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[quote author=MrKlorox link=topic=5780.msg20135#msg20135 date=1558011122]
Make sure ASIO4ALL is installed and reload the driver before ever opening the ASIO driver control panel. You should then be able to choose your sound devices.
[/quote]Only use ASIO4ALL if you want a pseudo ASIO driver. It just makes kernel streaming calls, and emulates ASIO. You might be able to achieve it without going through another layer of drivers.
The search engine isn't really that clever (unfortunately), so it is not the powerful tool it should be.
I agree that a manual would be handy, but Leif is a great code cutter, but not that strong in the manual department. Several have offered to contribute towards a manual, but there is so much in the package that most of us are not even aware of.
There's also the problem that, even if there is a manual, you have to find what you are looking for (similar to the search engine), so you may still miss the answer. Leif has made some application specific videos, and they are very good. We need him to use the same concept, but maybe go back to basics and start from installing the software to getting it up and running in various scenarios.
You probably need to read or re-read this topic which covers the BA1 interface. http://www.forumclaessonedwards.com/forum/index.php/topic,5552.msg19512.html#msg19512
I don't know BAE at all, as I came to this forum from a different route. I DO have BA1 running an HD core successfully, and I can take you through those steps. Basically leave everything as the default except the necessary input and output selections and you should get something. Fine tuning can come later.
This is a potted version which may help.
First, select your HD Core.
Main Configuration > Audio Processing Cores > HD Cores and drag the slider until it says "1".
Now configure the core.
Main Configuration > HD Processor > Core Feat – at this stage, leave all these options off.Main Configuration > HD Processor > Input > Interface > Select Kernel Streaming (if there are problems later, you might need to change this, but it is the best option).
Main Configuration > HD Processor > Input > Select Breakaway Pipeline 1 (This is the device you will need to select as output in your playout software).
Main Configuration > HD Processor > Input > Sample Rate > (I use 48000).
Main Configuration > HD Processor > Input > Block Size > (I use 960, but click on the "Auto-Configure Block Size" button and accept the recommendation).
Click on "Run Test" and aim for the lowest jitter rate. Usually the auto test option is best.Main Configuration > HD Processor > Output > Interface > Select Kernel Streaming.
All other output options can remain OFFMain Configuration > HD Processor > Speakers > Select Kernel Streaming
All other speakers options can remain OFFNow, setup ASIO.
Common Audio Devices > ASIO > Main > Select your ASIO sound card from the drop down.Common Audio Devices > ASIO > Main > ASIO Driver Control Panel. This should open your audio device interface. set sample rate, buffer size etc in there, exit out of it and then select "Reload Driver". This will map any changes back into BA1.
Common Audio Devices > ASIO > Main > Set Device Sample Rate. I turn this on and then select 48,000.
Common Audio Devices > ASIO > Inputs > Both Left and Right can stay OFF.
Common Audio Devices > ASIO > Outs > Common Monitor Output. I have both L/R OFF.
Common Audio Devices > ASIO > Main > HD Processor > Select your usual sound device outputs e.g SPDIF 1/2. These would be the channels you would select in software if not using BA1.
I'm sorry to appear so vague, but I really haven't been there yet. As I understand, the licence is tied to the MAC address of the network card, so a new licence will need to be generated. I'm not sure of the procedure, so this could be a learning experience for all of us.
Drop Keith (keith@claessonedwards.com) an email to find out, and please let us know how you go.
No, I don't believe that is the case. Your announcers should be listening to the source (straight out of the mixing desk, or wherever your mic processing output is). In the good old days, before all the processing that occurs these days, we used to listen "off air", meaning that we heard what the listeners heard. However, that only works if the STL is not far away and therefore the difference between the input signal and the returned (broadcast) signal is only milliseconds, which our brain cannot detect. This was an advantage because we could detect instantly if the broadcast chain went down (as in a transmitter or STL failure).
Nowadays, there are many delays between the studio and, in the case of (say) a terrestrial FM broadcast (which might be almost straight out of the studio console) but also streamed through (say) Icecast, listening off the Internet would confuse even the most seasoned broadcaster.
Oh WOW! Welcome back, Oh Lord High Poo Bah.
I've been soldiering on under fairly high duress here. Thanks for your clarification 🙂 🙂 :).It's very hard to make an in-depth assessment from such a small grab of the logs, but my first reaction was that the warning on jitter rate seems to be quite regular, so, maybe BA1 hits a trigger point where the jitter is considered to be too high, and this forces a restart of the service.
Although I have seen jitter rates up to 100% without audible glitches, I try to get it down to less than 20%. Sometimes the recommended buffer size is not actually the best, and it does require a little experimentation between the ASIO buffer size of the actual sound card, and the internal I/O buffers.
Audio processing typically uses buffers or "blocks" to scoop up data and feed it through the program. This avoids hundreds of hard drive or input stream accesses, or, if the hard drive or stream is busy or compromised by other demands, it avoids drop outs. The same thing happens on the output side, but the down side is the enemy of all audio processing – latency.
This means that it might take several seconds to fill up the buffers, and the same time to empty them, creating a gap between the real time audio and the processed audio. This can be most noticeable when processing music, because the lip sync or things like hitting a drum snare does not line up with the audio. In radio, the announcer's spoken word if fed through a processor, comes back to him delayed by milliseconds, making it very hard to monitor your own voice.Generally, leave the fifo settings at the defaults, but, if you can reduce them and not get dropouts or glitches, this may improve latency. Conversely, if, at the default, there are still glitches, increasing the size may improve, but at the expense of higher latency.
I don't stream, but I believe it is where you define the connection, username, password etc to your external streaming service.
Latency has nothing to do with sample rate. It is the delay between the original signal source (for example – your spoken word) and the final signal ater processing. If there is no processing, the two are the same. If the ASIO buffers are very small, the delay will be minimal, but there may be artefacts, so the trick is to increase the buffer size just enough to get rid of any glitches.
Alternatively, listen to the unprocessed signal straight out of the desk.
Joint stereo saves space by masking off (mono) any signal which is L-R identical, so you only hear the differences as true stereo. In theory, there should be no difference (except space and bandwidth savings), but the purists will insist that both channels should contain all the data, including that which is duplicated in each channel.
Your strategy is sound.
There are several variables here, each of which can be the problem, or may contribute in some way.1) The OS. Although Win10 has been around for a while now, it does not have the same track record as Win7. If it were easily possible, a solution could be to downgrade the Win10 PC to Win7 so the OS platform is the same. Maybe a corollary of this would be to update the Win7 PC to Win10 to see if that introduces issues that are not present or are masked in Win7.
2) The hardware/drivers. Unless the two PCs are identical, there can be a variation of components or drivers influencing the final results. It would be great if both devices were the same – ideally based around the more stable unit.
3) The BA1 Management Package. I'm not convinced that this is a game changer, but Leif is on record as saying that running BA1 as a service uses less resources and therefore may be more stable.
4) There may even be a hardware issue, such as bad RAM in the PC which fails. Simply replacing the RAM with the best you can afford might save a lot of frustrating testing to achieve nothing.
