Round 01 Arrangement
W.W.R. Lyrics
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E drop
0:41 – LOSSY Compression
...MP3
....AAC
.....OGG
|
2:38 – LOSSLESS Compression music
...FLAC
....ALAC
|
3:55 – LOSSLESS Compression _movies _
...Dolby True HD
....DTS HD
.....Dolby Atmos
|
4:30 – UNcompressed
...WAV
....AIFF
David Haggerty
I have a question that is somewhat related, in that it deals with sound processing, but I was having difficulty finding a video that related directly to my question, so I figured I'd put it out there (here) and see what happens.
I recently added an external sound processor to my configuration to give me better sound performance and reduced latency, whether it be in a digital audio workstation (Ableton), day to day operation, or both. I know this isn't the typical place to look for answers like this, but I've already familiarized myself with much of the information put out by production software makers, etc and I'm actually looking for some perspective from a different angle. The sound processor does a great job and with the right software I am able to run my DAW with ASIO driven sound while doing other things, like watching tutorials simultaneously, thanks to the help of Voicemeeter Banana (phenomenal software for in depth control of your computer's internal audio routing/processing btw, even if you don't use production software). The caveat is that I get intermittent pops, at random infrequent times, due to the nature of low latency sound being interrupted by CPU processes and not having the typically large buffer to compensate for the audio drop when the interrupt occurs. I have tried Kernel Stream, MME, and WDM sound configurations and it still occurs (seemingly) randomly. The only configuration that I found eliminated the drops was actually using Voicemeeter in an ASIO configuration. While this works wonderfully for sound quality, it somewhat defeats the purpose of having voicemeeter, because it just feeds the audio through at whatever value it is received which doesn't allow me to have the same level of control over gainstaging etc. except what is set in the software , itself.
I'm basically wondering if you have any knowledge of how to measure and reduce latency to an outboard processing unit (audio being my preference obviously, though I imagine many of the same rules hold true for other gear) that is passed over a USB connection?
Some factors that I feel are probably often overlooked, but relevant and I've been considering are...
1.)How to measure and test the prioritization of the different USB ports within your hardware configuration and operating system's framework (Windows 8.1 64 bit with 5 USB ports which I believe are 3 x 2.0 ports and 2 x 3.0 ports) and the only other static USB load is my wireless mouse and keyboard which are both connected via a single logitech unifying receiver.
2.) Do redundant drivers trying to access the same signals cause feedback loop or other types of conflicts and are there any reliable ways to eliminate them? I've been trying to eliminate unnecesssary processes like the computer's native mouse handling software while the has its own driver/software. I've also shut down realtek drivers since my sound is being processed by voicemeeter's drivers and my soundprocessor's drivers (Behringer UMC404HD).
3.) How do things like Phillips Hue bulbs effect bandwidth and network activity? I have 3 of them being controlled by the Hue Disco app on my tablet (Hue Disco is another really fun product to look into if you haven't already.)
4.) I've noticed Chrome has been hanging up more frequently for some reason lately, though I feel like it may be related. I haven't had to restart it for a while ( unless I chose to), but it still freezes for 1 or 2 minutes at very seemingly random times, before continuing playback and it's not buffering because there is no loading circle and the computer thinking indicator comes on the cursor. The System Interrupt is usually the highest process at these times according to task manger, though it's only using 2 percent of the CPU as its reported. Any thoughts on why this is occurring, how to identify and or control the interrupts, or different plugins and/or browsers that would help improve or likely cause the problem would be greatly appreciated.
Thank you, for anyone who takes the time to read this, especially if they can answer any/some of the questions or redirect me to a place where I am likely to find them.
IncapableLP
A little hint:
The licenses for MP3 are running out, so the Mp3 CODEC-FORMAT (important to differentiatew ;) ) is now free.
The Frauenhofer Institute, who invented MP3, actually recommends to use more modern codecs, like Vorbis or the ACC now.
I for my part use FLAC, but whatever ;)
DISCLAIMER:
That doesn't mean, that things get free, because some of the implemetations for MP3 use software, that was developed by Frauenhofer and THAT is not free!
Miguel Escutia
The "old" MP3 player that they show is actually a current lossless model, FIIO X3-II which also works as DAC and it plays everything you throw at it.
zdenek bradacek
DAC sound
Kip
how big is your DAC
Nerd254
@Kip it ain't floppy, that's for sure
Mpivovitz
@Nerd254 are you saying you a have a 3.5" hard disk?
Nerd254
@Mpivovitz nah mine is 8 inches 😎 old drives from the 90s :^)
Ender W.
AAC should be standard rather than mp3 if it offers higher quality at lower bitrate. In a lot of cases, it would result in smaller filesize while maintaining similar or higher quality than its mp3 counterpart.
wujekcientariposta
It's probably based around an algorithm that cuts out less data, so even if it's better quality at a lower bitrate the relative size will be about the same. That said mp3 is shit, it's popular becouse it was already popular and majority of people listen to music on a 2 dollar headphones and don't even know the difference between sound file formats. If you do have like a entry level auudiphile sound system you should be able to tell about 7-8/10 sound formats of your music just by listening.
buster colin
@wujekcientariposta so if mp3 is shit, what Isn't?
Stavex
buster colin flac, is good