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[quote author=”JesseG”][quote author=”timmywa”]Don, use the built-in jitter measurement tool to get your buffer #’s and size down to the lowest size while maintaining under 20% jitter. There’s some posts about this in another thread, I think up in a sticky post. Not sure at the moment.
You should be able to get it down to 2 buffers each, then just tweak the size until your jitter stabilizes around 10-20% or better.[/quote]
I don’t recommend doing this if you’re trying to get stable (at all) sound right now. Find out where the problem is, before tossing more wood on the fire. 😉[/quote]
Yes, listen to Jesse! He is wise.
Don, use the built-in jitter measurement tool to get your buffer #’s and size down to the lowest size while maintaining under 20% jitter. There’s some posts about this in another thread, I think up in a sticky post. Not sure at the moment.
You should be able to get it down to 2 buffers each, then just tweak the size until your jitter stabilizes around 10-20% or better.
First thing that pops out its the speed. Keep everything at 44100 khz. Check the main config file in edcast to make sure its set to 44100 as well.
Cuz, why not??
http://www.youtube.com/watch?v=P40TP1ughek[quote author=”Audio”]We will amend the jingles levels, but the dilemma we have now is to which level (?) since we have uncompressed musical tracks at all kinds at level as well.
Upon checking, our jingles are sitting at around 0 dB and so are some of our tracks (say Oasis) but we also have some tracks sitting closer to -10 dB (some Coldplay, Evanescence, etc.).
So which dB level would you recommend we normalize our complete library to? The reason I am asking is that I am unsure whether bringing up the levels on the -10dB tracks will hurt the audio quality of these tracks? To which level would you recommend we normalize our library to?
Thanks,
Audio[/quote]
Assuming your uncompressed tracks are .WAV, you can find a tool called wavegain ( http://www.rarewares.org/others.php ) and use it to bring all your audio to the same level. I’ve not used wavegain, but it’s partner, mp3gain has a default of -89db. Jesse recommended I lower that to -85db for what would be industry standard. If wavegain is the same, then I would recommend levels like I mentioned. You don’t need to be anywhere near 0db, as you wouldn’t have any headroom.
Hope that helps.
Someone correct me, but what he’s describing sounds like the processor’s AGC is lowering the output during the jingles (ducking) and once the next element comes on that is at a much lower volume, the AGC has to try and bring back the audio to the desired levels. If the processor’s ultimate goal is to provide a consistent sound and your track to track volume is so different, you’ll get that response on all but some of the fastest and loudest presets, like New York or Eruption, etc.
When I do live shows, I don’t use a mixer so I just amp my mic up quite a bit so that it ducks the music in the background. Sloppy? Yes. But with my available equipment, it does ok. I’ve had to experiment where to set the mic’s volume so that it doesn’t totally obliterate the music so that when I "hit the post" the processor doesn’t have to recover and the opening vocals are still drowned out.
Hope this helps.
In C:Program FilesEdcast there’s a number of CFG files (this is for stand-alone, it should be similar for plugin) Edit with notepad the edcaststandalone_0 file to turn logging to mode 1 for just errors. Do this for each of the Encoders labeled _1 _2 _3, etc… Default is mode 2, I believe and that logs now playing and speed changes (if you have a 64kbps stream and it varies to 63 or 65, it logs all that mess). Change that to 1 and you should be set. Have edcast closed while you make and save your changes.
Hope that is clear.
[quote author=”sebastien.wittebolle”]i think than a stream with lc-aac codec at 112kbps is a minimum for a cd quality.
You can stream with this version of edcast ; http://users.tpg.com.au/radiorio/[/quote]
I got this standalone version. I copied the aacencoder from winamp into the edcast folder. How does one use the aac+ High mode? Is that a different encoder? Also, would you use that with a 64bit aac+ stream? I have been thinking of going to just 48kbps aac+ but the old edcast would force it to use parametric stereo, this version lets me chose regular stereo…. might be nice that way.. What are your thoughts on this?
So along with this, kinda, YouTube has added a Vuvuzela button on their video player!! It’s that little soccer ball icon near the CC and size buttons below the video. Yes, you can ADD artificial Vuvu noise!! AWESOME!
[quote author=”jameskuzman”]
If you’re using the full setup, make sure you’re choosing "Breakaway Pipeline" as the input and your soundcard as the output. Kernel Streaming (KS) is the best choice, but if you’re having trouble with that (or it’s simply not an option) try Wave, or in a worst-case scenario, DS, and see if that makes a difference.
Jim[/quote]You’d put Wave above DS? Interesting. I guess Wave is a true source where DS is a contrived, simulated quasi-source… Someone better explain that? Jesse?
