Stereo Tool

Hi!
I am having a RME Hammerfall DSP Multiface and would like to run the signal from analogue in through stereotool and then stream its output through edcast.
Now edcast seems to just let me grab audio of a physical input but not the systemsound or any physical out.
So obviously I could just route the processed audio to a physical output and patch to a physical input where I grab it with edcast but is there any way to do it internally?
I am a macuser where the os would allow to create a virtual sound device which allows very flexible routing.
I am also a bit confuased by the asio options. Is there any benefit of using asio? I still have a delay of 10 seconds or so but havent checked the routing completely.
Help is greatly appreciated

Cheers,
Jens

You can use VAC (paid) or VB Cable (donationware) to route audio via a 'virtual sound card'. So that solves one issue.

10 seconds is insane, Stereo Tool itself requires about 120 ms (at maximum quality, you can go as low as 17 ms but it will sound worse). Unless it's the streaming delay, that's pretty normal (big buffer to handle internet package drops and re-requests etc.).

Non-ASIO will give you about 300 (!) ms extra latency.

As Hans said.. VAC is solution for ST output and so EdCast Input. VAC is not free but you can try this one http://vb-audio.pagesperso-orange.fr/Cable/
Here is our ST output and buffer http://i.imgur.com/JiOrw8E.png
Also be sure that Sync is ON http://i.imgur.com/qg5k551.png

Are you planning to use mp3 for stream?

thanks for your replies.
how relieable is virtual stereo cable?
the setup worked very well yesterday but i had a delay with the stream of 30 sec or so.
probably due to the streaming.
so my best bet wouldtven be to choose asio inputs.
i didnt get the whole buffersize thing. the larger the buffersize the more precise the lookahead function but the buffersize purely relates to st and not my soundcard or am i wrong with this. so i basically habe to add the stereotool latency to my soundcard latency (times two as far as i know it just shows the delay caused by one way (in or out)).
hmm so i could route my input with 0 latency through to monitor on out 1&2 have a st processed sound on out 3&4 and route virtually for streaming theough virtual audio cable.
what exactly is asio override for? it seems i can route to an asio as well as a nonasio device at the same time or why can i select different devices? it also just shows 8 asio outs which probably represent the outputs of my soundcard but where are the adat channels and the spdif out?

thanks for helping

jens

edit: Is the io configuration saved within the preset as well so that i could have st in autostart and my routing configured automatically?

It's reliable, as long as you keep the Synchronize turned on in Stereo Tool as Bojcha pointed out.

30 seconds is just an insane amount of latency - the difference between ASIO and non-ASIO is only 0.3 seconds.

Buffer size is really that, the size of the sound card buffer. Useful if your PC is also used for other things - if it has no time to process audio it won't immediately start to hiccup. No effect on audio or anything.

The ASIO numbers are numbers in your sound card, I don't know which is which (but since the latency is not caused by not using ASIO anyway that's probably not very important now); *ONLY* if you choose an ASIO override you will use ASIO (the buffer display turns green instead of blue). If it's blue you're not using ASIO. There can only be one ASIO sound card in an application, and the list of ASIO sound cards differs from normal list of sound cards.

Upon restart you get the settings that you had when closing ST. If you need multiple ST's with different settings, make a copy of the Stereo Tool binary (.dll or .exe file) and the settings will be stored in a different location, so that way you can have multiple setups that are loaded on startup.

Total latency = input sound card latency (usually 0) + processing latency (12-93 ms depending on Latency setting in Stereo Tool) + output buffer latency. For non-ASIO you have to add about 100-200 ms to this (depending on which Windows version you use) and you need to set the output buffer higher to avoid hiccups.

Maybe He thinks on stream latency

Thanks for your help!

I have two more questions.
Do you have any experience with using a generic asio driver on virtual winxp (or any other ms system stereotool will run on) maschine under parallels? I have a spare macbook in my practice which I need to run os x for my practice software but it would be awesome to be able to use this to process and stream. I know that there is a os x version of stereotool but as long as it is experimanetal status i would probably prefer to go with something 100% relieable or would you consider the os x version more stable than a windows version under parallels?

I saw the "packs" for the different modules on your website. Can you also license them piece by piece?
The software sounds much better than I expected however I am absolutely uncertain what is relieable for what. I know how a compressor and limiter works but how the declipper or a bassboost affect the sound is something i still have to find out.
I guess once the asio configuration is working fine I will be able to as changes are audible almost immediately.

Ohh and a last question. Does the io configuration get saved in the preset so I could basically configure a whole streaming system which is working straight away when turned on? Is hd speed important? I run samsung 830 ssd's in my laptops but would much more prefer to have the whole system on a stick so I dont need to have a "clean" system on each laptop which takes away ssd space.