I have not noticed the reported 7ms delay, but since I know it's there I want to get it as close to zero as possible and then recorde, worry-free. ;-) I used a number between interface reported and Utility results, did a loopback test with that number and adjusted it to align my two tracks. I'm at an offset of 387 samples, which in real-world measurements is a very small adjustment. Yet, I feel better knowing it's dialed in. Thanks for any and all comments.

 

didnt Tommy Say he Was on the Howard Stern Show? Real talk my man if minute latency is a problem you can go straight Analog.and ill even lookout for you ill sell you a Roland MC-303 Groovebox...thats with manual and only 6 pages are missing ... Same thing as the Original 303 promise

 

I'm not sure how to take your response.

But, why live with a small amount of latency if you can eliminate it? My quest is to find a simple, powerful way to record what's in my noodle. One last time . . . I'm trying to get a DAW setup setup.

 

Hello, the higher latencies are because you have an external audio interface. All USB device signals have to go through the USB cable. Therefore there is no value close to zero. If you buy a PCI sound card you will have much lower latencies.

If you feel like it, you can use this tool and post the results here.

LatencyMon: suitability checker for real-time audio and other tasks --> www.startpage.com/do/search

Test Results as an example:

Even though this test was carried out on a Mac, the Audient iD14 MKII was impressive in terms of latency. In Logic Pro X, the iD14 MKll provided very good latency values, as did Cubase. With 4.6 (4.7 in Cubase) seconds of latency with a buffer of 64 samples and 6.0 ms (6.1 ms in Cubase) with a 128 sample buffer, we have pretty good values. The connection was flawless throughout, with no annoying artifacts or the like.

This refers to latency, which is a delay in processing audio in real time. You can reduce the buffer size to reduce latency, but this can put more strain on your computer, which can cause audio glitches or dropouts.

Audient iD4 MKII and iD14 MKII - Audio Interface Review

 

Last edited by a moderator: Oct 24, 2024

Radio, Many thanks for the information and supportive response. I feel ever better about my interface. A PCI Soundcard is a PC thing, right? I don't think they are an option for a Mac. I'm at about 7ms, which seems small. Yet, I am going to offset that to get closer to zero. Why not? ;-)

 

Nope, that's generalized nonsense.

 

Hello Myfanwy, here you can see the difference between a PCI sound card (internal) and a USB audio interface (external)

PCI: Infrasonic - Quartet --> 44.1 kHz - 512 samples
USB: Audient - iD14 MKII --> 44.1 kHz - 256 samples

As you can see, the difference is big!

• PCIe – this is a card-based interface that you install directly into your computer’s motherboard. PCIe does a great job of diminishing latency to almost zero, and it gives you the ability to monitor/record virtually limitless tracks. PCI Express interfaces are very expensive and require quite a bit of technical know-how to install.

• USB 3.0, 2.0 and 1.1 – standard USB connectivity is probably the most common type you’ll see offered on audio interfaces. Standard USB interfaces are convenient and easy to set up, but they typically introduce more latency than other connection types.

• Thunderbolt – USB-C and Thunderbolt 3 or 4 are universally compatible. In fact, nowadays they’re pretty much one in the same. There are some interfaces that have dedicated Thunderbolt ports, but it’s all interchangeable. All in all, Thunderbolt is by far one the fastest and most reliable forms of connectivity offered on interfaces today.

• FireWire – at one point in time, FireWire was the standard connection type among interfaces. It was able to transfer data more consistently than USB, and was generally more reliable. You’d be hard pressed to find an audio interface that supports FireWire nowadays though. Almost no computer has a FireWire port anymore, and you’ll only find interfaces that offer it in used markets.

Ableton Live 10
48k, 64 samples: On: 4.58 ms - Off: 3.35 ms - Loop: 7.94 ms
48k, 265 Samples: On: 8.58 ms - Off: 7.35 ms - Loop: 15.9 ms
96k, 64 Samples: On: 3.92 ms - Off: 2.68 ms - Loop: 6.59 ms
96k, 256 Samples: On: 5.92 ms - Off: 4.68 ms - Loop: 10.8 ms

Test: M1 Mac mit Logic Pro und Cubase
- 4,6 Millisekunden bei 64 Sample-Puffer
- 6 Millisekunden bei 128 Sample-Puffer

Test: Presonus Studio One 7 (44,1 kHz - 256 Samples)
Input latency 7.30 ms --> Output latency 8.50 ms

Test: M1 Mac with Logic Pro and Cubase
- 4.6 milliseconds with 64 sample buffers
- 6 milliseconds with 128 sample buffers

Audient - iD14 MKII use the " USB-C "
1. USB 2.0: 480 Mbps (60 MB/s)
2. USB 3.0: 5 Gbps (640 MB/s)
3. USB 3.1 Gen 1: 5 Gbps (640 MB/s)
4. USB 3.1 Gen 2: 10 Gbps (1.25 GB/s)
5. USB 3.2: 20 Gbps

Spoiler: Choose your buffer size and sample rate

Choose your buffer size and sample rate

In this section we will introduce some basic terms such as audio latency, buffer size and sample rate and suggest recommendations for live use. Audio latency is simply the amount of time that passes between the sound being generated and then perceived by your brain. Basically, it is a delay.

For example, if you are 10ft away from the speakers, and since the speed of sound is approximately 1,000 ft/s in air it means that it takes 10 ft : 1000 ft/s = 0.1 seconds (or 10 milliseconds) forsound to travel from the speakers to your ears. The latency here is about 10ms.
Buffersize is basically the number of samples that will be collected before your audio plugins get to process them. Your audio interface is an analog-to-digital as well as digital-to-analog converter.

It takes any audio input, converts that into digital form (numbers) and then on the output side – converts those numbers back to analog audio. Sample rate determines how many samples your audio interface will capture every second and do the above-mentioned conversions. A common sampling frequency for live use is 44.1 KHz.

For example, if your buffersize is 256 and yoursampling rate is 44.1 KHz (44,100 times per second, as Hz means cycles per second) then your latency will be 256/44,100 seconds which is 0.0058 seconds or 5.8 ms. If your buffersize is 256 and the sample rate is 96 KHz you will get 256/96,000 = 2.7 ms latency.

You can experiment with this: If you change the buffer size to 128 and leave the sampling frequency at 44.1 KHz – your latency will be 2.9 ms and so on. These values directly affect the performance of your PC, as smaller latency values require your computer to respond more quickly to process all those samples in time without producing any glitches.

It has been shown that people can perceive differences between 3ms – 10 ms, and that our brain cannot distinguish anything below 3ms.

 

Again, you are just generalizing it, and that's simply wrong. Copy and paste information about USB standards doesn't make it right. It depends on AD/DA converter delay, audio driver buffer size and additional safety buffers or macOS' core audios "safety offset".

It's true that maybe you can go down to 16 samples buffer size with a PCIe connection and can't do this over USB, but everything below 32 or 64 doesn't make any sense to me as the CPU processing power for any audio processing loop running at such small sizes is going up rapidly.

The USB protocol does not generally introduce any latency, it only limits how low you can set buffer sizes and stay stable.

About bandwidth: 256 channels of 48 kHz 24 bit audio sum up to about 35 MB/s. RME showed 140 channels of I/O (so 280 channels total) stable over USB2:

https://rme-audio.de/rme-usb-technology.html

What do you want to show with these numbers? You set a PCI interface to 512 samples buffer size and a USB interface to 256? Doesn't make any sense.

 

Last edited: Oct 24, 2024

First of all, no beginner understands your technical jargon.

I just showed the topic starter @tommyzai that there can be no O latencies with a USB audio interface.
He was trying to reduce the latencies, so I described examples from other Audient iD14 MK2 users so that he knows roughly what the average latency is.

PC interface? And does that make sense:
A PCI card has a clear advantage over a USB audio interface. If you want low latencies, buy a PCI card.
My old PCI card can run up to 512 samples without any cracking or dropouts.

The Audient iD14 MK2 USB audio interface can only be operated without problems up to a maximum buffer size of 256 samples.
If I set it to 512 samples, it starts to crackle.

 

The user does not have an RME but an Audient iD14 MK2 USB audio interface!

 

Sorry, but you obviously lack basic knowledge and mix things up. If your interface starts to crackle if you set the buffer size above 256 samples, maybe it's a driver or compatibility problem. It's not about how high you can set the buffer size, but how low to get low latency, smaller buffer size means lower latency.

You assume your PCI card is "better", as it runs without problems at 512 samples buffer size and your USB interface is '"worse" because it crackles if you set it to 512 samples. And you assume that's a general thing about PCI(e) and USB and that makes no sense at all. It doesn't help the OP in any way, it's only confusing.

He doesn't have a PCIe card either, also he's using a Mac, so he would need an external thunderbolt enclosure to house a PCIe card, it's you who came up with the suggestion of a PCIe card to get lower latencies.

Audient or RME doesn't make a difference here, both are USB2. As I said, latency depends on AD/DA converter delay and buffers and the driver stability at low buffer sizes.

 

I have the Audient ID22 and an RME FireFace UCX II, and there's a noticeable difference in latency when using the same settings. The Audient interfaces are decent, but they have a similar latency to my old UAD Apollo Twin. The RME's latency it significantly better.

You're always going to be limited by the performance of audio interface. RME is the way to go if your budget allows it.

 

Read what he wrote again and what he is talking about.

Yes PCI always beats USB. I would also like an RME PCI card, they used to be the best, instead of the Audient iD14 MK2, RME is very expensive. Just because of the latency. At the time I bought a Firewire Mackie mixer because of the latency. The best that could be achieved with the Infrasonic Quartet PCI card is 512 samples. I was happy and satisfied and please don't assume that I'm too stupid. My PC is optimized for audio editing. The best that can be achieved with the Audient is 256 samples on my system.

I answered the OP's question. You should ask yourself what you are saying here. I also find your assumption that it would confuse him very presumptuous and arrogant. I wrote it in such a way that a beginner would understand it. End of discussion.

 

Yes, because the Fireface has very fast AD and DA converters with only 5 samples AD and 6 samples DA and uses 32 samples output buffer on Windows and 24 samples on macOS. On macOS you additionally get 24 samples core audio safety offset for in and out which can be reduced to 12 samples each with the "short safety offset" setting in the driver control panel.

With 64 samples buffer size in your Windows DAW, you get exactly 5 + 64 + 64 + 32 + 6 samples, that's 171 samples (3.9ms at 44.1kHz), on Mac it's 5 + 12 + 64 + 64 + 12 + 24 + 6 samples, that's 187 samples (4.2ms at 44.1kHz)

Audient, like most other manufacturers, don't tell you anything about these specs but it should still be fast enough for the average musician, if it's not one of the tightest professionals ever.

 

Audio Latency Explained
Understanding audio latency is crucial to your workflow. Below we explain latency and how it works.
https://audient.com/tutorial/audio-latency-explained/

Optimising Windows Computers For Audio
https://support.audient.com/hc/en-us/articles/202335209-Optimising-Windows-Computers-For-Audio

iD Driver V4New technology has allowed us to make significant improvements to our drivers, giving you lower latency at higher CPU loads. So what does that mean?

- Run bigger sessions at lower buffer sizes
- Add more plug-ins and instruments to your projects without increasing latency.
- Process audio in real time with minimal delay
- Increased stability on all systems
- 32 and 16 sample buffer sizes now available (best for highly optimised computers)

https://support.audient.com/hc/en-us/articles/360000919643-iD-Driver-V4-What-s-New

Audient iD14 MKII vs Audient iD24 which has the lowest latency input?
https://audiosex.pro/threads/audien...d24-which-has-the-lowest-latency-input.72645/

Audient iD4 MKII and iD14 MKII - Audio Interface Review

 

You still don't get it, it's not the best to get larger buffer sizes, it results in more latency. A buffer size of 512 samples results in at least 23.2ms RTL, plus driver buffers and AD/DA delay.

You tell the OP it's best to buy a PCI card, which at first is no easy option for his Mac, then you tell the PCI card is the best and superior to USB, because it can go up to 512 samples buffer size which is complete nonsense and results in an nearly unplayable RTL of over 23ms. It's lower buffer sizes the OP needs for less latency. That's confusing and I don't know what makes you call me arrogant, while I just pointed out that you mixed things up and insist on higher buffer sizes to be superior and can't be achieved with USB in general.

 

Maybe you should have a look at these links yourself.

The great thing about RME is that they give you every specification to calculate the resulting latency for your system, and that it spot-on.

Just had a look at the Audient website with info about the "new iD Driver V4"

Audient iD14 Windows using 96 kHz 24 bit audio:

- old V3 driver - 64 samples buffer - 5.938 ms
- new V4 driver - 64 samples buffer - 4.948 ms
- new V4 driver - 32 samples buffer - 4.625 ms
- new V4 driver - 16 samples buffer - 4.458 ms

RME Fireface UCX II Windows for comparison:

- calculated the same way as in the example a few posts above
- 171 samples - 64 samples buffer - 1.78 ms
- 299 samples - 128 samples buffer - 3.11 ms

And that is with 96 kHz already! Using 44.1 or 48 kHz would likely double it and gets you in the 10 ms range with very small buffer sizes and high CP'U load.

Conclusion:

You can run the RME interface with 128 samples buffer size and still get lower latency than you can ever get with the Audient interfaces even at the lowest buffer sizes.

The 16 and 32 samples buffer sizes are pretty useless and only stress your CPU very much. A difference of 0.5 ms makes no sense at all to use such small buffer sizes.

Audient don't give numbers for 44.1 or 48 kHz, but for real time software monitoring, getting in the range of 10 ms could be a problem for very tight and experienced musicians if there is any latency adding up due to plugins, monitoring DSP and so on.

I would recommend to record at 96 kHz with 64 buffers, 5 ms is a very good value, and 4.5 ms is an absolute inaudible improvement. But 32 or 16 samples buffer size result in very high CPU load and maybe even incompatibilty with some plugins.

 

Oh this is going to be an interesting discussion. I understand what you're saying, but if I can get 512 samples, I'll take them. Too low sample rates drive up the CPU. I would have liked 1024, but then the PC starts complaining. I didn't say he should buy a PCI, don't twist my words. He wanted latencies of 2 seconds, ideally close to zero. If he uses a PCI card, he'll almost get there. With the example values I've given, he should realize that he won't get 2 seconds or less. Instead, on average 5-7. It's so difficult to understand.

512 samples: I didn't say that it's best to use high rates, but rather the settings I use, because I can say that from my own experience. They always come with latencies. Musicians get by best with a value of 10 ms or less - this corresponds to the latency value that you find on MIDI devices between pressing a key on the keyboard and hearing the sound.

"RME's consistent low latency and zero CPU technology, both of which are not possible on USB, have made the products of the Hammerfall DSP series famous. This also applies to the RPM (as well as Multiface), which guarantees the best performance thanks to PCI technology using a PCI or CardBus card. RME's low latency drivers offer latencies down to 1.5 ms, support for ASIO 2.0, GSIF and MME (Wave), and thus excellent operation with all professional tools."



What buffer size and sample rate should I use?

TIP: Always test settings for buffer size beforehand along with any software and hardware system requirements to give you a better idea of how well your computer will perform with low buffer sizes and higher sample rates. This will keep you from running into issues while you’re in the middle of recording a project.

The most common buffer size settings you’ll find in a DAW are 32, 64, 128, 256, 512, and 1024. The most common audio sample rates are 44.1kHz or 48kHz. Sample rates of 88.2kHz, 96kHz, 176.4kHz, and 192kHz are also used, although these are frequently used with computers that have a lot of memory and processing power. So, if you have a computer that only has 8GB of RAM, then your computer may struggle recording at 88.2kHz sample rate and a buffer size of 64 samples.

DAWs and audio interface standalone software will often show you the current amount of latency based on the settings currently selected. However, if it doesn’t and you want to figure out the amount of latency at the current buffer size and sample rate, then divide the buffer size by the sample rate as mentioned above.

All that said, there’s no “industry standard” buffer size and sample rate, as it’s all dependent on your computer’s processing power. However, recording at 128 to 256 at a sample rate of 48kHz is acceptable for most home recording on modern-day computers.

The diagram below will show you the approximate latency at the most common buffer sizes and sample rates used in home studios. We say “approximate” because it’s dependent on the driver being used and the computer’s processing power.

Why does my DAW or interface software show a different amount of latency than what’s calculated?
Suppose you notice a discrepancy between the calculation and what is showing in your DAW or audio interface software. That is because the calculation doesn’t take into account that there are actually two buffers.


Approximate latency for common buffer sizes and sample rates

Source: https://www.sweetwater.com/sweetcare/articles/which-buffer-size-setting-should-i-use-in-my-daw/

 

That's a text published by RME at least 20 years ago. Anyone still knows what CardBus or GSIF was?

Times have changed and you can get an overall RTL of 1.78 ms at 96 kHz using 64 samples buffer size like in the example above with a RME Interface connected over USB 2.0.

 

It is not an accusation that they cannot stop doing it. Of course I have been dealing with it since I started making music (1998) and of course I only post links that I can support and understand myself.

Yes, RME can't be beaten - everyone wanted a Hammerfall back then, but they didn't have the money. Like you, if the user's money is no object, I would always recommend RME.

The Audient interface costs €205 and the RME Fireface UCX II €1,369. But the OP asked about Audient.

Dear Myfanwy, you have recognized and calculated this correctly.

Experienced musicians or professional musicians also spend more money. RME can shine with latencies and therefore highlight that in their advertising. Audient advertises what they do best!

Yes, that is a good recommendation and a nice conclusion. But remember that if a user has an Audient and wants to know how to reduce the latency, another much more expensive RME audio interface or sound card will not help him. It was a tiring but exciting conversation and I hope we will soon become friends and bury the insinuations.