In the past I've bought a USB network adapter for a laptop when I couldn't get network drivers for the builtin network hardware that didn't have too much latency.
FWIW I keep my audio buffering up at about 10 or 20 seconds, then I don't have problems when a drive needs to spin up, etc.
Last post by NEMO7538 -
Thanks, I've managed to migrate my old profile without issues by copying values.
I want to report another small issue, I don't know exactly if it's old or happens only in the new versions:
When seeking the volume bar in the deskband controls to the left, volume level goes to 100% if mouse is right on the left edge of the bar or even outside controls on the left. Although it displays the lowest possible value (it's still filled a bit), if your scroll your mouse one step up you will get from 100% (0dB) to "Mute", then on the next step you will get to ~-45dB.
I think it's a bug and volume level should go to "Mute" (0%) in this case.
I've tried two things: doubling the audio buffer in foobar2000 (to 2000 milliseconds) and disabling the Ethernet controller.
Doubling the audio buffer didn't help. I still had audio artifacts when the hard drive and ethernet controller were heavily used (legal torrenting in the background). These artifacts also occur on Youtube and I have audio AND video artifacts with VLC as well (here is a video of this problem occurring, this is not my video but the "buzz" is exactly the same).
I've disabled the Ethernet controller and ran LatencyMon again. This time the results are OK, and I didn't hear any artifacts.
Feature Request - in peak hold mode, I'd like to be able to set gravity to a very low value. The setting appears to be limited to integer values, i. e., it will not accept a decimal point.
I listen to a lot of downloaded audience concert recordings. I've been experimenting with using the Foobar equalizer along with your spectrum display with peak-hold as a rough tool for spectral sculpting. (Have you noticed that the eq's bands are all, effectively, at A & D#?) I expand the amplitude to 80 dB, & the bandwidth out to D#10 (19.912 kHz - close enuff to 20 for gov't work, amirite?), and I put splitters in the display areas for the spectrum & eq to align A1 with the 55 Hz slider, and A9 (14080 - anybody wanna quibble?) with 14k. This aligns the eq sliders very nicely with the the spectral display. I enable peak hold, with a very long hold times (a minute or more!), and have been playing with gravity values.
The scheme would work ideally with a VERY slow peak decay - 1 is quite a bit too fast for this purpose, but 0 will hold the extreme peaks forever, so that's no good either. I'm trying to tweak the peak decay, via hod & gravity, to work as a sort of an averager. Not what they're designed for, I know, but I come from an applications engineering background, and finding new ways to use existing tools is sorta what I do. Anyhow, a fractional setting would be an interesting thing to experiment with.
Just wondering of this is possible - thanks for a really useful tool, in any event.
CueRipper, mostly because the checking against CTDB is reliable enough for me to not having to deal with slow/paranoid read speeds of EAC. And because I use(d) CueTools (same download package) to periodically check my files for corruption, it's a no-brainer to just use CueRipper as well. I also faintly remember some unicode related issues with EAC, but that seems to be a thing of the past.
A side note about your planned backup format, there is really no reason to backup to wav instead of any lossless format like flac/wavpack/etc. Image vs split tracks is a matter of preference, on the other hand. I prefer to have split tracks + .cue because I use wavpack hybrid, so I just copy the lossy part to my phone - no need to transcode, worry about transferring all tags or having to manage a separate collection.
Last post by Arnold B. Krueger -
Are there any companies out there that do not give into all the audiophile crap you see everywhere, but actually base the quality of their products on objective measurements?
Why should any of us care? Audiophiles and even trained engineers vary considerably in terms of their ability to correlate measurements with actual sound quality and IME are mostly on the deficit side.
For example one the the most prolific measurers around, John Atkinson of Stereophile is a proud Golden Ear, and he's hardly alone.
In a vast number of cases, measurements have lead to a specifications race, where adding leading zeros becomes the basis of unfounded claims for sonic quality. For example, almost every piece of modern audio electronics has THD measurements of 0.05% or better, while audibility under the most ideal circumstances starts above 0.1% and may be as high as 10%.
Furthermore, published measurements are tuned by using.g the results of tests where equipment is intentionally pushed into clipping to raise measured distortion to a number that is consistent with marketing goals, when common sense suggests that the correct thing to do would be to decrease the power rating a few watts,.
There's that titleformatting sandbox plugin for foobar and highlight scripts for notepad++ and maybe couple other editors. You might want to look into those.
Thanks, was not aware of foo_tfsanbox.
Last post by Zao -
You have an awful lot of components and DLLs loaded in your crash dump.
I would start to suspect that you're encountering the problem of exhausting TLS slots and that some DLLs fail to load due to this, causing all sorts of havoc in modules that don't expect it.
Please try culling away a bunch of unused components and see if this and the foo_wave_seekbar situation improves.
Is there any way to monitor 2 folders to 2 different playlists? Currently I'm only monitoring one folder and that goes to "All music" playlist. If I add another folder it will also show up on the same playlist.
Can I somehow separate the two in different playlist, while also monitoring both?