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Wow the new site design is even worse than the last redesign.
At least what they should do is make the server list way more compact and have like 30 stations per "show more".
If you’re running two instances of Breakaway at the same time, you need two instance licenses… if they are on the same or different machines it doesn’t matter.
[quote author=”vpnmaster”]Nice, let’s wait for JesseG answer.[/quote]
I don’t have the time to check out Raduga. Sorry.[quote author=”uncontrollednoize”]I’ve been trying all versions of ‘live’ from .69 up to the latest version… for some reason they all sound different.[/quote]
Have you tried blind ABXing the different versions to see if it’s placebo?[quote author=”uncontrollednoize”]I’ve been bouncing between 44 & 48K on the Breakaway Pipeline and love the clarity @ 48K, but have been getting distortion on the low end and ‘sizzle’ on the high end…[/quote]
So the distortion that’s getting added is perceived as more clarity? 😉 Maybe it’s compensating for non-linearity in your playback system and/or room.[quote author=”uncontrollednoize”]Now, I am hopeful that ‘Passive Aggressor’ may help since it is a more ‘open’ preset, but in the mean time, does anyone have any ideas or preset settings that can help?[/quote]
Try Rustonium (with the gain up if you want, it’s made to be as "quite" as Reference on purpose and for "BDJ") in the mean time if you like a processed sound, or Zenith if you don’t like a processed sound. Also check out Reference Hard. 🙂In some ways "PA" sounds more open than even Zenith, and in some ways has more of a longer term gain riding like Jill FM except without the pumping. So it may or probably won’t be a sound you will like.
Finally… maybe even just try Jill FM, with the bass up a little if you want. It’s a very solid sound that is almost as loud as New York, without any of the extreme multiband slam sound, instead a bit of "pump".
Ok, had a few hours to really get the feel of the app. Very cool so far, and shows potential to become one of the best out there, not too far off.
That being said… the mp3s are decoded at full scale, so there’s no headroom to allow for the lack of peak control, and hence mp3s will clip. Also… during cross-fades even with lossless formats, you can get clipping.
I didn’t check to see if the Winamp DSP stack is compiled against the 24/32 bit DSP abilities of the new SDK, since I don’t know that I have any Winamp plugins off hand that use that. I probably could check, but yeah.
Other than that maybe/not being a workable option…
So my recommendation either way is to the developer. To decode mp3 with some decent headroom, or if the audio path is floating point… to allow for the final output to be turned down some in a way that can be used to avoid any clipping at the final outputs.
[quote author=”Audio”]- Apply Replaygain, though they offer two desired volume settings for "active" and "inactive" ReplayGain: between 0 and -40dB??? what settings would you recommend here?[/quote]
I’m not sure that will do what WavGain does. WavGain already processes internally at 32bit float, and optionally dithers back to the output bit depth (same as input file by default).
Check here, abotu half-way down the page
http://www.rarewares.org/others.php
and start by reading the manual. =)I just downloaded v0.1.5.3. Going to do some testing. 8)
[edit]
and now downloading MySQL, hehe. been a while since this machine was used for local web development.
[/edit]Wow, I’m hearing some really weird distortion and it’s almost like there’s some signal that’s bleeding into the sound too or something. It sounds like something in a frying pan. There’s also clipping, but that could be in the original…
Next song, not hearing the clipping distortion or bleed.
A few songs later, hearing a lot more of that crunchy frying pan clipping, but no bleed.
Something definitely doesn’t have enough headroom. What automation are you using? Are you running the input into EdCast at about -1.0dB peaks to leave a little headroom for the codecs decoding?
[quote author=”timmywa”]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.[/quote]
That’s basically where I was headed with my next recommendation, so thanks for suggesting it. 🙂
the industry standard is 83 dB SPL, and ReplayGain does a decent enough job in getting those correct in relation to the original CD audio standard… but the default ReplayGain level is 89dB I’m assuming because they valued decreasing the dynamic range of hyper-processed stuff (which sort of seems like a battle already lost) to having enough headroom for stuff that already sounds great and in some cases gets turned up even louder causing the peaks to require limiting. :/
i recommended 85 dB because it’s between the two, but more on the side of 83dB. Tim, what percentage of songs would you say actually get turned up at 85dB?
Another benefit of what you’re doing here is that crossfades and mixing of things together you’ll generally already have decent enough headroom where nothing will have to limit the signal to prevent clipping. (or it just plain won’t be clipping anymore) For the most part. 🙂 The result of that is a cleaner more dynamic sound. Like when the jocks are talking. (unfortunately it won’t clean up the language, lol)
[quote author=”Audio”]Since we are going to be normalizing everything to the same level, what would you say would be the ideal level to use ahead of BBP?[/quote]
The peak level tells you nothing about the actual loudness though, and it will encourage a "loudness skirmish" within your own plant and production studio.How about getting the Breakaway RTA (check latest software topic) and using dB LKFS to set where your actual loudness should be at. -23 dB LKFS is pretty close (within 1dB) to where the EBU P/LOUD group’s R128 draft is having the new international standard for where reference loudness at. The idea for TV broadcasters at least is that programs can be mixed at that loudness level, and put straight onto the air without any audio processing at all. 🙂
But even if someone will be processing for loudness at the end, the idea itself still benefits everything before then greatly. For instance, one could reduce the range of the AGC/Multiband so that stuff that’s more quiet stays more quiet.
Anyways yeah… Breakaway RTA, check it out. 🙂 Mainly for the level meter. You can resize it skinny so that’s all that shows, if you want.
[quote author=”Audio”]you mentioned converting everything to 32 bits ahead of the normalization, which software would you recommend for us to do so?[/quote]
Whatever software you’re using to normalize aught to be able to do this (including a decent dither back to 24 or 16 bits depending on what your playout software is compatible with), or it’s probably not very trust worthy software to begin with.So wait though… you didn’t say how you deduced that it’s AVG doing this. Did you shut down AVG and that fixed it? or what?
[quote author=”Audio”]One option would be lower the jingles level, but how do I go about doing this without messing up the quality?[/quote]
That’s my thought. To do it with high quality… would be to convert it to 24 or 32 bits if it’s 16 bits right now… lower the levels… then dither back to 24 or 16 bit if you have to. Minimum quality lost, and most likely an inaudible change. Other than the music being at the same loudness again. 🙂
Cheers,
-J[quote author=”timmywa”]Dude, Jesse…. You should be arrested for teasing!!! Where’s that pesky download link???[/quote]
It wouldn’t be of any use to you anyways… since you can’t change the stream location of it for instance, and it’s protected by the same stuff Breakaway is. 😉 It’s just one application of the stuff I worked on that’s behind the scenes for it. It’s loading winamp plugins too, which aren’t in that screenshot.I might eventually make a multi stream version of it. We’ll see. It’s not very high on my list of priorities until whatever lies beneath the next 1.5 years starts to unfold a little more… and then maybe we’ll see.
[quote author=”calavan888″]Thank you guys I will try as you said. Does this cause edcast to disconnect at random?[/quote]
EdCast just plain doesn’t have very robust netcode at all. If your connection lags even for a split second, that could cause the dsp pipeline (not to be confused with Breakaway Pipeline VACs) to hickup and I’m not too sure how Breakaway would handle that but I wouldn’t be surprised to know that EdCast doesn’t properly handle that at all since it’s causing the dsp pipeline to do that on purpose… in an attempt to prevent host-side buffering when playing regular audio files in Winamp.I know Leif’s netcode was a tad more robust in his encoder, and I built one that allows for up to 10 seconds (or whatever you want, I’ve done 30 seconds before) of buffering and as long as the networking output buffer catches up… the listener won’t hear any skips or buffering. It even has a super sexy "Lag-O-Meter" as the colored background of the connection button/s which is aware of the network device connection state and buffering of the network stack itself… which shows you how much of the network output buffer hasn’t been able to be sent yet.
I haven’t shown this in public before, but check it out 🙂
about 9 months old now. my customized LU perceptive loudness meter, and on the right the connection button with Lag-O-Meter background. 🙂 and I had a ton of FTP uploads going to force a bit of lag.
