How many msec is 170 samples at 96k?

Thanks. I can't remember how to calculate this.

 

EDIT: Delete

 

Last edited: Nov 7, 2024

Thanks so much!!!

 

This tool is wrong by a factor of 1000! It's 1,77083ms. Try yourself and enter 96000 samples for 96kHz SR, should be 1000ms, right?

Don't copy and paste everything you find in the interwebs, my friend.

 

Last edited: Nov 7, 2024

how many milliseconds is 170 samples at 96KHZ
ChatGPT said:

To convert samples to milliseconds, use the following formula:

Milliseconds=SamplesSample Rate×1000\text{Milliseconds} = \frac{\text{Samples}}{\text{Sample Rate}} \times 1000Milliseconds=Sample RateSamples×1000
For 170 samples at a 96 kHz sample rate:

Milliseconds=17096000×1000\text{Milliseconds} = \frac{170}{96000} \times 1000Milliseconds=96000170×1000
Calculating this:

Milliseconds=1.7708 ms (approximately)\text{Milliseconds} = 1.7708 \, \text{ms (approximately)}Milliseconds=1.7708ms (approximately)
So, 170 samples at 96 kHz is roughly 1.77 milliseconds.

 

Last edited: Nov 7, 2024

Wow, that's elementary school math. But needing an online calculator for this, copying and pasting a screenshot and rewriting the obviously wrong solution is just ridiculous.

 

Or simply break it down to 170/96.
96000 samples/sec -> 96 samples/msec

 

Milliseconds=1.7708 ms. but in Europe, being typed as Milliseconds=1,7708 ms

The "funny" thing is he is probably trying to add this as a latency offset, and Logic would just have you add it as 309 samples anyway.

 

Thank you for your help and correcting the facts.

 

Your comment does not exactly show strength of character to make fun of my mistake.
Am I the butt of a joke here? You are quite arrogant. Everyone makes mistakes...!

 

Sorry if you feel insulted. But in general, copying, pasting and repeating things without checking if they are true or even understandig them is the main cause of misinformation and fake news. So I apologize for naming it "ridiculous", I'd better call it dangerous.

Sure, everyone makes mistakes, but as an experienced audio guy and sound programmer you should instantly realize that 170 samples can never ever be enough information for 1.77 seconds of audio. It's like if I get the information that a standard apple weighs about 180 kg on a random website and just repeat that without thinking. I'd totally understand if someone would find that ridiculous and wouldn't call him arrogant.

In this case it is not dangerous, but imagine someone asking for a wire gauge for a given current and someone just copies a wrong value without checking it. Then the user doesn't check it by himself and causes his house to burn down.

Many of your posts are informative, but many are not, for example all the copied and pasted macOS tips beginning with "I'm not a Mac user, but" as if other people are unable to use google. Why should a non macOS user google things he never experienced himself and so has no clue if they are useful for other users? As far as I remember, you have been banned before, so my simple advice is think about what you are posting before posting it.

 

Let's give peace a chance, thank you for your sincere reply, I appreciate it. My goal has always been to help people, but every now and then, like with your Mac example, I didn't think carefully and followed my impulse. Sorry about that, I'm working on getting better.

Yes, that's right, last night I answered this strange question from our latency man in a simple way. I simply looked for an online calculator and found one and simply entered the sample number he said was 170 and 96 Khz and unfortunately didn't check the result and posted it here.

Then this morning I got a nasty surprise. Oh God, I fell right into the trap. People are already doubting my expertise. Sorry, sorry, sorry...! Thank you for your advice.

 

Thank you all for clarifying. I got it. I was clever enough to realize that Radio was simply off by decimal points. I'm grateful to him and the clarifiers. Side note . . . while continuing my endless latency research I summarized that 96k helps reduce lag. ;-)

 

I find it always helpful to calculate the distance sound travels in the corresponding time. Live sound engineers also like to describe delay in meters or inches. That gives a good real life impression.

For example your 1.77 msec equal in 60.7 cm or 23.9 inches. That's less than your ears are away from usual speaker placement, even near field.

I'm really interested if you have fixed the RTL offset problem with the Audient interface, as I know someone who is about to buy an ID24, and asked me about it.

 

In Reaper, I'm turning off interface reported latency and using the measurement generated by RTL Utility, cross-checked by Kenny Gioia's two-track loopback comparison. Both yield exact results, so I'm feeling good. Note, however, each time I restart, the measurement changes slightly. Yet, I'm close. I could also use the over-reported measurement by Audient, then use a negative off-set, but at present I'm going with the other. The Audient, in combination with my Mac Mini and DAW are hovering around 7ms at 48k and 5ms at 96k (before off-set), so I think latency won't be noticeable, and if it is . . . . I can always move the audio into place.

 

I don't see any problem at all and Audient is always recommended, price-performance ratio is very good.

Audient iD24 – USB Audio Interface Review

Hello @Myfanwy, here's a book recommendation, I read it and was amazed, he is Professor of Acoustic Engineering at the University of Salford where I carry out research and teaching focusing on architectural acoustics, signal processing and audio perception. I am also an author and radio broadcaster having presented many documentaries on BBC radio and written books for academics and the general public.

Trevor Cox - The Sound Book: The Science of the Sonic Wonders of the World
www.goodreads.com/book/show/22253763-the-sound-book

Acoustic Absorbers and Diffusers from Trevor Cox (PDF) Free Download
Theory, design and application
https://danylastchild07.wordpress.c...r_j-_cox_acoustic_absorbers_and_diffusers.pdf

 

My Audient over-reports. I could probably leave it un-off-set and it would be just fine, but while I'm at it . . . why not dial it in a little better.

 

Are you satisfied now or do you still have unanswered questions? At the beginning of our discussion we were also at 5ms!!!

 

Aah! Thank you, that's exactly what I wanted to know! I experienced the same thing with the Behringer UMC204HD, I have posted this in your other thread and gave some comments. It was quite a bit of latency jitter, worst numbers I got were 37 samples early and 34 samples late.

My RME interfaces are rock stable and deliver the same RTL at the same setting everytime and I have no need for any correction. Reported RTL and measured RTL are always exactly the same, so every recording or bouncing through hardware is sample accurate.

But you really only need this to be so accurate if you are doing things like parallel compression or routing one or more signals of a multi miced recording through hardware for example.

You have to decide for yourself if it's ok for you to have this little bit of unpredictable offset. But now I can finally understand your problem and it's origin, you simply can't dial in a constant value, because it changes everytime the audio driver is restarted, like with the Behringer interfaces. You can only hope that Audient fixes this in a driver update.

A relative simple workaround in Logic Pro is connecting a loopback cable and after opening Logic and setting your sample rate and buffer size, open the I/O plugin in one channel, select the connections and click the "Ping" button, it will report the offset and you can directly correct it for the I/O loop. If you want to get recordings to tracks to be accurate, you can dial in this same value in Logics Audio Settings as "Recording Delay". If somehow the driver is restarted, by changing buffer size or sample rate, you have to repeat this, because the value may have changed. I think this answers all your latency problems in the other threads, too.

 

Hello @tommyzai, jitter actually only occurs when you have multiple devices connected, which is why you make one device the master that sets the clock. I came across something that might be of interest to you. It's not the audio device that has faulty drivers, but other components of the PC (such as the graphics card, SSD, hard drive, CPU) that produce unpleasant electromagnetic interference and unwanted signal noise. I've attached the link, have fun researching. These signals can go through the USB cable.

Jitter and high-frequency interference – the PureClock is supposed to fix it
www.igorslab.de/en/what-chirps-you-with-an-external-usb-dac-and-an-oehlbach-pureclock-audio-practice/