Should I be using the minimum phase filter for gaming for faster response time?
All my other gear is built for speed but the linear sharp filter seems like it is better for the sound, but bad for delay in response time of the sound.
44.1 kHz 96 kHz 192 kHz
SD Sharp 6.25 5.63 5.63
SD Slow 5.3 4.68 4.68
Sharp 29.4 28.8 28.8
Slow 6.63 6.0 6.0
NOS --- --- ---
44.1 kHz 96 kHz 192 kHz
SD Sharp 14 13 19
SD Slow 12.4 12.3 18.2
Sharp 36.5 36.6 42.2
Slow 13.8 13.8 19.7
NOS 8.4 8.7 14.7
44.1 kHz 96 kHz 192 kHz
SD Sharp 0.3 0.14 0.099
SD Slow 0.28 0.13 0.094
Sharp 0.38 0.38 0.22
Slow 0.3 0.3 0.1
NOS 0.19 0.14 0.077
I have tested .15 to visually react and press a button. I've gotten faster, but that's pretty much my average.
https://www.humanbenchmark.com
try it out.
My monitor is 1ms for real though. Real world I think i do get 3ms in overwatch though. So to get your monitor to run 1ms real world isn't possible.
Assumiing multiplayer - add network delays...
Pinging audiosciencereview.com [104.27.182.13] with 32 bytes of data:
Reply from 104.27.182.13: bytes=32 time=31ms TTL=51
Reply from 104.27.182.13: bytes=32 time=31ms TTL=51
Reply from 104.27.182.13: bytes=32 time=30ms TTL=51
Reply from 104.27.182.13: bytes=32 time=48ms TTL=51
It would take me at least 31 and as much as 48 ms per above to post a goofy reply to someone's post in this game.
Awesome that you actually tested this all, was just wondering if the lone 13.8 there is correct ?44.1 kHz 96 kHz 192 kHz
SD Sharp 0.3 0.14 0.099
SD Slow 0.28 0.13 0.094
Sharp 0.83 0.38 0.22
Slow 0.3 13.8 0.1
NOS 0.19 0.14 0.077
5ms is around the time people can start to feel/ not feel a delay. This number is coming from the guitarists that were using digital processing effects, musicians using monitoring with effects plugins, and many people who work in pro audio scene. It generally holds true for gaming as well.
You can use "code" from in the "..." pull-down menu in the editor.Sorry for the bad formatting. Tab doesn't work and multiple spaces are automatically reduced to one. How to get the above look like a readable table?
44.1 kHz 96 kHz 192 kHz
SD Sharp 6.25 5.63 5.63
SD Slow 5.3 4.68 4.68
Sharp 29.4 28.8 28.8
Slow 6.63 6.0 6.0
NOS --- --- ---
I once had a worst-case live event: I (bass player) had no monitor of the drums and had to rely on what I heard from the drum-set which was some 5 meters away (15ms) but the drummer had my direct signal on his monitors. He constantly thought I was trying to slow him down because I played way behind the beat, given the late arrival. We managed through the gig... making lot's of funny faces to communicate...You said it first I know two guitarists that can't stand overdubbing with more that 5ms latency.
I have tested .15 to visually react and press a button. I've gotten faster, but that's pretty much my average.
https://www.humanbenchmark.com
try it out.
Thanks @MC_RME. As always, your posts are extremely valuable here.It doesn't matter - I will explain. Let's start with a table of the delays caused by the various filters of the ADI-2 DAC, as stated in the AK4490 data sheet:
Code:44.1 kHz 96 kHz 192 kHz SD Sharp 6.25 5.63 5.63 SD Slow 5.3 4.68 4.68 Sharp 29.4 28.8 28.8 Slow 6.63 6.0 6.0 NOS --- --- ---
Remember that NOS is Super-Slow, and there is no value stated in the data sheet. Well, doesn't matter as these values are in samples (!). The worst case is the Sharp filter with 29.4 samples. At 44.1 kHz that equals 0.83 ms. Yes, less than a millisecond. If you use the (also phase linear) Slow you are at 0.3 ms. The statement 'significant delay with a linear filter compared to minimum' is true for typical EQs in the low frequency range, but not always for higher frequency aliasing filters.
But the main point is - these are just the DA filters. You have to add:
- buffers within the game for audio output
- buffers in the audio subsystem (WDM/ASIO etc)
- buffers in the audio subsystem processing (if not bypassing the kernel mixer)
- hardware buffers in PCIe, USB etc.
- hidden buffers in the driver
- buffers in the audio hardware
- processing the data within the audio hardware
That is everything from 1 sample up to a thousand samples, per item! The total amount for pure playback easily reaches 100 samples even in best case.
Measuring the latency from USB event to analog output is quite complicated - and somehow meaningless, because whatever tool you write will not represent how another program handles this. Using ASIO it gets more easy - but Games don't use that. So let's look at something that is easy to measure: the time between the arrival of the digital data at the SPDIF input until an analog output signal shows up. Doing this with the ADI-2 DAC we get (again in samples):
Code:44.1 kHz 96 kHz 192 kHz SD Sharp 14 13 19 SD Slow 12.4 12.3 18.2 Sharp 36.5 36.6 42.2 Slow 13.8 13.8 19.7 NOS 8.4 8.7 14.7
Compare these values to the pure digital filter values and you will see they are naturally higher. The SPDIF receiver (1 sample) plus internal data processing (6 samples at single speed) are added. And as we measure we now also get values for the Super-Slow (NOS) mode. Here is the above in milliseconds:
Code:44.1 kHz 96 kHz 192 kHz SD Sharp 0.3 0.14 0.099 SD Slow 0.28 0.13 0.094 Sharp 0.38 0.38 0.22 Slow 0.3 0.3 0.1 NOS 0.19 0.14 0.077
But all these numbers have to be taken in context. As mentioned by others the overall latency will be around 5 to 6 ms - if you are lucky. So whatever filter you choose you won't notice any difference. In the ADI-2 DAC's Settings dialog you can choose a buffer size down to 32 samples. That setting is also valid for WDM and your Games, which typically work best at 128 samples.