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Hello!
First of all I want to thank you for this incredible normalizer/limiter/enhancer AND offering it for such a favorable price!
You cannot imagine what relief it was, when I tried the demo, not expecting a lot, and find the ultimate and perfect normalizer for everything I have and use: music, movies and as it seems even games (not tried yet).
If I had known of its existence in the last decade, it would have saved me days(!) of tweaking media players, sound cards and Windows to get the experience I was looking for; that never happened of course, contending against the Loudness War, low quality recordings and varieties of movie genres.
I have found it by accident while looking around in an audio forum once again for a method for applying DRC on movies.
It seemed too good to be true, but here it is! 😀 Any of my so often used test tracks sounded simply as they should while maintaining the same perceived volume but without losing essential dynamics! And by moving a slider I might hear all those concealed instruments, effects and background details. 😉Anyhow, here are some findings I want to share with you:
1.) This is not a bug per se and rather belongs to the category “why should someone do this”. If a slider is being operated in BAE by the mouse while changing the volume in Windows (for example by media keys on the keyboard), which would trigger BAE to move the volume slider, it starts to freeze and closes shortly after that event, subsequently sound stops. Of course it has not been designed for such an action (it would have to move up to 2 sliders simultaneously) and this would not happen if just the mouse is being used.
2.) While being opened, the BAE GUI uses a good portion of the GPU, even if no sound is being played. On my Radeon RX 590 it is 10%, while on my Radeon RX 5700 XT it is just 3%. So it is very hardware dependent and may not sound so bad but this denies its power saving mechanism.
Therefore I would suggest to add this to FAQ because the solution is simple: shrink its size until all meters are gone, then GPU usage is 0%! 🙂3.) It lacks DPI scaling or any form of scaling which makes it very hard to use on 4K screens. This has already been posted several times here in the forum. If such a feature is out of the question technically and/or financially, may I suggest to add a simple option to just double its sizes? I imagine this could be done much easier than rework it to support DPI scaling? That would aid 4K resolution issues instantly and 8K issues in the future (double again).
I remember WinAMP having such a feature back in the day where scaling was not an issue, but in case someone had a ludicrous high resolution CRT monitor.Best Regards,
WilhelmWelcome, Wilhelm. Thank you for your comments and suggestions. The 4K res has been raised before as you said. I’m not sure how hard it would be to implement this, but Leif is aware of it, so it may just pop up on a later release.
Did you have any issue with Virtual Audio Cables crashing in Windows 10? It apparently happens in some iterations of the 20H2 release, and a resolution to the problem is in beta testing, so should be in the wild very soon.
Hello Milky!
Thanks for the warm welcome!
I’m using build 20H2 (fresh install) and have never noticed any problem with Kernel Streaming mode, if you mean that in particular.
It runs on a machine with AMD Ryzen 3600, X470 Chipset and Realtek onboard audio using the latest driver Microsoft has to offer: “Realtek Semiconductor Corp. – MEDIA – 6.0.8963.1”Supplementary note to article 2 in my initial posting:
Windows has transparency effects enabled by default and if such window (or Start menu) is moved on top of BAE, it doubles its GPU hunger! That generates another 10% usage on the Radeon RX 590 which is eminently bad on my HTPC, where I need up to 30% of the GPU for upscaling and image refinement via madVR, while it must not get noisy. If BAE is not resized or minimized, it adds hefty to that utilization.
So it would help a great deal if there was a startup option to have it minimized, as a task bar icon, or to preset its size without animiated meters.PS: I can help myself but not everyone might be capable to tweak it that way.
Ah, but there is! If you right click anywhere over the RH panel (or BAE logo) when BAE is open in full screen, there’s an option to “Close Main Screen”. This drops it to an icon in the Systray. If you click ONCE on the icon, you get a tiny popup where you can mute, bypass or adjust the volume. If you double-click, you get the full screen back. It remembers this state when the PC is shut down, so it will only open full screen if you want it.
Have you had a chance to tweak the buffers and things in the setup? If you ran the wizard, BAE makes a few guesses. You can then fine-tune these selections.
Right click on the main screen logo and select “Settings”. First click on “Test” and read the information displayed in the central window. Amongst others, it will display an input and output jitter percentage. The aim is to get that jitter as low as possible.By tinkering with the KS, WAV and DS options, and the size and number of buffers, you should be able to get jitter to around 10% or better. If you can’t, but can’t hear any glitches, that’s fine, but fine-tuning is worth it.
Also, by default, BAE upscales from CD quality (44100) to DVD quality (48000). In most cases, this is pointless, so it is better to select 44100 for both input and output, and then adjust the buffers accordingly.
Usually (but not always), buffers in multiples of 441 (100th of sample rate) works best with 44100, but every sound card, and PC setup is a little different.
Hello Milky,
sorry for my very late reply. During the pandemic I was hardly motivated to do anything that reminded me of things I must not do, like parties.
Now that all begins to start up again, I thought I take a look what’s going on here.Thank you for answering some of my questions in advance. 😀 I thought so that buffer size and sampling rate must correlate that way.
I have experimented with high and low settings and it benefits most from doubled buffer sizes but fewer buffers (2 or 3).
Because I play back a mixture of 44.1 and 48 kHz material, I chose 48 kHz in and out, as well as buffer size to 960 instead of 480. This helps with jitter a lot and keeps it around 5%.Speaking of optimization, I’ve got two more questions please:
First, the sound card settings. I’m pretty sure if BAE is set to 48 kHz in & out, the sound card should be set to 48 kHz too. Is there any (subtle) benefit of setting the sound card to a higher setting, especially a multiple of BAE’s? My new Sound Blaster AE7 can handle up to 384 kHz which is default.
Second, its bit rate. Mine defaults to 32 bit. Most others to 24 or 16 bit. What’s the recommendation here?
Best regards,
WilhelmPS: My home parties will never be the same again with BAE! <3
PPS: The download link on the main page is still the previous version 1.40.03 https://www.claessonedwards.com/home-audio/
As you have discovered, a bit of tweaking can make a difference to the jitter. However, the trade-off is latency. More buffers equals more time taken to fill them. If you are playing music videos through BAE, you may notice that the lip synch is off compared to the audio. If that isn’t an issue with you go with the lowest jitter. If you can’t adjust anything to get the lip synch back, you may have to sacrifice a bit of jitter for smaller buffers. So long as you don’t hear any dropped frames or artefacts, it should be fine.
A lot of music is recorded in the studio with higher bit depths of 24 or 32 bit, but typical audio devices only play back using 16 bit depth. The “Red Book” standard for CD quality/WAV files is 44.1 kHz/16 bit.
Like you, I like the sound card, playback software and BAE to all be aligned to the highest common denominator, but this may mean that your music is being upscaled multiple times. If you play predominately CDs or WAV files from CD at 44.1 kHz and only the odd recording sampled at 48 kHz, you are forcing your soundcard, BAE and playout software to upscale every track, and they will only be blank frames anyway, because there is no additional information to insert in the extra samples.
After fooling around by maximizing all buffer settings I have actually seen huge latency in my music videos for the first time. It is simply unusable.
However, no need for that! 🙂
Jitter results using the wizard:
Tiny: 25-50%
Small: 11%
Medium: 11-25%
Large: around 11%, but jumpy between 0% and 25%
Huge: skipped due to crackling noise after hitting the test buttonSo I began tweaking the Small Buffer setting, starting with input 441/3/44100 and output 480/3/48000 (all 2 channels).
Because I have to play back a mixed bag of 44100 and 48000 kHz material, I set both (and the sound card) to 48kHz; some is true CD WAV audio, some is coded in 48000 like OPUS from a well known platform…
Doubling buffer size to 960 results in 5% jitter. Maxing it to 2048 it gets as low as 2-5%. But I think I like the thought of having it exactly double.
Lowering buffers to 2 or higher than 3 is no benefit, so I leave it at 3.PS: After finishing the wizard it would make probing way more comfortable if it would get back to the Settings window instead of closing it and returning to the main window.
Useful information, WilleHelm.
However, so much of this is affected environmentally, and even by your PC specifications, so it is always a “balancing act”. I’d like to write the perfect “Tuning Tool”, which would take Performance Specification (CPU/RAM/HDD/Cache/Sound Card etc) into account and display the exact buffer sizes to use.For now, we just have to fine tune as best we can.
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