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That’s even better, Sgierk. That way you also ensure that the input isn’t clipping (pre-attenuation).
///Leif
Scotty, add an attenuator as the first link in the DSP chain. That will take care of it.
http://www.claessonedwards.com/plugins/
///Leif
Hi Scott!
The VAC default is 10ms per interrupt — I had him change it to 1 for my licensed version (Breakaway Pipeline). 1ms per interrupt enables lower latency through the pipeline.
Regarding the audioscience card, I’ve never tried it with Breakaway Live, and it’s not impossible that it’s using an ASIO data format not supported by Breakaway.
Breakaway currently supports 16 and 32 bits. 24 bits is currently not supported at all, and will yield silence. Most 24 bit cards actually deliver the audio in padded in 32 bit format, so it works. However, Asus Xonar DX, when set to 24-bit, really delivers 24-bit, and as such doesn’t work with Breakaway Live.
However, Asus Xonar DX can be set to 16-bit, and then it works perfectly. Perhaps the AudioScience card has a similar setting?
///Leif
I suppose building a tall concrete tower and putting the transmitter + processor closer to the antenna would reduce transmission line losses, but I can’t imagine the losses of proper high voltage cabling (run from the BOTTOM of the tower and up) would be so severe that this solution would actually SAVE money in the end.
I have never, ever heard of this being done.
///Leif
No, connecting it that way would mean you would be recording unprocessed, not processed.
If it gets confusing, draw a schematic on paper first, laying out exactly how you want the signal to go. It will make much more sense.
Breakaway cannot send to a pipeline and the soundcard at the same time, so you must send Breakaway Output to a pipeline. You can then download the free version of Virtual Audio Cable from http://software.muzychenko.net/eng/vac.html and use the program Audio Repeater from there, to also send the pipeline audio to the sound card.
///Leif
Make sure you’re matching the formats. StereoStocker will open the pipeline as 44100hz, 16-bit, Stereo.
///Leif
Looks like we scared him away already 🙂.
The simple truth is that the only two instruments you need, to verify an FM processors output (including the output dac, i.e. the sound card), is an oscilloscope and spectrum analyzer.
*Any* and *all* issues will be detected. Phase problems? If they’re bad enough to cause overshoots, the scope will tell you. Frequency response? Same. Non-linearities? The spectrum analyzer will tell you — you’ll see it as garbage around the pilot, where BBP normally has none at all (80dB pilot protection).
///Leif
Hi Giovanni!
I would imagine the best configuration is to select Breakaway Pipeline as the output device in DirEttore, and then select Breakaway Pipeline as the Input Device in Breakaway Live. Finally, select the sound card as the Output Device in Breakaway Live.
///Leif
I’ll check. Before that though, could you make sure to check the Input scope, to make sure the distortion isn’t already in the original (distorted CD)? It’s not uncommon for distortion to already be in the material nowadays (more rule than exception actually), and CGSmooth is pretty mid-heavy and could possibly exaggerate it.
///Leif
Yeah, you could use StereoStocker with a pipeline.
///Leif
Hi Modman!
Was the station you were comparing to, by any chance, playing MONO at the time?
Because, if the signal is mono, or nearly so (even if the pilot is on), you won’t lose any loudness from switching to mono.
However, if the signal is wide stereo, switching to mono cuts out a LOT of information, and loudness can drop significantly.
This is why there is a Mono button in Breakaway Broadcast. It doesn’t just turn off the pilot, it also sums to mono BEFORE the processing, instead of after. This way, any loudness loss will be compensated for by the compressor.
This problem is inherent to Stereo vs Mono, regardless of medium. The only way to avoid it is to do L+R/L-R processing, but doing that on FM would mean you’d be 3dB quieter *all the time* instead of just when listening to wide stereo programming with a mono receiver. 😉
It can not possibly be a PEQ issue — that would have shown up during calibration, and/or on your modulation monitor.
Best,
///LeifThanks, Conor!
I used to have a 6200.. Same engine as 1100 if I’m not mistaken. Sold it on eBay quite a while ago now! 😉
///Leif
The signal you get this way is PRE-processing, that is, BEFORE the processing is done, that is, UNprocessed.
If it was processed, you’d end up double processing.
All you have to do is (inside BBP I/O configuration), select Live (instead of wave/ks/ds), and then select the Breakaway Live instance. Done! 😉
///Leif
Hi Stuart!
Between Phase Linear and Low Latency, the difference is.. Well.. Phase linearity.
If you’re listening carefully, the difference is clearly audible while switching back and forth, especially with aggressive processing settings. If latency isn’t an issue, it’s a no-brainer — use Phase Linear. If latency IS an issue, again it’s a no-brainer — use Low Latency.
Extra low latency mode cuts down on the look-ahead time of the bass bands. In Normal Low latency / Phase Linear mode, the look-ahead for the bass bands (b1, b2) is 2ms. In Extra Low Latency mode this gets cut down to 1ms, translating to ever so slightly dirtier bass. However, if latency requirements are *that* tight, again it’s a no brainer!
BBP is always phase linear. CPU optimized vs Maximum Quality in BBP switches clipper modes. Maximum Quality uses more stages so that it can spread out the work, mask the distortion slightly better, and have ever so slightly better peak control. However, CPU optimized is very close is quality (let’s say 95% of the quality with 50% less cpu use) so if you’re short on CPU, this decision is also a no-brainer.
///Leif
