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[quote author=”guillou”]This is a vmware ! lol[/quote]
What gave it away, Sherlock? The VMWARE TOOLS icon in the tray perhaps? 😉
///Leif
Hi Diego!
Out of those two, I definitely recommend the Juli@. Realtek ALC662 is 96 kHz only, not 192.
I don’t have the tilt value, but maybe someone else does 🙂.
///Leif
Hi Jeff!
0.90.44 is also a very old version. The current version in 0.90.69. Try that instead — whatever the problem was, it’s possible that I’ve already fixed it!
///Leif
Hi Appie!
Yes, it runs perfectly in Windows 2000, provided that you install a reasonably new DSound.dll. The version that comes with Windows 2000 is too old — my DirectSound code depends on a newer function in dsound.dll.
However, to be honest, this is a pretty silly dependency, considering that it also supports WaveOut and KernelStreaming. I think I will do away with it, so that it will run on any stock windows 2000 installation as well.
///Leif
Sounds like a driver problem. If you have a different sound card to try with (for example a USB headset would work), we could find out for sure whether the problem is with breakaway or the driver.
///Leif
Indeed. It’s a proprietary technology, with yearly licensing due to ibiquity, but the worst part is — it’s AWFUL. It sounds truly horrendous. You’re not missing much.
In my humble opinion, if they had used aacPlus, it would have been worth something. However, they didn’t — due to political reasons (i.e. Lucent wanting a piece of the pie) they’re using the PAC codec. When they later realized that there was just no way to make it sound acceptable at the low data rates HD radio provides, they licensed just the SBR portion from aacPlus. So, now you get PAC artifacts doubled in TWO places in the spectrum. Woohoo!
///Leif
Hi Ken!
It’s not possible. HD radio is not actually a subcarrier of FM, but are actually separate carriers, next to the main carrier!
Then again, I’ve heard that there are FM rebroadcasting receivers (like the fanfare) that are wideband and accurate enough to capture the HD carriers as well. If so, it’d be possible to record it, but you’d need a lot higher sampling rate than 192, which is as high as any sound card i’ve ever seen goes.
///Leif
quote :I’m confused. If you can do it with 2 channels, why not 64?I’m not doing it inside the pipeline (a driver presenting two wave devices synchronized to its internal clock). I’m doing it in my program (a user-level application using two separate wave devices with different clocks). The number of channels, as I said, doesn’t matter. Indeed, if you can do it with 2, you can do it with 64.
///Leif
It depends on the program material. That peak means they had a lot of low frequency stereo at the time. Playing piano will usually do it!
If you’re using the ITU BS.1268 limiter in BBP, that peak will completely go away, and be level with the rest of the subcarrier.
That peak probably means the station would be difficult to receive cleanly in stereo in a high-multipath environment!
By the way, look, I know some people love building audio-processing chains, but seriously, don’t put any other clipper after BBP. You will at most gain 0.24dB ( see http://www.indexcom.com/papers/loud2.html ) but for that imperceptible loudness advantage you lose the careful, distortion-free way in which BBP clips its output. Any tiny little bit of clipping of the output will be immediately audible on difficult program material. Composite clipping, to be effective, has to be done as part of the processor — not as an afterthought.
///Leif
Howdy!
Nice catch, Ken! Thanks. Oops 😉.
Breakaway RTA runs completely standalone. You could hook it up to anything.
Thus, I have no idea what happened there, Timmy! What audio device did you tell it to use? Did it end up picking up your laptop microphone or something like that? 🙂
When calibrating a speaker system with Breakaway Live (next release), you need an RTA and a calibrated microphone. I realized that it would be inconvenient to actually have the RTA built into Breakaway Live, as you’d then be forced to route the RTA microphone into the same computer running Breakaway, which may or may not have an available audio input for this purpose!
So, instead I just decided to make the RTA completely free. You could use it as a convenient loudness meter, or to calibrate a system with a hardware eq, or to calibrate a system with Breakaway Live. Your choice! 😉
I still have to write the layout algorithm though.. It’s a major pain in the neck — I have yet to manage to write any sort of automated procedure for this, so i basically have to code everything manually, like "if the width-to-height ratio of the window is X and there is at least Y width-pixels available then put the oscilloscope HERE", and write enough of those to cover every conceivable window size. Stuff like that. The layout algorithm for the main window of BAE/BBP/BL is over 1000 lines of code like this! There’s gotta be an easier way.
In the case of Breakaway RTA, if docked to the left or right side of the screen, it needs to show only the ITU loudness meter (since it fits the tall and skinny model). If docked to the bottom and is skinny enough, it also needs to show just the ITU loudness meter, but it could also give you the option to show ITU, RTA, Scope1 and Scope2 side by side, since those four side by side would also fit nicely in a taskbar!
///Leif
quote :Because of the number of channels? VAC 4.x shows up in ASIO4ALL as 64 channels in/out, and I have definitely used it successfully as an 8-channel device. I have also used Audio Repeater KS as an 8-channel device. Perhaps Audio Repeater KS could serve as the basis for your SRC product.No, the problem is adding SRC inside a pipeline, not the number of channels, for the reasons I already mentioned 🙂.
quote :The entire Windows audio environment makes it obvious that Windows PCs were never intended to be audio devices! Trying to get bit-perfect audio from input to output is all but impossible.True that. However, with proper hardware, it’s really not that bad. Your issue stems from using ASIO4ALL. If you used a REAL asio device, you would not be having this problem — it would be properly synchronized just like it should, and you’d be able to get bit-accurate input to output.
Sync problems in WASAPI still? That sucks. Hopefully they’ll fix it before Windows 7 release.
///Leif
Actually, coming right now.
http://bredband.leif.cx/browse/rta
This is barely even alpha. Try at your own risk!
Layout algorithm is basically non-implemented. If you pull the window size to extremes, or dock, it will probably crash. 🙂
///Leif
quote :You could probably make a minor product just out of a multichannel Breakaway Pipeline with SRC. I could certainly use an 8-channel version at the moment.That would be a nice thing to have! But, I’m not sure it’s possible either.
I didn’t write the Breakaway Pipeline. It’s the only part of the Breakaway products I didn’t write myself — I licensed Virtual Audio Cable 4.x. Even if I had written it though, I have reasons to believe it won’t work 🙂.
Previous versions of Virtual Audio Cable (3.x) had a synchronous mode, where the cable had no clock, but instead followed the clock of the application feeding data to it. This came with some issues, but it did solve the synchronization problem. It was possible because VAC 3.x was a native Wave driver.
I once asked Eugene why there wasn’t a synchronous mode in VAC 4.x. If I remember correctly, he answered that due to the way WDM driver architecture is, if the driver stands still for even a couple of milliseconds (while waiting for whatever it’s synchronizing to), the audio driver will be forcibly reset by Windows. WDM drivers has to function within all sorts of different frameworks (like the DirectSound emulation) so I can see how issues like this would come up.
The cleanest way would be for you to write that WASAPI support, I think. By the way, when you get that working, let me know if you’re interested in selling the WASAPI interface code 😉. I could use it, and can’t write it as long as my primary system runs XP.
///Leif
