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Audient id14 Zero Latency?
Old 2nd November 2015
  #1
Audient id14 Zero Latency?

Does anyone know if it has zero latency hardware monitoring?

thanks
Old 3rd November 2015
  #2
OK so it looks like we don't know. Audient aren't clear themselves to be fair - when I called them they said yes but I read in their literature somewhere that source monitoring was in fact a cue mix with under 1.6 ms delay. So its one of those 'near zero' latency devices. That's well under the 6-20 ms latencies up and down a usb cable, but still probably perceptible - I already have a 'near zero' device running at about that, but DI guitar feels weird through it. You can play, but it's a little off.

I could buy a mixer, separate DI with dual outs, or even a Y cable and plug in a micro amp, but I'm interested in true hardware 0 latency/direct monitoring to avoid clutter. I'm quite happy with a clean signal for recording with the right groove than a delayed vst amp sim with great tone - I don't go too grungy or heavy anyway.

So far I've come across Steinberg UR22 and Roland UA55 that explicitly have hardware direct monitoring, but I fear they're not up to the audio quality of the ID14. Am I wrong? I'm not on a tonequest, but RME/Audient level quality would be nice. Do I need to save up more or get the Audient plus mixer/amp?
Old 3rd November 2015
  #3
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You always need some time for actual AD/DA conversion (one direction from 0.25 to 1ms @ 48k depending on used chip) and processing in the DSP (usually couple of samples)..
Something like zero latency is possible only, when you completely stay at analog domain.
So it is quite simple, every audio interface with DSP monitoring, digital mixer or multi FX unit has typically fixed total delay from 1 to 3ms (FOH desk with remote stageboxes for instance).
In case of audio interfaces, analog input monitoring is implemented as a simple blendpot in case of all cheaper models without DSP or as more elaborate monitor controller (SPL Crimson for example).
I'm not really fan of those simple stereo blendpots, because they can wear over time and can be subject of crackles, low level channel imbalances etc. Also DSP mixer usually offers more options for cue mix.. (panning, mixing several different ASIO channels from DAW, EQ and Reverb in case of RME TotalMix.. ).

Personally I doubt, it will really affect performance (especially if player don't know about it). 1ms ~ 1feet distance of sound propagation in air. Lot of people play through various digital boxes (from any simple digital delay, Line6 effects to Kempers), which shares same inherent delay, without any feel of disconnection or timing issues. Similarly if you don't have head directly at your amplifier grill, its distance from your ears in room can easily create few milliseconds of natural delay. Or guys which plays live gigs with in-ear monitors and digital monitor desks for easy recall.
So I'm quite skeptic about that, and when someone concerns about that or request analog monitoring, when he knows about possible delays with (near zero) digital.. I can usually fulfill his wish just to make him happy (that's most important), but to me it doesn't have any rational basis. And frankly, their timing issues are much more related to their natural feeling, practice and instrument playing abilities IME.

Michal
Old 3rd November 2015
  #4
Thanks Michal, interesting to learn about the 1ms/foot propagation. That kinda puts it in perspective. If the Audient is 2-3 ms with its cue mix then that should not be noticeable, as even a practice amp can be a good few feet away and not cause issues.

Can anyone feed back on their DI experience with the ID14 please?
Old 3rd November 2015 | Show parent
  #5
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audientworld's Avatar
Quote:
Originally Posted by Shaolin View Post
Thanks Michal, interesting to learn about the 1ms/foot propagation. That kinda puts it in perspective. If the Audient is 2-3 ms with its cue mix then that should not be noticeable, as even a practice amp can be a good few feet away and not cause issues.

Can anyone feed back on their DI experience with the ID14 please?
Hi everyone,

I can confirm that the "Cue Mix" part of iD14 (and iD22) is the direct monitoring part of the interfaces which will give you roughly 1.6ms latency at 44.1kHz which reduces as you increase the sample rate (I believe roughly 0.6ms at 96kHz off the top of my head). So it isn't zero latency, which as mentioned is only really possible in the analogue domain, but it shouldn't really be perceivable at <1.6 ms.

Thanks,
Harry
Old 4th November 2015
  #6
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shatz's Avatar
I just got my ID14. I'm using it at home for editing and midi recording. So far I'm getting pretty bad performance with it. At my commercial facility I use an Antelope Orion 32, and I'm able to run it at 48 kHz, 64 buffer with minimum latency setting and runs perfect. With the id14 I have it set to 256 buffer low latency setting and I'm getting pops when trying to use Sampletank 3 in Pro Tools HD 12. I was using an old M-Audio Profire 2626 and the performance was better with that as well. So far I'm not impressed.

How do I go about setting this up so latency is minimal and performance doesn't suffer?
Old 4th November 2015 | Show parent
  #7
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Quote:
Originally Posted by shatz View Post
I just got my ID14. I'm using it at home for editing and midi recording. So far I'm getting pretty bad performance with it. At my commercial facility I use an Antelope Orion 32, and I'm able to run it at 48 kHz, 64 buffer with minimum latency setting and runs perfect. With the id14 I have it set to 256 buffer low latency setting and I'm getting pops when trying to use Sampletank 3 in Pro Tools HD 12. I was using an old M-Audio Profire 2626 and the performance was better with that as well. So far I'm not impressed.

How do I go about setting this up so latency is minimal and performance doesn't suffer?
We've discussed slightly different thing in our posts above - it was about direct monitoring latency, when input audio isn't passing through DAW, but is routed directly at interface to some cue bus.

Your issue - latency and performance during VSTi playback is bit of complex thing, because almost everything at your computer can be involved - used computer components, drivers for the interface, drivers for other devices, used DAW (there are differences in effeciency among them), project itself (plugins and its placement in project, which can affect scheduling on multicore CPUs).

Especially important IME are other drivers and processes at your system, which can affects lowest obtainable latency. Simplified technical reason is, at Windows you have dedicated queues for processing of events from all active system drivers and whatever driver (like for WiFi interface, bluetooth, antivirus software, whose network scanning module operates at that level), which takes longer time to process, can delay others including audio interface driver. So audio transfers to your interface can take longer time than on tuned system and pops and crackles appears with the same buffer length.
So think about that as a prerequisite for getting good low latency performance using Windows regardless of used interface.
Vendors usually provides several settings at USB ASIO driver (eg. profiles for low latency, standard, relaxed.. ), which controls additional buffering, besides chosen DAW buffer length, to smooth out audio delivery.
Firewire and PCIe interfaces are usually bit more efficient with audio transfers and doesn't require so long additional buffers, because they can use DMA, which isn't really possible to do with standard USB.

So I believe, you can achieve lower latency without crackles at your iD14 (eg. 64s and low-latency profile), but you have to tune your system and experiment with it.
Orion32 uses very similar 3rd party class compliant USB audio driver like Audient interfaces, so performance and latency figures will be pretty close IMO.
For start, I would recommend to take your iD14 to your studio and install it to that same computer as your Orion32.. (no worries, both drivers can coexist). Then compare its performance with same project.
If it will be better than at your home computer, focus to its optimization.

Here are couple of my favorite links about that.
https://youtu.be/BnbfVFqkKOA
https://www.native-instruments.com/e...io-processing/

Plus you can find also quite a lot optimization information (not specific to Audient) at other threads here (search for DPC latency).

Michal

Last edited by msmucr; 4th November 2015 at 10:52 AM..
Old 4th November 2015 | Show parent
  #8
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Quote:
Originally Posted by msmucr View Post
We've discussed slightly different thing in our posts above - it was about direct monitoring latency, when input audio isn't passing through DAW, but is routed directly at interface to some cue bus.

Your issue - latency and performance during VSTi playback is bit of complex thing, because almost everything at your computer can be involved - used computer components, drivers for the interface, drivers for other devices, used DAW (there are differences in effeciency among them), project itself (plugins and its placement in project, which can affect scheduling on multicore CPUs).

Especially important IME are other drivers and processes at your system, which can affects lowest obtainable latency. Simplified technical reason is, at Windows you have dedicated queues for processing of events from all active system drivers and whatever driver (like for WiFi interface, bluetooth, antivirus software, whose network scanning module operates at that level), which takes longer time to process, can delay others including audio interface driver. So audio transfers to your interface can take longer time than on tuned system and pops and crackles appears with the same buffer length.
So think about that as a prerequisite for getting good low latency performance using Windows regardless of used interface.
Vendors usually provides several settings at USB ASIO driver (eg. profiles for low latency, standard, relaxed.. ), which controls additional buffering, besides chosen DAW buffer length, to smooth out audio delivery.
Firewire and PCIe interfaces are usually bit more efficient with audio transfers and doesn't require so long additional buffers, because they can use DMA, which isn't really possible to do with standard USB.

So I believe, you can achieve lower latency without crackles at your iD14 (eg. 64s and low-latency profile), but you have to tune your system and experiment with it.
Orion32 uses very similar 3rd party class compliant USB audio driver like Audient interfaces, so performance and latency figures will be pretty close IMO.
For start, I would recommend to take your iD14 to your studio and install it to that same computer as your Orion32.. (no worries, both drivers can coexist). Then compare its performance with same project.
If it will be better than at your home computer, focus to its optimization.

Here are couple of my favorite links about that.
https://youtu.be/BnbfVFqkKOA
https://www.native-instruments.com/e...io-processing/

Plus you can find also quite a lot optimization information (not specific to Audient) at other threads here (search for DPC latency).

Michal
I've managed to setup the card so it doesen't pop and crackle all the time. Stability is pretty good, however the minimum latency setting stresses the CPU really hard. I've noticed +50% CPU usage at that setting without doing anything (idling in windows desktop). And I have a pretty good computer, so I am stuck to using LOW at best. Also the firmware is messed up, there is also no way to turn off the soundcard on the interface itself (other then plugging off the cables which is annoying), and if you leave the soundcard on there is a bug that prevents your computer going to sleep. They are "working on fix", for months now. So yeah. I don't understand the logic behind the, lets fix the scroll thingie but introduce such annoying bug for it.. wow. Drivers could be better imo. Other then that the interface has great sounding mic preamps, awesome DI input which is really really good (sounds like an old console table that David Gilmour used on another brick from the wall solo) and pretty decent headphones amp which is able to drive my DT880pro at 250ohms. I wish it had RME drivers, then it would be perfect interface.
Old 4th November 2015 | Show parent
  #9
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shatz's Avatar
I forgot to mention that, but it's the same exact computer, I just bring it home with me. So nothing changes except for the interface.

EDIT:
The id14 performance is not anywhere close to the Antelope. I know the price tag is a huge difference, but I should still be able to run one instance of a piano and not get pops.

Quote:
Originally Posted by msmucr View Post
We've discussed slightly different thing in our posts above - it was about direct monitoring latency, when input audio isn't passing through DAW, but is routed directly at interface to some cue bus.

Your issue - latency and performance during VSTi playback is bit of complex thing, because almost everything at your computer can be involved - used computer components, drivers for the interface, drivers for other devices, used DAW (there are differences in effeciency among them), project itself (plugins and its placement in project, which can affect scheduling on multicore CPUs).

Especially important IME are other drivers and processes at your system, which can affects lowest obtainable latency. Simplified technical reason is, at Windows you have dedicated queues for processing of events from all active system drivers and whatever driver (like for WiFi interface, bluetooth, antivirus software, whose network scanning module operates at that level), which takes longer time to process, can delay others including audio interface driver. So audio transfers to your interface can take longer time than on tuned system and pops and crackles appears with the same buffer length.
So think about that as a prerequisite for getting good low latency performance using Windows regardless of used interface.
Vendors usually provides several settings at USB ASIO driver (eg. profiles for low latency, standard, relaxed.. ), which controls additional buffering, besides chosen DAW buffer length, to smooth out audio delivery.
Firewire and PCIe interfaces are usually bit more efficient with audio transfers and doesn't require so long additional buffers, because they can use DMA, which isn't really possible to do with standard USB.

So I believe, you can achieve lower latency without crackles at your iD14 (eg. 64s and low-latency profile), but you have to tune your system and experiment with it.
Orion32 uses very similar 3rd party class compliant USB audio driver like Audient interfaces, so performance and latency figures will be pretty close IMO.
For start, I would recommend to take your iD14 to your studio and install it to that same computer as your Orion32.. (no worries, both drivers can coexist). Then compare its performance with same project.
If it will be better than at your home computer, focus to its optimization.

Here are couple of my favorite links about that.
https://youtu.be/BnbfVFqkKOA
https://www.native-instruments.com/e...io-processing/

Plus you can find also quite a lot optimization information (not specific to Audient) at other threads here (search for DPC latency).

Michal

Last edited by shatz; 4th November 2015 at 03:54 PM..
Old 4th November 2015 | Show parent
  #10
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Quote:
Originally Posted by msmucr View Post

personally i doubt, it will really affect performance (especially if player don't know about it). 1ms ~ 1feet distance of sound propagation in air. Lot of people play through various digital boxes (from any simple digital delay, line6 effects to kempers), which shares same inherent delay, without any feel of disconnection or timing issues. Similarly if you don't have head directly at your amplifier grill, its distance from your ears in room can easily create few milliseconds of natural delay. Or guys which plays live gigs with in-ear monitors and digital monitor desks for easy recall.
So i'm quite skeptic about that, and when someone concerns about that or request analog monitoring, when he knows about possible delays with (near zero) digital.. I can usually fulfill his wish just to make him happy (that's most important), but to me it doesn't have any rational basis. And frankly, their timing issues are much more related to their natural feeling, practice and instrument playing abilities ime.
x 1000000
Old 4th November 2015
  #11
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Quote:
Originally Posted by shatz View Post
I forgot to mention that, but it's the same exact computer, I just bring it home with me. So nothing changes except for the interface.

EDIT:
The id14 performance is not anywhere close to the Antelope. I know the price tag is a huge difference, but I should still be able to run one instance of a piano and not get pops.
Hmm, I don't know what exactly is culprit there.. I don't have any iD14 nor ProTools for further precise help and some step-by-step guides, my experience comes from installation of iD22 at friend's Windows computer before some time, who still use it. Performance wasn't really RME, but nothing really excessively bad. Definitely in-line with other similar USB interfaces like Scarletts or URs by Steinberg.

But as I wrote before, I still believe iD14 (which shares same USB driver with iD22), can be usually tweaked to perform with shorter buffers than 256 samples without crackles.
You've reported good latency figures (for USB interface) with Orion32 at the same computer, so most probably it won't be mentioned problem with other device drivers or processes, causing latency spikes.. that's good thing and we can most probably rule it out. Assuming, there isn't other difference in computer usage (eg. WiFi at home, wired ethernet at studio.. use of same physical USB port).
Then there can be some other things, which can cause that.. Like some compatibility issue with your USB chipset and iD14, so it can perform suboptimal.. similarly you'll sometimes find updated ASIO drivers for interfaces, where is solved some problem with specific DAW (like ProTools in your case).
When I'm debugging problems, I usually try trial version of Reaper (small installation, easy to get) to rule that out.
I think, it will be worth of attempt to contact guys at Audient support, with exact description of your problem, used system.. I've contacted them once for some pre-sales question and they were super helpful. They should certainly tell you about obtainable latency figures with iD14, possibly proceed through some diagnostic steps. And generally speaking, vendor support is sometimes able to provide for instance some beta version of upcoming driver or firmware, which can address particular issue.

Michal
Old 4th November 2015 | Show parent
  #12
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shatz's Avatar
Quote:
Originally Posted by msmucr View Post
Hmm, I don't know what exactly is culprit there.. I don't have any iD14 nor ProTools for further precise help and some step-by-step guides, my experience comes from installation of iD22 at friend's Windows computer before some time, who still use it. Performance wasn't really RME, but nothing really excessively bad. Definitely in-line with other similar USB interfaces like Scarletts or URs by Steinberg.

But as I wrote before, I still believe iD14 (which shares same USB driver with iD22), can be usually tweaked to perform with shorter buffers than 256 samples without crackles.
You've reported good latency figures (for USB interface) with Orion32 at the same computer, so most probably it won't be mentioned problem with other device drivers or processes, causing latency spikes.. that's good thing and we can most probably rule it out. Assuming, there isn't other difference in computer usage (eg. WiFi at home, wired ethernet at studio.. use of same physical USB port).
Then there can be some other things, which can cause that.. Like some compatibility issue with your USB chipset and iD14, so it can perform suboptimal.. similarly you'll sometimes find updated ASIO drivers for interfaces, where is solved some problem with specific DAW (like ProTools in your case).
When I'm debugging problems, I usually try trial version of Reaper (small installation, easy to get) to rule that out.
I think, it will be worth of attempt to contact guys at Audient support, with exact description of your problem, used system.. I've contacted them once for some pre-sales question and they were super helpful. They should certainly tell you about obtainable latency figures with iD14, possibly proceed through some diagnostic steps. And generally speaking, vendor support is sometimes able to provide for instance some beta version of upcoming driver or firmware, which can address particular issue.

Michal
Hi Michal, thanks for your help and detailed responses. Much appreciated.
Old 4th November 2015 | Show parent
  #13
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Quote:
Originally Posted by yuka View Post
I've managed to setup the card so it doesen't pop and crackle all the time. Stability is pretty good, however the minimum latency setting stresses the CPU really hard. I've noticed +50% CPU usage at that setting without doing anything (idling in windows desktop). And I have a pretty good computer, so I am stuck to using LOW at best. Also the firmware is messed up, there is also no way to turn off the soundcard on the interface itself (other then plugging off the cables which is annoying), and if you leave the soundcard on there is a bug that prevents your computer going to sleep. They are "working on fix", for months now. So yeah. I don't understand the logic behind the, lets fix the scroll thingie but introduce such annoying bug for it.. wow. Drivers could be better imo. Other then that the interface has great sounding mic preamps, awesome DI input which is really really good (sounds like an old console table that David Gilmour used on another brick from the wall solo) and pretty decent headphones amp which is able to drive my DT880pro at 250ohms. I wish it had RME drivers, then it would be perfect interface.
Generally when you decrease buffer length, it always leads to increased CPU usage consumption. This applies to DAWs, where larger chunks of data are processed more efficiently, and also to drivers, because short buffer means more calls and interrupts with its associated overhead.
USB interfaces with very efficient proprietary drivers (like RME), PCIe ones or Firewire interfaces with DMA.. has base CPU consumption for audio transfers via its driver so low, that its increase will be apparent with shorter buffers.
It can be seen by great latency table by Vin Curigliano
DAW Bench : DAW Performance Benchmarking
Check the CV and NCV columns, which tells how many instances of Kontakt is possible to use without audio dropouts.
Increase of CPU usage at your system during its apparent idling can be caused by mapping of Windows sounds through iD14 sound device. Audio transfers via WDM interface of its driver aren't much different from ASIO and are also affected by buffer settings. Sometimes just one minimized application like web browser can leave opened audio device although there is nothing playing.
Just for the test, you can try to disable iD14 from Windows playback devices list (right click context menu) to see if that "idle" load decreases.

I see your other listed iD14 negatives, but it would probably fall into cosmetic bugs category for me personally.
I treat every DAW computer, I'm installing, basically as more or less dedicated machine, where all power saving features should be disabled, when it is used for audio work.. This includes also settings, which prevent USB audio device to power off in saving mode, at USB root hub driver preferences. Similarly I usually employ customized High performance power profile at Windows (when someone doesn't want that, I will create shortcuts for easy switching).
Lack of power switch is common thing for bus powered USB interfaces and can be solved by simple in-line power switch, when required USB Cable with Switch ID: 1620 - $2.95 : Adafruit Industries, Unique & fun DIY electronics and kits
(just one notice for its usage, there can be quite loud thump into connected speakers when turning on the interface, so just pay attention to power it before speakers).

And I also like Audient's sound (based on, what I've heard from iD22). I think they packed really good sounding interface to that affordable category (actually exceeding quite lot of more expensive pieces).. intuitive GUI for cue mix is plus for musicians. Quality of preamps, converters, volume control and headphone amp was really surprising for me at that price level and I feel those as its strong points.
Drivers (I'm not talking about mixer or GUI) are average.. IME not better or significantly worse among other similar USB class compliant interfaces, which are mostly implemented at XMOS microcontrollers with 3rd party driver.
Step up in terms of performance and efficiency are RME or MOTU USB devices, but I tend to prefer RME at Windows due to quite few issues with MOTU, I had in the past.
Somewhat in-between are Steinberg USB devices with their own driver and Yamaha USB streaming chip, which are quite reliable performers with lower CPU load.
So if one has sufficient budget for interface purchase, it is quite clear recommendation to me.. RME, possibly augmented by external preamps (for instance by Audient ). I feel that almost mandatory for live performance or heavy use of VSTi.. finding of alternatives is usually just waste of money and time.
In lower price levels, one has to pick and Audient and SPL are strong at audio quality. I feel them mainly as audio hardware companies, so I see they hadn't developed their audio streaming at FPGAs or dedicated chips, similarly I can see, they used 3rd party driver. But depending on intended use, its latency performance can be absolutely fine for lot of folks out there. Especially when they're not using any amp simulators, monitoring direct and when they use VSTis, actual performance means drawing of rectangles into piano roll (or they're sloppy chord pushers like me ), so longer latency isn't really an issue. On the other hand, they can appreciate quality of preamp and DI when recording live instruments or singing.

Michal
Old 5th November 2015 | Show parent
  #14
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Quote:
Originally Posted by msmucr View Post
Generally when you decrease buffer length, it always leads to increased CPU usage consumption. This applies to DAWs, where larger chunks of data are processed more efficiently, and also to drivers, because short buffer means more calls and interrupts with its associated overhead.
USB interfaces with very efficient proprietary drivers (like RME), PCIe ones or Firewire interfaces with DMA.. has base CPU consumption for audio transfers via its driver so low, that its increase will be apparent with shorter buffers.
It can be seen by great latency table by Vin Curigliano
DAW Bench : DAW Performance Benchmarking
Check the CV and NCV columns, which tells how many instances of Kontakt is possible to use without audio dropouts.
Increase of CPU usage at your system during its apparent idling can be caused by mapping of Windows sounds through iD14 sound device. Audio transfers via WDM interface of its driver aren't much different from ASIO and are also affected by buffer settings. Sometimes just one minimized application like web browser can leave opened audio device although there is nothing playing.
Just for the test, you can try to disable iD14 from Windows playback devices list (right click context menu) to see if that "idle" load decreases.

I see your other listed iD14 negatives, but it would probably fall into cosmetic bugs category for me personally.
I treat every DAW computer, I'm installing, basically as more or less dedicated machine, where all power saving features should be disabled, when it is used for audio work.. This includes also settings, which prevent USB audio device to power off in saving mode, at USB root hub driver preferences. Similarly I usually employ customized High performance power profile at Windows (when someone doesn't want that, I will create shortcuts for easy switching).
Lack of power switch is common thing for bus powered USB interfaces and can be solved by simple in-line power switch, when required USB Cable with Switch ID: 1620 - $2.95 : Adafruit Industries, Unique & fun DIY electronics and kits
(just one notice for its usage, there can be quite loud thump into connected speakers when turning on the interface, so just pay attention to power it before speakers).

And I also like Audient's sound (based on, what I've heard from iD22). I think they packed really good sounding interface to that affordable category (actually exceeding quite lot of more expensive pieces).. intuitive GUI for cue mix is plus for musicians. Quality of preamps, converters, volume control and headphone amp was really surprising for me at that price level and I feel those as its strong points.
Drivers (I'm not talking about mixer or GUI) are average.. IME not better or significantly worse among other similar USB class compliant interfaces, which are mostly implemented at XMOS microcontrollers with 3rd party driver.
Step up in terms of performance and efficiency are RME or MOTU USB devices, but I tend to prefer RME at Windows due to quite few issues with MOTU, I had in the past.
Somewhat in-between are Steinberg USB devices with their own driver and Yamaha USB streaming chip, which are quite reliable performers with lower CPU load.
So if one has sufficient budget for interface purchase, it is quite clear recommendation to me.. RME, possibly augmented by external preamps (for instance by Audient ). I feel that almost mandatory for live performance or heavy use of VSTi.. finding of alternatives is usually just waste of money and time.
In lower price levels, one has to pick and Audient and SPL are strong at audio quality. I feel them mainly as audio hardware companies, so I see they hadn't developed their audio streaming at FPGAs or dedicated chips, similarly I can see, they used 3rd party driver. But depending on intended use, its latency performance can be absolutely fine for lot of folks out there. Especially when they're not using any amp simulators, monitoring direct and when they use VSTis, actual performance means drawing of rectangles into piano roll (or they're sloppy chord pushers like me ), so longer latency isn't really an issue. On the other hand, they can appreciate quality of preamp and DI when recording live instruments or singing.

Michal
Great reply Michal, very good info here. I am aware of the CPU usage as to something quite normal however I am very suprised because it feels like a firmware bug, since all other settings (other then minimum 64 samples) are not producing any CPU usage. Also I previously had EMU 0404 USB interface which never stressed my CPU at anything above 1-2% in idle at any buffer setting.

Also one of the thing I noticed is that Audient ID14 doesen't utilise something as I previously had called "SOFT clipping". Maybe I am mistaken and I am in no means an expert in this field, but whenever something CLIPS at Audient ID 14 there is a huge POP and Crackle which makes everything unusable. I know cliping is something you need to take care off in the leveling stage, but for example when I use the DI input and plug in the guitar, I could set it a bit higher gain level to make the sound distorted, it creates an awesome buzzing distortion, but at every hard stroke it clips and pops really hard make it unusable. When I used EMU 0404 I could enable soft clip and then put gain considerably higher while still not getting unusable clips and pops. Sure the sound would clip but it wouldn't produce this loud pops.. It would produce natural distortion\buzz. Something I very much miss. Similair thing I found when using a mic. I had to have a pop filter for my Shure SM57 otherwise it was impossible to feed the mic enough gain to make it warm and still prevent POPS.
Old 5th November 2015
  #15
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DistortingJack's Avatar
 

Just saying, using the iD14 on Logic recording DI guitar with Scuffham modeling at 88.2 kHz sampling rate has no noticeable latency with a 128 buffer size.
People whining about 1.6 ms conversion delay recording God knows what, but here I am recording a metalhead sweep-picking like a mofo without any timing problems or whining whatsoever.

Blaming the tools, yadda yadda.

Quote:
Originally Posted by yuka View Post
Also one of the thing I noticed is that Audient ID14 doesen't utilise something as I previously had called "SOFT clipping". Maybe I am mistaken and I am in no means an expert in this field, but whenever something CLIPS at Audient ID 14 there is a huge POP and Crackle which makes everything unusable. I know cliping is something you need to take care off in the leveling stage, but for example when I use the DI input and plug in the guitar, I could set it a bit higher gain level to make the sound distorted, it creates an awesome buzzing distortion, but at every hard stroke it clips and pops really hard make it unusable. When I used EMU 0404 I could enable soft clip and then put gain considerably higher while still not getting unusable clips and pops. Sure the sound would clip but it wouldn't produce this loud pops.. It would produce natural distortion\buzz. Something I very much miss. Similair thing I found when using a mic. I had to have a pop filter for my Shure SM57 otherwise it was impossible to feed the mic enough gain to make it warm and still prevent POPS.
Don't clip your levels.
One of the big things about high-end conversion and analogue circuitry is that you can have your levels 30 dB down and still not have background noise in your final product.
Setting gains properly is part of the workflow in any professional environment โ€“ I can only imagine what would happen with you at the helm of a 32-channel band recording...

Last edited by DistortingJack; 5th November 2015 at 12:06 PM..
Old 5th November 2015 | Show parent
  #16
Here for the gear
 

Quote:
Originally Posted by DistortingJack View Post
Just saying, using the iD14 on Logic recording DI guitar with Scuffham modeling at 88.2 kHz sampling rate has no noticeable latency with a 128 buffer size.
People whining about 1.6 ms conversion delay recording God knows what, but here I am recording a metalhead sweep-picking like a mofo without any timing problems or whining whatsoever.

Blaming the tools, yadda yadda.



Don't clip your levels.
One of the big things about high-end conversion and analogue circuitry is that you can have your levels 30 dB down and still not have background noise in your final product.
Setting gains properly is part of the workflow in any professional environment โ€“ I can only imagine what would happen with you at the helm of a 32-channel band recording...
I totally agree with you as I said and you didn't understand me correctly, I was merley saying that a CLIP sounds rlly harsh it makes a large POP, because the peaks get too much attention, they are not softened. I was spoiled by many soundcards that add soft-clipping as a native option, soft clipping makes this peaks sound less harsh and more soft so when you jam on some backing tracks for friends sometimes you can push that gain abit harder and not get that killer POP that makes it unplayable, of course your wouldn't record that signal. But again I can use plugins that do the same thing, or use less gain and push it with Amplitube or something, but is just a thing I noticed. (Also maybe there is such a thing builtin already or something and I am totally wrong here )

P.S I love Scuffham some awesome cabs there
Old 10th March 2016
  #17
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Basspartout's Avatar
Just went from a Scarlett 6i6 to the Audient ID14. Really love the sound of the ID 14 but the latency is much worse than the Focusrite. If there is no driver or firmware update soon I think I will return it and go for a RME. Too bad, because the sound quality you get for 259,- Euro is really wonderful.
Old 10th March 2016 | Show parent
  #18
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DistortingJack's Avatar
 

Quote:
Originally Posted by Basspartout View Post
Just went from a Scarlett 6i6 to the Audient ID14. Really love the sound of the ID 14 but the latency is much worse than the Focusrite. If there is no driver or firmware update soon I think I will return it and go for a RME. Too bad, because the sound quality you get for 259,- Euro is really wonderful.
I'm assuming that it's a PC and that something is wonky.
Focusrite are renowned for having terrible latency and Audient really aren't.

Muck about with settings for a while because it would be unusual if that were the case.
Old 17th March 2016 | Show parent
  #19
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Quote:
Originally Posted by DistortingJack View Post
I'm assuming that it's a PC and that something is wonky.
Focusrite are renowned for having terrible latency and Audient really aren't.

Muck about with settings for a while because it would be unusual if that were the case.
I agree 100%. I just switched out a Focusrite for an Audient and the difference has blown me away. Much lower latency, FAR better pres and convertors and the DI is fantastic as well.

It is like going from 2d to 3d honestly. The focusrite really is not even close.

It definitely sounds like a PC issue. I tweaked my Win 10 PC a lot before I got it optimized, but the final straw was pulling out the focusrite and replacing it with the ID14. Clicks and Pops gone. Random noises sometimes getting in when tracking, gone. Tone and depth of the instruments and the mix - WAY better.
Old 17th March 2016 | Show parent
  #20
Gear Maniac
 

Quote:
Originally Posted by Methlab View Post
I agree 100%. I just switched out a Focusrite for an Audient and the difference has blown me away. Much lower latency, FAR better pres and convertors and the DI is fantastic as well.

It is like going from 2d to 3d honestly. The focusrite really is not even close.

It definitely sounds like a PC issue. I tweaked my Win 10 PC a lot before I got it optimized, but the final straw was pulling out the focusrite and replacing it with the ID14. Clicks and Pops gone. Random noises sometimes getting in when tracking, gone. Tone and depth of the instruments and the mix - WAY better.

What latency are you getting? What is your mobo?
I agree ID14 blows the Focusrite away with sound quality, but there are latency issues (that the FR didn't have on the same setup).
Old 17th March 2016 | Show parent
  #21
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Quote:
Originally Posted by pdpd View Post
What latency are you getting? What is your mobo?
I agree ID14 blows the Focusrite away with sound quality, but there are latency issues (that the FR didn't have on the same setup).
I can get it around 2-4ms depending the tune, which is fine for me.

I'm running Windows 10 with a pretty old school setup, but it works real well for tracking since I only record 1 track at a time :

Motherboard - GIGABYTE|GA-880GMA-UD2H R

CPU - AMD|Phenom II X4 965 3.4G AM3 RT

16 gigs of Gskillz RAM

The latency issues I had with the Focusrite are far worse than anything I have experienced so far with the Audient. That being said, I am due for a full Mobo/CPU upgrade so I could achieve even better results with a faster CPU.

Setting up the Audient was so much easier. I just downloaded the drivers, installed them and the device was already up and running in StudioOne when I turned it on. Too easy.

It also has a lot more latency options, and FAR better software. Of course, the tone is clearly much better as well.
Old 17th March 2016 | Show parent
  #22
Gear Maniac
 

Quote:
Originally Posted by Methlab View Post
I can get it around 2-4ms depending the tune, which is fine for me.

I'm running Windows 10 with a pretty old school setup, but it works real well for tracking since I only record 1 track at a time :

Motherboard - GIGABYTE|GA-880GMA-UD2H R

CPU - AMD|Phenom II X4 965 3.4G AM3 RT

16 gigs of Gskillz RAM

The latency issues I had with the Focusrite are far worse than anything I have experienced so far with the Audient. That being said, I am due for a full Mobo/CPU upgrade so I could achieve even better results with a faster CPU.

Setting up the Audient was so much easier. I just downloaded the drivers, installed them and the device was already up and running in StudioOne when I turned it on. Too easy.

It also has a lot more latency options, and FAR better software. Of course, the tone is clearly much better as well.
2ms in total? That is great... out of interest what Latency and Buffer Size settings are you seeing under the 'Setup' menu in the iD software?
Old 17th March 2016 | Show parent
  #23
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stella645's Avatar
 

Quote:
Originally Posted by pdpd View Post
2ms in total? That is great... out of interest what Latency and Buffer Size settings are you seeing under the 'Setup' menu in the iD software?
Just to inject some reality here:

At 96k with a 32 sample buffer Audient quote 4.24ms. At 44.1k it's 6.33ms.

Thing is that there is no 32 sample option for Windows (on the iD22 at least)
So real world latency is way above the figures being thrown around here.
this is not to say it is unworkable, just don't expect miracles.
Old 17th March 2016 | Show parent
  #24
Gear Maniac
 

Quote:
Originally Posted by stella645 View Post
Just to inject some reality here:

At 96k with a 32 sample buffer Audient quote 4.24ms. At 44.1k it's 6.33ms.

Thing is that there is no 32 sample option for Windows (on the iD22 at least)
So real world latency is way above the figures being thrown around here.
this is not to say it is unworkable, just don't expect miracles.
Yep - I cant get below 256 samples here without crackling... and thats on a brand new i7 (very similar to results I got on my previous i3).

Latency I can get that is usable is ~13.6ms
I used to be able to get ~6ms on my Focusrite (same hardware - both the i7 and i3 machines)
Old 17th March 2016 | Show parent
  #25
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Quote:
Originally Posted by pdpd View Post
Yep - I cant get below 256 samples here without crackling... and thats on a brand new i7 (very similar to results I got on my previous i3).

Latency I can get that is usable is ~13.6ms
I used to be able to get ~6ms on my Focusrite (same hardware - both the i7 and i3 machines)
My settings are "low" and 128.

I get 4.9ms on my heaviest project. That's the actual reality of my setup, to address the other poster.

13.6ms - something is up there. It sounds like an optimization issue somewhere.
Old 18th March 2016 | Show parent
  #26
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Quote:
Originally Posted by Methlab View Post
My settings are "low" and 128.

I get 4.9ms on my heaviest project. That's the actual reality of my setup, to address the other poster.

13.6ms - something is up there. It sounds like an optimization issue somewhere.
I'm afraid, you're talking just about output latency actually reported by software..
To get some comparable value with what's reported by other users, you'll need to connect analog loop from output to input and measure it either by use of some special software like Oblique RTL utility or measure delay offset using your DAW with latency compensation turned off.
I haven't tried the iD14, but iD22 at 44.1k with 64s ASIO buffer has around 7ms in the best case.

4.9ms analog roundtrip latency with 128s buffer is very optimistic value even for fastest PCIe cards.

Michal
Old 18th March 2016
  #27
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m_gant's Avatar
 

Is anyone using it with a Mac & Logic Pro reported RTL?
Old 18th March 2016 | Show parent
  #28
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Quote:
Originally Posted by msmucr View Post
I'm afraid, you're talking just about output latency actually reported by software..
To get some comparable value with what's reported by other users, you'll need to connect analog loop from output to input and measure it either by use of some special software like Oblique RTL utility or measure delay offset using your DAW with latency compensation turned off.
I haven't tried the iD14, but iD22 at 44.1k with 64s ASIO buffer has around 7ms in the best case.

4.9ms analog roundtrip latency with 128s buffer is very optimistic value even for fastest PCIe cards.

Michal

I see what you are saying. It has not affected my recordings to where I need to go to that measure. Everything is in time and there is no discernible lag when I record, so I don't worry about it.
Old 18th March 2016
  #29
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stella645's Avatar
 

Quote:
Yep - I cant get below 256 samples here without crackling... and thats on a brand new i7 (very similar to results I got on my previous i3).
That does sound like a problem.....I can manage 64 or 128 in early stages of a project.
Only after adding some pretty heavy plugs do I need to think about 256 or upwards....And this is on a 6 year old i7 930 2.8Ghz.
Old 19th March 2016
  #30
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Basspartout's Avatar
Very interesting how different latency experiences are regarding the ID 14 and Focusrite.
Apart from my AMD Phenom six core PC with Win7 I tried the ID 14 with my Intel i7 Laptop Win10 from last year (both with 8 GB ram) and there the ID 14 worked significantly better.
So what was my choice? Getting a new (Intel) PC for my ID14?

Before I did that I decided last week to give the SPL Creon a try and behold: The Creon works exactly as good as the Focusrite 6i6 on my 'old' AMD.
So I'd say it is NOT necessarily my PC who's to blame here, right? It worked fine for three years now with the 6i6, why wouldnt it with another Interface.
It shows however, how it is always a matter of finding the right gear for your particular system and test your compatibilities.

The main crackling and latency problems with the ID14 on my system were caused
by Kontakt 5, especially when K5 reverbs were used, Addictive Drums 2, Arturia Analog Lab, Sample Tank 3. Interestingly enough not by UHE Diva for example.

And another interesting fact: all these VST mentioned produce some CPU peaks from time to time, visible in my DAW CPU meter (Studio One 3.2) BUT: they are not producing any crackles, meaning these CPU peaks are not audible with the Sacrlett or the SPL Creon.

The ID 14 is a really fine piece of gear, I'd say for the price you can't get a better sounding interface, it really sounds amazing, but the Windows drivers might not be as compatible as they could be so far.

Many thanks at this point to the very friendly and patient Audient support btw
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