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[quote author=”Guillou”]You said, you had tested it with modern music.
What’s modern music for you : dance ? pop ? rock… ?[/quote]Yes.[quote author=”Zeb”]Great Stuff Jesse!
Would you use it for Rock?
Which band(s) would you recommend one emphasizes for Rock?
Thanks!
Zeb[/quote]
Personally I wouldn’t try to get any more stereo out of any of the bands from where I have it set. I tried to get the MAX stereo effect, without effecting some other things badly that I talked about above, like voice intelligibility. It won’t be flawless, there’s bound to be 2-3 tracks it makes a tad less intelligible but so far in my testing the settings seem to be pretty safe actually, despite the huge sound.
All respect to admin aside… Ots has ALWAYS had issues with supporting certain soundcards that have ZERO problems with any other software that I’ve ever heard of.
Blame based on possibility (not even probability) is not a solution in my book.
That aside, there could be more at play than just Ots, Breakaway, and your drivers. To just assume that to be the case, on the part of admin, is perhaps more of a delusion than just assuming your drivers must be crap. 😉
[quote author=”Bryan Durio”]For the record, I’m checking out the v0.90.04 alpha version, and it uninstalled BAE 1.20.26. Haven’t gotten it to do ASIO yet.[/quote]Are you sure you’re not confusing BAE with the Breakaway Pipeline?
yeah AKG K240DF are quite good cans. Also great…
AKG K81 DJ cans ($50 cans that blow away $1000 cans like Denon AH-7000 for instance)
Beyerdynamic DT-880 Pro
and of course… Sennheiser HD-600/650all of those last 4 are very similar sound. I prefer and use the DT-880 Pros because they sound like Senn HD-650s with the bass of Senn HD-800s without the sticker price of either.
8)
With Ando Media’s acquisition of Spatial Audio, I’m really doubting there’s going to be another update to EdCast. Certainly not the complete rebuild it needs. It would certainly be a conflict of interest between Ed (who is a VP at Ando), and Spacial’s business. Don’tcha know.
And not all encoders are made equal. In fact… most suck. Great example, iTunes implementation of FhG F.A.S.T. Even at 320kbps it sounds worse than Lame at 128kbps.
[quote author=”George”]Did you try SOS4 (Spacialaudio)?
Testing SOS4 last week seems to be very stable (the latest version fix couple of problems)
and supports LAME and AAC+ v1/v2.[/quote]
Is there an option to disable the built in audio processing? Cos that would almost for sure tend to ruin Breakaway’s flawless sound.And to the people "begging" for Leifcast, Leif already explained that the issue is a business one, of licensing. So there’s really no point in asking for it to be released, unless you wanna buy Leif an unlimited software license for the codecs involved. I’m SURE he wouldn’t argue with that. 😉
[quote author=”sneradio”]Stop by my tune-in page for a listen.[/quote]
Love it. Super clean, dynamic, sounds like no processing at all. And yet no volume knob reaching. Isn’t this what processing is supposed to do? 8)I didn't really have one, so I decided to make one.
Very wide sounding, lots of air, but also doesn't "blur" the vocals at all. Sometimes makes the vocals even more upfront. Try this on some modern music. <!– s8) –>8)<!– s8) –>
Personally for net radio, I would suggest Live every time. Even without any clipping on decoding of a 128kbps (or+) mp3 stream… you will still be louder than almost everyone else, with a nice sound from one of the louder presets.
Or you can pick a more open preset, and get lots more TSL. 😉
[quote author=”sgeirk”]This setting could be fine for the internet, who knows? But for FM broadcast, one really needs to be aware of what effects overdoing it has on mono compatibility. You just can’t overlook what it might sound like in a partial or full blending situation, much less flat-out mono.[/quote]
Absolutely, and this goes for net-radio too, as most lossy codecs that have joint-stereo and/or inter-channel masking of some kind, get really "annoyed" by stuff like this, and throw bits into trying to keep the stereo image sounding like it should, instead of things like making speech sound like speech, and keeping Kenny G from sounding completely like kazoos getting molested.
At any rate… there’s also creativity concerns. Most music (with vocals) has the important lead vocals in the center, mono or near mono. If you widen the music more than it should be, especially in the vocal "meat" and presence ranges, then you will end up making the vocals harder to hear, and less ineligible. For a radio station, that seems like a really important thing, doesn’t it?
Interesting, me and Leif were talking about some of this last night/yesterday/earlier/whatever. 🙂
[quote author=”Benny”]2. In Live: limiter or soft-clipper with variable clipping.[/quote]
Basically to add distortion. There’s plenty of effects out there to saturate your audio. The one thing that would be really nice, and I’ve suggested it to Leif before, is to have the option to have the input Winamp plugins AFTER the AGC/WB1. This would allow for at least a basic wideband gain control which would help out plugins that are not so program dependent, like a saturation effect would be.[quote author=”Benny”]3. In Broadcast: distortion-canceled clipper with distortion controller and variable clipping.[/quote]
The hardware version will have more control over the clipping, in addition to more advanced clipping, and of course done directly on the MPX with Leif’s killer headroom trick to give you 4-5 db more space in the treble. It’s really amazing how it sounds even in comparison to how awesome the $199 "BBP" already sounds. Which is partly why the hardware unit isn’t going to be $199 + cost of hardware. 😉 Making Leif a happy boy is the other part. 8) 8) 8)Happy New Year, and here’s to the future.
[quote author=”bamp”]May I ask something too ?
If you have enough bandwidth (like me) is it wise to use OGG FLAC ?Also if I finally choose ogg vorbis what is the best settings in edcast (Quality setting or bitrate) ?[/quote]
OggFLAC is very stable over a good network connection. That’s what Slim Server uses, for all of the Computer/Network -> Slim Devices connection of audio.
for Vorbis, quality based setting is better in general, but for streaming you generally will want to have a predictable bitrate with a codec that has longer frames. Of course you could always see how a quality-based setting works on your connection & software used.
[quote author=”Lee XS”]OK nice one, I’m going to give the aoTuV Vorbis 5.7 a try at 128 kbps and see how it goes…..
What do I have to do (or not do)or download differently from Leifs settings on this thread to get this going? any info or weblinks would be great.[/quote]
just EdCast and preferably IceCast v2 for Tx, and a vorbis codec for EdCast… which you can get here:
http://www.rarewares.orgOptionally you can try FAAC with the LD profile, short for "Low Delay"… which works in EdCast. But FAAC developers admit it’s not the best tuned AAC codec out there, and you also would have to optimize the Tx and Rx buffering as well.
What is also a viable option, if you have some skills, is to use the Linux version of Nero’s free AAC codec to hook into a low delay streaming app via std-in/std-out… Nero’s codec supports the LD profile.
There’s also Nullsoft’s transcoder for Linux with CT’s codec, but I’m not sure if you can input directly from a soundcard’s input via std-in or what. If it accepts RIFF WAV format on the input, then that might be an option too, if you configure the shoutcast server to have very low Tx delay, and use a very small buffer on the Rx. However… I’m not sure it even supports the LD profile. I think the profile might be based on presets picked via the desired bitrate, same as Nullsoft’s Windows build of the codec.
Food for thought if you ever want to go with LD-AAC.
