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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?
[quote author=”wouter”][quote author=”JesseG”]Yes. You can use the best mp3 codec there is… Lame 3.93.1
It also does Vorbis and FLAC in an Ogg container. LC-AAC using FAAC. Also AAC in LC, HE, and PS profiles using Coding Technologies codec that is distributed with Winamp which is easily the best.
It will stream to Shoutcast or Icecast, or a compatible server like Steamcast. 🙂[/quote]
hmm sounds much better maby simple cast is f***** up my sound because straight away from breakaway to my speakers it sound good and at the stream it sounds bit crap as you all said
have you may got a link to that streaming program??[/quote]
From a post or two above:
Can’t comment on the music. It looks like your stereo balance is a little strong to the left. Unless it was just that track. I’ll keep looking. Why do you cut off the high-end? It looks like you’re even lower than 16khz.
Marious,
Love the look and feel of RadioDJ! I’m in the process of importing all my music into the db now. (lengthy deal!) I will then categorize and create some clocks. So far, it looks daunting. If you have quickstart info, that would rock!
Can I suggest something on the audio side? I’d like to see adjustable ReplayGain support (as in latest Winamp). I use ReplayGain on all my music and imaging. And if you can support 24-bit playback at 44,100khz, that would rock too. (I know Breakaway doesn’t yet, but soon, I imagine)
I’d like to see some sort of collaboration with RadioDJ and Breakaway… not sure how that would work, but maybe it will trigger some ideas.
Is that latest build available on your website?
Thanks!
Yay! New presets!! I request Rustonium, Passive Aggressor (if avail), Motor City and new Zenith.
You can take back Eruption, EWW!
Leif,
One further question… On Audio Repeater, the goal is to get below 10% jitter at 2 buffers. I think this goal is different in BAE… 2 buffers may be a bit low for solid audio… My question is if I get the lowest jitter through repeater at 2 buffers, 1024 size, do I actually use 2 buffers at 1024 inside BAE? or should I use 4 buffers at 512, or some other combination?
Or is my logic all messed up and the number of buffers should be the same inside BAE as it is in repeater?
Side question, when I tuned it with repeater, the input on pipeline was pretty high, maybe 1024, it seemed odd that a virtual device would have to have such a large buffer size. I could see an old audio card needing it as the hardware is crap, but this isn’t hardware at all.
I hope all this makes sense. Thanks!
