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Viewing 15 posts - 751 through 765 (of 1,890 total)
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  • in reply to: breakaway broadcast asio? #8296
    Leif
    Keymaster

    On a machine on which a friend swore you couldn’t install XP due to the hard drive controller, I was able to boot XP Embedded from a USB stick, skipping the hard drive altogether.

    There’s always a way. 🙂

    ///Leif

    in reply to: BBP / BBP ASIO / Live 0.90.80 with Stereo Enhancer #7876
    Leif
    Keymaster

    Wow that’s weird!

    Never ever heard of that. That seems really strange.

    If it happens again, please look at the input oscilloscope, see if you see anything strange.

    Best,
    ///Leif

    in reply to: Track prepping #8255
    Leif
    Keymaster

    I’ve never used audacity myself. Does anyone else have a recommendation?

    ///Leif

    in reply to: breakaway broadcast asio? #8292
    Leif
    Keymaster

    Hi Sven!

    The system specs look great to me, except you might want to install XP instead.

    Breakaway Broadcast ASIO is low latency — it uses low latency sound card interface and has low latency algorithms. You definitely want that one.
    Breakaway Live is not an FM processor.
    Breakaway Broadcast is not low latency.

    BBP ASIO is compatible with Airomate 2.

    We usually send out the authorization keys within 24 hours.

    Best regards,
    ///Leif

    in reply to: 1db of attenuation before edcast for low bitrate stream #8287
    Leif
    Keymaster

    Timmy, it’s simple.

    You know that the output of Breakaway Broadcast is extremely tightly peak controlled. You’ve seen what the (de-emph) output oscilloscope looks like.

    So, let’s say you set the BBP output level (into the encoder) to -1dB. Then, you tune into the stream with Winamp, and turn Winamp’s volume controls to full (and turn off the EQ).

    Then, set Breakaway RTA to look at the output of Winamp, and look at the scope. You’ll be able to visually tell where the -1dB boundary is (and if not, hop into the I/O settings in Breakaway RTA and set the Scope Lines to -1.0 dB.

    Do you see "too many" peaks go beyond the lines and hit the rails (0.0 dB)? If so, you need more headroom. Otherwise, maybe you could get away with less.

    So how many is too many? Well, it depends on when peaks occur (on bass/sibilance/voices/percussion etc), and how often they occur.

    For example, if you see lots of peaks appear when there’s S sounds, then that’s horrendously bad. Clipping S-sounds in this manner would sound extremely dirty — even turning S-sounds into F’s.
    On the other hand, if you see just a little peak here and there, particularly if they’re of the single sample variety, then that’s nothing to worry about, and perfectly acceptable.

    Also, make sure to listen. In headphones. The adjustment between loudness and audio cleanliness is a critical tradeoff wherever it occurs.

    Best,
    ///Leif

    in reply to: OverModulation…!!!!! #8279
    Leif
    Keymaster

    Jesse, Vorbis actually does not beat Layer 2 in audio quality if the bitrate is high enough for Layer 2 (for example 320 – 384k), and neither does Layer 3. The reason is that Layer 2 is a subband codec, so it has a much simpler filter bank, which yields more accurate reproduction with much less generation loss for previously lossy-encoded content. With Layer 3 and Vorbis, you get severe generation loss even if you run them at maximum bit rate.

    Flac would be an alternative, except it’s not CBR. I think I’d rather avoid VBR as it adds another layer of uncentainty and unpredictability.

    Sneradio, it will be win32-software like all my other products. Recommended usage will be two machines with solid state drives.

    One with Breakaway Live and the STL transmitter, at the studio site. This way, you also get a low latency processed feed back to the studio.
    One with the STL receiver and Breakaway Broadcast, at the transmitter site.

    All you’ll need between these (if it all goes well) in an internet connection, or even wifi with proper directional antennas at both ends.

    ///Leif

    in reply to: BBP – the best choice ever #8290
    Leif
    Keymaster

    Hi Ricardo!

    If the CPU usage was 0 / 44 before, and 77 / 44 later then 0 (or 77) must be the GUI process, and 44 must be the audio process.

    I’ve never ever heard of CPU usage in BBP going crazy like that. I do recommend a fresh install — it’s the first thing I’d try.

    No problem with reactivation — just install and do it.

    ///Leif

    in reply to: OverModulation…!!!!! #8273
    Leif
    Keymaster

    Thanks, guys 🙂.

    Rocco, I do plan to add parametric EQs to breakaway live and bbp. That should take care of it.

    Here’s my STL design criterias:

    * FIXED LATENCY. Buffer latency tends to drift (i.e. increase and increase) with other solutions. In mine, if you set it to 10 seconds, that’s what you’ll get.
    Latency will be adjustable. If you have a very stable network you should be able to have under a second of delay.

    * Local back-up player (mp3/wav/flac) if the link goes down.

    * PCM and OGG Vorbis format at the very least (since they’re free to use). Mpeg-1 layer 2 and aacPlus if i can work out the licensing — no promises there. I’ll pass on MP3 — it’s not very well suited for an STL, and it’s not free.

    I haven’t started on the project yet, but it’s very high up on my list, since it gets breakaway processing a foot in the door at a few stations. 🙂

    ///Leif

    in reply to: 1db of attenuation before edcast for low bitrate stream #8283
    Leif
    Keymaster

    Depends on the bit rate and encoder. aacPlus with SBR may need a little more. MP3 may need a little less.

    Check with an oscilloscope hooked up to the decoder. Breakaway RTA works great for this.

    One thing to remember is that aacPlus decoders have built-in limiters, and MP3 decoders don’t.

    aacPlus has virtually no peak control (due to SBR) whereas MP3 passes a processed waveform pretty well, with only a few overshoots.
    So, you’ll want to make sure you don’t feed the aacPlus encoder to hot that the limiter at the decoder end starts working, as it creates extreme amounts of IM-distortion.

    On the other hand, if a sample of MP3-overshoot gets clipped here and there, that doesn’t really matter.

    To completely avoid limiter action, you may need 3dB attenuation for aacPlus, but 1dB is almost certainly enough for mp3. You’ll have to try it yourself and find what tradeoff works best for you. Let me know what you come up with 🙂.

    ///Leif

    in reply to: OverModulation…!!!!! #8269
    Leif
    Keymaster

    Yes, putting the PC at the transmitter is definitely the best way to do it, so the STL doesn’t have to send processed audio.

    Next on my list of software to write is software for doing a stable STL over the internet. Stay tuned 🙂.

    ///Leif

    in reply to: OverModulation…!!!!! #8267
    Leif
    Keymaster

    Rocco, YES. My god yes.

    Try an A/B comparison with a properly calibrated BBP against a properly calibrated 8500, and you’ll know what I mean — it’s a night and day difference. Make sure to listen in a good system at reasonable volume (not cranked up too loud, as that alone masks a lot of the distortion you get from the 8500!).

    ///Leif

    in reply to: OverModulation…!!!!! #8265
    Leif
    Keymaster

    Hi Rocco!

    I think you’re talking about my composite clipper 🙂. It’s not part of Breakaway — it will be part of a hardware processor I’m developing.

    However, with Breakaway Broadcast Processor (BBP) you do get perfect L/R peak control, so it can be as loud as any current hardware processor, even the ones that have a composite clipper, since traditional composite clippers do not squeeze extra L/R loudness in the way mine does. BBP is already a good choice, and at an incredibly low price.

    Of course, once my hardware processor comes out, you might want to look at that too 🙂. It will be a little less expensive than the current top of the line processors (such as omnia 6 and 8500), but it certainly won’t be as cheap as Breakaway.

    Best regards,
    ///Leif

    in reply to: Hiss on stereo ??? #8032
    Leif
    Keymaster
    quote :

    Another point: I have cut the HPF PCB trace on the card’s ADC (pin 19 as stated on the mod instructions) and for some reason some DC+high freq oscillation appeared on the MPX output. The pilot tone got a companion sinusoid and got pretty deformed and the only remedy was to rewire the previously cut trace. Other than that, the mod was perfect !

    Whoa, that’s not supposed to happen!

    I’d check and make sure that pin is still going to the ADC (input) chip. I don’t know how it could modify the output, unless they changed the board design. The mod worked for me.

    Adding a cap to the output is also a good idea! This should remove or reduce the need for tilt correction.

    Best,
    ///Leif

    in reply to: Hiss on stereo ??? #8030
    Leif
    Keymaster

    Hi Ricardo!

    No, you won’t need a NIC, but you’ll need a USB flash drive so you can copy and paste the activation code for offline activation. That way you can use another computer with internet access to handle the activation.

    ///Leif

    in reply to: Sorry…this one is a bit off topic #8264
    Leif
    Keymaster

    I understand. If I’m bored one day with nothing to do, maybe I’ll write one. 🙂

    ///Leif

Viewing 15 posts - 751 through 765 (of 1,890 total)