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January 12, 2010 at 4:07 am in reply to: Question: Should I Increase INPUT vs OUTPUT (44100 vs 48000) #5093LeifKeymaster
I understand exactly what you mean — that IS the question I answered! Please read the response carefully. Which part are you confused about?
///Leif
LeifKeymasterIt’s not a bug in MpxTool, it’s a bug in FM Stereo 🙂.
What you’re seeing IS the RDS signal being demodulated into audio, and this happens with EVERY radio. The reason is a simple matter of bandwidth.
The stereo subcarrier is demoduated by frequency-shifting it down 38 kHz, so that 38 kHz becomes 0 (DC) on the output, and 23/53 kHz becomes 15 kHz on the output.
In 99.99% of MPX decoders (inside radios), real-world filter limitations mean that the decoding goes outside this range, so even the 19 kHz pilot appears on the output.
In MpxTool I designed a brick-wall filter which completely stops the 19 kHz pilot, and completely passes 18 kHz and below. So, in MpxTool, everything between 20 and 56 kHz gets wrapped down into the audio, for 18 kHz frequency response on FM.
The pilot has no bandwidth — it’s a pure tone at 19 kHz, so it doesn’t get through. The RDS signal, on the other hand, has bandwidth, and has information between 54 and 60 kHz. Thus, the 54 – 56 part gets into the audio at 16 – 18 kHz, just like in a regular radio. It’s just the nature of FM.
There IS a way around it though!
MpxTool is (afaik) the only mpx decoder in the world which allows you to choose to decode the whole stereo signal, only the lower sideband, or only the upper sideband.
You can select this with the Mono/Stereo switch — it also has LSB and USB.If you choose LSB, MpxTool will decode the entire stereo image from just the 20 – 38 kHz part, and ignore anything above. With this option enabled, NO rds leaks through into the audio. As far as I know, there are no drawbacks to this method, other than being a highly non-standard approach to decoding the MPX signal.
///Leif
LeifKeymasterAlright, sounds like I need to make a couple of changes. I will add a tone generator to BDJ. This will let us find out which part is skipping — the OTS->BDJ link, or BDJ’s audio engine itself.
I should also probably add the option for Audio Realtime Priority back in. If BDJ is what’s skipping, this may help.
To be honest, I completely forgot to even test the Volume Control code. My keyboard (IBM Model M, 1991) doesn’t have the keys, so it’s easy to forget. I will test and get feature working for the next version. 🙂
Bryan, it uninstalled just Breakaway Pipeline, not BAE. You can still use your older version of BAE with the new pipeline! You shouldn’t have to change anything at all — it should just work.
Best,
///LeifLeifKeymasterI absolutely love my Sennheiser HD650s! Big, accurate and powerful sound. And, when they’re too big to bring, I’m quite happy with Etymotic ER-4P.
///Leif
January 12, 2010 at 3:23 am in reply to: Breakaway 1.30.02 with transparent toolbar for Windows Aero! #5048LeifKeymasterGood point. I’ll work on it 🙂.
///Leif
LeifKeymasterYes! Breakaway DJ is the answer. Apart from low latency, it also has the 12-band PEQ which (together with the free Breakaway RTA app, and a measurement microphone) allows you to calibrate your speakers.
Best regards,
///LeifJanuary 12, 2010 at 3:20 am in reply to: Question: Should I Increase INPUT vs OUTPUT (44100 vs 48000) #5091LeifKeymasterHowdy!
There’s a simple answer:
Most consumer sound cards (including on-board) perform best at 48000hz (or ONLY at 48000hz). However, most music is distributed at 44100hz (CDs).
The conversion has to happen somewhere — the question is where. In the sound card driver, in windows audio back-end, in the player, or elsewhere. Breakaway’s sample rate converter is cleaner than any of the aforementioned ones, which is why 44100 input and 48000hz output is the default in Breakaway, so that Breakaway gets to do the conversion. 🙂
///Leif
LeifKeymasterVery good point. I honestly don’t remember whether it also appears in Bypass or not, but it would certainly be a good idea for it not to. I’ll check and make sure.
///Leif
January 11, 2010 at 2:15 pm in reply to: New ASUS motherboards with sound : VIA VT2020 (BD192/24 ENVY #8244LeifKeymasterEven brand-name computers are usually standard ATX, so it should work, but you never know until you’ve tried it. Why not just build a system from parts? 🙂
///Leif
LeifKeymasterWhen importing a playlist without Breakaway DJ but still using ASIO, how does it work?
///Leif
LeifKeymasterSo far it has proven to be completely stable. I myself would trust it live at this point.
Yes, there will definitely be updates and new settings. The reason for the small number of presets now is that the existing presets in the other Breakaway versions were pretty much all geared towards broadcasting and/or smaller speakers. The presets in Breakaway DJ have been all been re-tuned for PA systems.
Best,
///LeifLeifKeymasterYes — from the great test results so far, I’m confident enough to sell it 🙂.
Yes, it will coexist just fine with all other Breakaway products.
Best,
///LeifLeifKeymasterIf you’re using Breakaway, it’s best to put the sound card in 48 kHz 16-bit mode, to avoid further conversions. Sound Blaster cards (and most on-board solutions) perform well at 48 kHz but not so well at 44.1, which is why the defaults in Breakaway are 44.1 kHz input, 48 kHz output. Most music is 44.1, and Breakaway contains a high quality sample rate converter, for the best possible quality 🙂.
///Leif
LeifKeymasterHey, in case you’re interested, here’s the web store link 🙂.
http://store.claessonedwards.com/Breaka … dj-001.htm
///Leif
LeifKeymasterAh! Now I see what you mean.
There’s no easy way to casually route audio between different programs. I’ve been thinking about developing some kind of solution for this, but can’t seem to find the time between every other project.. Maybe Jack for Windows is an option?
Otherwise, perhaps your sound card has built in routing features? Many do. If you could make your sound card route the output to the appropriate input, you should be able to open it as a Wave or KS device in Breakaway RTA and get the audio in that way.
///Leif
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