the settings, or is gain overwritten by whatever recording app you use?

Really wish DPA would just have their app record as well...

oh well - the shure app seems to be working well.

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Re: DPA d:VICE just announced
« Reply #383 on: May 05, 2019, 04:19:16 PM »
app is inconsistent for me. they say it does a bunch of things like prevent input overload, etc. with level-adjusting apps like voicerecorderpro and metarecorder it seems to be overridden functionally

i open it up to make sure it sees both mics and the HPF is off, levels are ganged approriately (stereo vs dual) then close it and use apogee metarecorder to adjust levels.

its kinda  frustrating that support cant explain exactly how it works, if there is a unity gain, and the best settings to set levels. they must have farmed out both hardware and software on this one.

it took a week or so to get an answer from TPTB on whether the d:vice is synchronous, adaptive, or asynchronous. I got this answer and told them to ask again just in case there was a typo or something lost in translation. was hoping the class II operation was asynchronous or at least adaptive. i have heard a click now and again in some recordings but hard to tell if that was just some room noise or something else, not necessarily clock adjusting

The answer from our tech team is;
By connection to PC USB Audio class I is used, in synchronous mode
By connection to Iphone USB Audio class II is used, in synchronous mode


a little background on USB audio from Cambridge

Synchronous USB DAC is the lowest quality of the three often used in low end products, so none of ours course! (that’s right, we’re being smug!) These accept the packets of data whenever the data sends them causing glitches every few seconds due to the differences between the two clocks.

Adaptive is where the DAC constantly adjusts its clock so that it can accept the data being sent from the computer whenever it sends them. The constant adapting of the DAC’s clock means that there is no continuous, accurate master clock in the DAC, which causes jitter in the audio stream.

In both of the above the computer dictates the timing of the data packets being sent.

Asynchronous – this is the most complex to implement but it is a huge improvement on the other types. This is because it requests the data packets to be sent in time with its own clock’s timing, thus providing the lowest jitter and sounding by far the best.


Is jitter in this setup something to be concerned about? hard to say. The effect of jitter seems to be a function of your playback setup as well.

From John Siau (Benchmark), answering in response to feeding DAC1 class 1 usb and also a discussion of playback jitter reduction (emphasis mine):

Up until Windows 7, asynchronous USB interfaces offered no advantages other than a possible reduction in jitter for DACs that lacked proper jitter attenuation.  The Benchmark DACs have excellent jitter attenuation, and the asynchronous interface offered absolutely no improvement in jitter performance.  The jitter rejection is nearly perfect in both cases.

The V-link will work well if it passes 24-bit audio, and supports the sample rates you are playing.  Jitter is never a concern with any Benchmark converter and this means that the v-link's relatively poor-quality synthesized clock will not be an issue.


Doug Oade (again emphasis mine):

The fact jitter will be at unacceptable levels simply demands you use a DAC with an async USB input. That will reduce jitter to the DAC clock's jitter. S/PDIF, which is supported by the controller, has a more stable clock easing the requirements of the DAC. But even a very stable S/PDIF clock in a portable ADC can be improved by reclocking. I use an Apogee Big Ben for that, when needed, in my studio but a high end DAC with an async USB input outperforms the Big Ben. All high end DAC manufacturers now offer async USB inputs with femtosecond accurate clocks. The difference is easy to hear.


i have a tenuous understanding of the effects of jitter, but if you have jitter in the input portion of the recording, how is this clock jitter not a permanent artifact of the recording. in other words, if the data is written 'wrong', how can it be 'read right' later?

and if sound is so dependent on the playback device (which most people in the world are listening on low end equipment, phones, car stereos, computers with dubious interfaces, integrated devices, etc), would it be wise to "clean your clock" by feeding it through a high-end dac>balanced interconnects>high-end low-jitter ADC like an AD2K etc.
« Last Edit: May 05, 2019, 04:35:17 PM by jerryfreak »
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Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #384 on: May 05, 2019, 10:54:58 PM »
reading back a bit and some people mentioned fear of 4060s overloading d:vie in loud situations. if youre in this situation run the HPF in d:vice app it will take that low end energy out and drop your peak levels by several dB

ive run my 4061s at absolute minimum levels (with no HPF) to good results. mic noise is greater than d:vice S/N so no harm in normalizing later. 4060s id be running at lowest levels i can in all cases

i have a pair of microdot legacy (non-core) 4061s ill be experimenting with now and again
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Offline detroit lightning

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Re: DPA d:VICE just announced
« Reply #385 on: May 05, 2019, 11:18:25 PM »
reading back a bit and some people mentioned fear of 4060s overloading d:vie in loud situations. if youre in this situation run the HPF in d:vice app it will take that low end energy out and drop your peak levels by several dB

ive run my 4061s at absolute minimum levels (with no HPF) to good results. mic noise is greater than d:vice S/N so no harm in normalizing later. 4060s id be running at lowest levels i can in all cases

i have a pair of microdot legacy (non-core) 4061s ill be experimenting with now and again

I think the monitoring in these apps leaves a bit to be desired. When recording magpie salute last summer it looked like I was going way over, but it ended up ok.

That said, with the 4060/6060’s that’s a good tip.

With the 4061/6061’s...I think those into a dvice can take an awful lot. (Clearly need more science backing this up!)

Offline detroit lightning

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Re: DPA d:VICE just announced
« Reply #386 on: May 07, 2019, 08:44:43 PM »
app is inconsistent for me. they say it does a bunch of things like prevent input overload, etc. with level-adjusting apps like voicerecorderpro and metarecorder it seems to be overridden functionally

i open it up to make sure it sees both mics and the HPF is off, levels are ganged approriately (stereo vs dual) then close it and use apogee metarecorder to adjust levels.

its kinda  frustrating that support cant explain exactly how it works, if there is a unity gain, and the best settings to set levels. they must have farmed out both hardware and software on this one.

it took a week or so to get an answer from TPTB on whether the d:vice is synchronous, adaptive, or asynchronous. I got this answer and told them to ask again just in case there was a typo or something lost in translation. was hoping the class II operation was asynchronous or at least adaptive. i have heard a click now and again in some recordings but hard to tell if that was just some room noise or something else, not necessarily clock adjusting

The answer from our tech team is;
By connection to PC USB Audio class I is used, in synchronous mode
By connection to Iphone USB Audio class II is used, in synchronous mode


a little background on USB audio from Cambridge

Synchronous USB DAC is the lowest quality of the three often used in low end products, so none of ours course! (that’s right, we’re being smug!) These accept the packets of data whenever the data sends them causing glitches every few seconds due to the differences between the two clocks.

Adaptive is where the DAC constantly adjusts its clock so that it can accept the data being sent from the computer whenever it sends them. The constant adapting of the DAC’s clock means that there is no continuous, accurate master clock in the DAC, which causes jitter in the audio stream.

In both of the above the computer dictates the timing of the data packets being sent.

Asynchronous – this is the most complex to implement but it is a huge improvement on the other types. This is because it requests the data packets to be sent in time with its own clock’s timing, thus providing the lowest jitter and sounding by far the best.


Is jitter in this setup something to be concerned about? hard to say. The effect of jitter seems to be a function of your playback setup as well.

From John Siau (Benchmark), answering in response to feeding DAC1 class 1 usb and also a discussion of playback jitter reduction (emphasis mine):

Up until Windows 7, asynchronous USB interfaces offered no advantages other than a possible reduction in jitter for DACs that lacked proper jitter attenuation.  The Benchmark DACs have excellent jitter attenuation, and the asynchronous interface offered absolutely no improvement in jitter performance.  The jitter rejection is nearly perfect in both cases.

The V-link will work well if it passes 24-bit audio, and supports the sample rates you are playing.  Jitter is never a concern with any Benchmark converter and this means that the v-link's relatively poor-quality synthesized clock will not be an issue.


Doug Oade (again emphasis mine):

The fact jitter will be at unacceptable levels simply demands you use a DAC with an async USB input. That will reduce jitter to the DAC clock's jitter. S/PDIF, which is supported by the controller, has a more stable clock easing the requirements of the DAC. But even a very stable S/PDIF clock in a portable ADC can be improved by reclocking. I use an Apogee Big Ben for that, when needed, in my studio but a high end DAC with an async USB input outperforms the Big Ben. All high end DAC manufacturers now offer async USB inputs with femtosecond accurate clocks. The difference is easy to hear.


i have a tenuous understanding of the effects of jitter, but if you have jitter in the input portion of the recording, how is this clock jitter not a permanent artifact of the recording. in other words, if the data is written 'wrong', how can it be 'read right' later?

and if sound is so dependent on the playback device (which most people in the world are listening on low end equipment, phones, car stereos, computers with dubious interfaces, integrated devices, etc), would it be wise to "clean your clock" by feeding it through a high-end dac>balanced interconnects>high-end low-jitter ADC like an AD2K etc.

I’m going to have to wrap my head around all this...

Side note, I’m about to tape a jazz show and Messing with settings - the levels set in the dpa app definitely seem To have an impact on the overall levels. So now it’s about finding unity I suppose...

Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #387 on: May 07, 2019, 10:24:55 PM »
they do but you may be able to do this experiment (works with apogee app, which you can use for free to demo, 1 minute recordings)

set levels in dpa app to minimum - open apogee app, they are at minimum. adjust in apogee app to maximum, go back and see that they were boosted in dpa app as well

when i start messing with levels in both apps sometime level adjustment locks or shows weirdness, so i close dpa app once im happy with hpf and mono/stereo/dual settings
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Offline scottE

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Re: DPA d:VICE just announced
« Reply #388 on: May 10, 2019, 10:55:15 AM »
Perhaps strange question but is there any way to connect soundprofessionals  mics to the d:vice? Not on a big budget so step one would be getting d:vice and use it with the sp mics and then later on buying dpa mics if necessary

For example with this adapter: https://www.ebay.com/itm/MICROPHONE-ADAPTER-FOR-SENNHEISER-3-5mm-JACK-PLUG-TO-SCREW-ON-1-PIN-DPA-MICRODOT-/273220934619
« Last Edit: May 11, 2019, 10:58:06 AM by scottE »
SP-BMC-12
SP-SPSB-6

PCM-A10

Online jerryfreak

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Re: DPA d:VICE just announced
« Reply #389 on: June 03, 2019, 02:01:51 AM »
did my first master/satellite with metarecorder and 2 d:vices last night

1. im 99% sure they are not TC or clock linked
2. i ran into unpredictable behavior which resulted in lost recording
« Last Edit: June 03, 2019, 04:13:57 AM by jerryfreak »
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