![]() ![]() The gain and color of the direct sound can be adjusted. The four knobs on the right manipulate the different stages of the impulse response, starting with the earliest part: the direct sound. A demo of this is in the first minutes of the Altiverb 8 guided tour video. You should use it fully wet, but even then you hear quite some dry-ish material, because there is a direct path from speakers to microphones. This is the absolute most realistic sounding virtual playback of your audio in any space in Altiverb. You can also, right there, move the speaker around. Using the positioner tab, available only in mono and stereo input Altiverbs, you can switch on the speaker, in which case you can hear where it stands. So there is a direct relation between the amount of inputs and the speakers used. When making impulse responses, we used either one speaker position in the center of the stage for mono input, or two speaker positions for stereo input. ![]() Do not playback loud, to protect the people around you, and do not record loud to make sure you don’t clip the recording.Īltiverb can have mono or stereo inputs.all of these measures will increase the low end in a more complex and natural way than post-eq. If you feel the result sounds too thin, try a speaker with bass reflex ports on its front, or a bigger speaker, or change out directional microphones for omnidirectional microphones. Neutral is crucial if you want to reverberate dialogue and match it to something real. ![]() Few speakers when free standing like in the video, sound neutral on-axis at a couple of meters distance. ![]() You will find your speaker to be the most critical component.Decide if you would be satisfied if that were Altiverb. To test your setup on the location: play dry music through your speaker and compare the recording in the room to the dry recording.Carefully measure that everything is symmetrical, with large setups, a large error is easily made.Space the microphones far apart, several meters if possible.I haven't looked at present-day Linux for that but that's pretty much a lot to ask from a general-purpose O/S. The thing is that general software doctrine and "get every last cycle outta the thing" doctrine intersect only now and again and only uncomfortably.Įdit: For all that, the guarantee that must be met for an interface to be serviced 96000 over USB with a sample depth of 16 is 1/6th of a millisecond - no exceptions. It's all actually quite a fascinating subject but there's a lot of boring stuff in there. I'm using "latency under 1msec" as a very broad definition of "realtime". Now - all that being said, I can ( and have ) gotten realtime performance outta Linux kernels but it wasn't easy. The horror :) that was Windows Multimedia at least allowed getting the performance, latency and jitter needed for the old chips. Which is surprising but not all that surprising. I was on a project then and we tried to make RT Linux work and it just wouldn't. It wasn't bad but I guarantee you it was not the same. That meant "realtime priority threads" which was. 2004, 2006 WindRiver basically decided to put VxWorks on a "this is for charging the foo outta legacy defense contractors" basis and started touting "real time linux" which really meant "Linux with the realtime extensions". At the time of Linux' founding, RTOSes were RTOSes and Linux was Linux.Īround. I think architecturally Linux has an advantage over Windows when it comes to real-time audio processing. And stop believing ignorant claims online by people like OP who don't actually understand what they're talking about. Measure, test, see what works FOR YOUR PARTICULAR CASE. In fact for some PC configurations (it's certainly the case with mine) and some audio hardware (in my case a 2nd gen Scarlett 2i2) ASIO4All do that job better than vendor ASIO drivers. Just by working for a company creating audio hardware some group of programmers are super great at implementing ASIO drivers isn't really how it works, unfortunately. Your audio interface provided ASIO drivers actually themselves do a very, very similar thing - as no audio hardware is made with one particular API (be it ASIO, CoreAudio, ALSA or whatever) in mind. The WDM bits ASIO4All wraps expose those buffers. You obviously have no clue what a virtual machine is, do you?ĪSIO4ALL is wrapping parts of a kernel WDM module, which in itself already has all it takes to be a low-latency driver (really not much worse than, say, CoreAudio on Mac is) except Windows themselves don't utilise them as such as low-latency audio is utterly unimportant to Microsoft.ĪSIO is Steinbergs API to interface low-latency audio buffers in audio hardware. ![]()
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