Hi guys,
i recently had a bad experience, and maybe one or another may eventually run into similar problems.
So this is not really a question but rather some hint or advice to others.
You know, i mostly play at home, and when i do, i have all my gear built up in my “man cave”.
Because i also listen to music, play games and watch films in this room, i purchased a decent Audio-Video-Receiver (DENON X3500H with pre-outs to make use of my active speakers) and some additional speakers to get some serious surround sound - cool for films and gaming!
Before of that i only had a pair of nearfield-monitors with double subwoofer support - so stereo sound was perfect. I also had a PEQ and active crossover device for this.
Now that i have the new AVR, stereo sound is still excellent, but because of the internal soundprocessing i experienced quite a heavy audio lag which i could not get rid of with any ASIO adjustments.
As long as i play keyboard, this lag is fairly tolerable but when i play guitar the latency that is produced by the AVR is more than noticeable!
Of course the AVR has a “Pure Direct” mode where all the sound processing is bypassed.
Activating this mode removes the audio lag, but in my case it also deactivates the subwoofer support, which then results in a very thin sound. There is also a so called “Game mode” where the sound processing should be reduced to a minimum, but there is hardly some diffrence.
My solution to this will be a small switch box (symmetrical!) which lets me change the input for the nearfield monitors (AVR or mixer) while i additionally connect the subs (each of them has two inputs L/R) via the active crossover.
So i hope this will solve my problems then.
In the meantime (until i will receive the input switch) i will use the “Pure Direct” mode and play with a thin sound.
Sure i did. But since there only is the option for an additional delay for audio to video, this seems to be quite useless. I’d rather need a contrary function… some kind of audio forecast, but i’m afraid this doesn’t exist.
I have a very similar configuration with a very old Yamaha AVR. I even use the stereo to 5.1 built-in conversion without noticeable additional latency.
Did you try to measure the additional latency of the AVR ?
Nope, didn’t try that yet… but i can clearly hear that a lag is there and that there also is a remarkable diffrence between processed “Stereo” and “Pure Direct”. So i dont actually want to put any energy or effort into really measuring how long it is - although i’m a bit curious too…
I understand, the question was rather, is it a small latency that adds to another small latency (if you audio interface) or is it really a huge one. If it is not too high, I thought you could perhaps shorten it by avoiding a DA and AD conversion by going digitally from your audio interface to your AVR. Your AVR has probably an SPDIF input, if you audio interface has one too (mine has one), it is perhaps worth the try… But well, I am not sure it saves a lot… I don’t know much about the protocol behind the SPDIF.
That’s something i could actually try… i don’t have my audio interface connected to the AVR directly since i use a mixer for the instruments, but both, the mixer and the interface, lack a digital output anyway.
But i still have a Behringer DEQ2496 from my former configuration and this device does have an optical output and also the neccessary routing options… maybe i will try this tomorrow.
Let us know if you try it and if it brings something. But, anyway, thank you for pointing out this issue. I will take care when replacing my old Yamaha AVR by a newer one with HDMI connectors
Meanwhile i changed my audio interface… i ditched my Steinberg and purchased a Motu M2.
Now i can play in AVR’s “Game Mode @stereo” without noticeable latency.
I am not really sure… those two interfaces (or their drivers) behaved completely diffrent regarding the settings.
With the Steinberg i could set ridiculous values for buffer size without having problems or noticing massive changes between, say 256 or 32 samples… my standard setting with the Steinberg was 44.1kHz and 128 samples. When i changed over to the Motu i kept those settings and it sounded so terrible that i first thought it was broken, but increasing the buffer size to 256 caused the sound to turn great, and also the latency was absolutely OK. So while i could play with the buffer sizes on the Steinberg within a huge range without having sound issues, it is just one step between heaven and hell on the Motu… strange things going on.
Maybe the driver installation of the Steinberg was faulty (most likely)… i had to re-instal them a few times after updating Windows when they refused to work correctly (i.e. with Cubase).
