People have forgotten what "audio quality" is

RetroComputingCollector said:
Everyone now is just listening to compressed MP3s on low-wattage bluetooth speakers with SBC compression. No-one builds their own Hi-Fi systems with good speakers and amps anymore. Almost no-one listens to lossless or analog anymore. Even though the sound quality is technically worse, I go for analog almost always because of this awful loudness war that is ruining our music. I would not call myself an audiophile because that implies I have magic rocks and $1000 dollar extsnsion cords, instead I would just call myself a person who can't take the sloppy shit that is passed off as "listening" anymore.
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If it's any comfort, I have always listened to music on a legendary set of massive floor speakers with corresponding amps and lossless audio. It's fantastic. You might to change that "no-one" to "one". :)
 
covacat said:
the ai says is something like Extra CMakeFiles but can't verify because shithub is down again
probably this
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Thank you, yes. It's in ports.

But it is then missing a part of Qt6, qtpaths. We only have that for Qt5:
Code: 
CMake Error at /usr/local/share/ECM/modules/ECMQueryQt.cmake:84 (message):
  No Qt6 qtpaths executable found.  Can't check QT_INSTALL_PREFIX as required
 
well current flat tv sound is pc speaker quality
when led tv first came around the sound was passable but now it is crap (maybe the higher end are a bit better but most lgs, samsungs and sonys ive seen are crap -- like a "Panashiba" boombox)
 
Espionage724 said:
I hooked a random speaker up to replace a blown subwoofer on a cheap 2.1 pair out of a thrift store :p
View: https://www.youtube.com/shorts/xNPVCxfVpN4


I don't know much about amp internals, but like the basics of connecting 2 Stereo speakers to a PC and the amp no-frills making the speakers louder. Something like that is a huge upgrade from my TV's HDMI speakers.
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Well done!
Hacking at it's finest :)
 
loveydovey said:
Rant warning. There's no EasyEffects on FreeBSD? You can make any music sound mindblowing with that. That's one shocker about switching to FreeBSD from Linux that I am having a hard time stomaching. Is there something like EasyEffects on FreeBSD? Do you all not want to use FreeBSD to listen to amazing sound?
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Post in thread 'Sound enhancers for freebsd (like old pulseeffects. or like easyeffects in linux)' https://forums.freebsd.org/threads/...r-like-easyeffects-in-linux.98204/post-706199
 
A
RetroComputingCollector said:
Everyone now is just listening to compressed MP3s on low-wattage bluetooth speakers with SBC compression. No-one builds their own Hi-Fi systems with good speakers and amps anymore. Almost no-one listens to lossless or analog anymore. Even though the sound quality is technically worse, I go for analog almost always because of this awful loudness war that is ruining our music. I would not call myself an audiophile because that implies I have magic rocks and $1000 dollar extsnsion cords, instead I would just call myself a person who can't take the sloppy shit that is passed off as "listening" anymore.
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Analog is better because it is a dynamic system that shifts and drifts (spectral composition, filter cutoffs etc due to analog components imperfections that were judged by a musical ear of the builders) and has this physical-like quality. It alsmost tells a story.Many digital vsts have tried to emulate this behaviour but most of the time fall short (and thats imo a good think, life cant be digitized)
 
To be fair, an mp3 at reasonable bitrate does a lot less damage to the sound than -say- a 1980s tape cassette deck.

Standards for media are higher now. Sadly, transducers (loudspeakers) are not clearly better now.
 
Someone said:
A

Analog is better because it is a dynamic system that shifts and drifts (spectral composition, filter cutoffs etc due to analog components imperfections that were judged by a musical ear of the builders) and has this physical-like quality. It alsmost tells a story.Many digital vsts have tried to emulate this behaviour but most of the time fall short (and thats imo a good think, life cant be digitized)
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I have a lot of VST instruments that i use in my production. Some of them claim to have that "analog sound" but, as soon as i hit the keys on my legendary Roland Jupiter 8, i realize how far fetch those claims are. Digital is always going to be digital. Analog sound can not be emulated.
 
I remember pretty well when people were telling that 96kbps MP3's were indistinguishable from the real thing and citing double-blind "studies". Then they moved to 128kbps and continued saying the same thing. They increased it later to VBR & 320kbps.

I can distinguish between 320kbps MP3 & lossless audio with good equipment and real music before the loudness war. And I don't think I have perfect hearing, having occasional tinnitus as the best alert of high blood pressure.
 
MrBSD said:
I have a lot of VST instruments that i use in my production. Some of them claim to have that "analog sound" but, as soon as i hit the keys on my legendary Roland Jupiter 8, i realize how far fetch those claims are. Digital is always going to be digital. Analog sound can not be emulated.
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Misters Nyquist and Shannon agree.
Similar stuff has been going on with photography: digital vs silver halide.
Sampling rate, number of bits, you still have a step function between 2 values.
If you sample fast enough, so the deltas are small enough you can reproduce beyond "normal human hearing/seeing". But graphing the values you still have a sine wave represented by small steps.

A lot of the original work was done for Telephony where you are dealing with the human voice. That work discovered human voice is basically bandpass filtered from roughly 300Hz to 3400Hz. Not very big so sampling rate of roughly "2x" gets you to the standard 8Khz sampling of most voice stuff.
Voice is not Sound. Music has a much greater range of frequencies from the bass/subbass to high notes. Resonant frequencies/harmonics also play a huge role so accurate reproduction needs greater sampling rate than first thought.
Encoding can help overcome some, but that is still just math interpolating between two distinct points.

I always look at digital vs analog like drawing a circle.
I can draw a "circle" with straight lines. Start with a square, go to an octagon, go to 16 sided. Each time the "straight line" portion becomes shorter and shorter, but you only have an approximation of a circle.
Analog to digital, much of the same. Faster sampling means shorter "straight lines" so a better approximation of the analog, but until you get to infinite sampling rate you can only get an approximation, even with interpolating between values.

Bottom line: reproducing the sound on your equipment can you hear the difference.
My opinions only.
 
I admit that audio/music and home theatre are more than a casual hobby to me and that my audio system cost more than some nice small cars (at least new, I bought some components used). It is a 14 channel system (9.1.4) and the main amplifier (there are four mono blocks in addition) does require 240V power (which is very unusual in North America). The system is made in the US and the speakers are made in Canada. I do have a Thorens turntable (TD-126 MK III, IIRC) which I barely ever user. Sure, digital has the loudness wars, but analog records have severe dynamic restrictions, bad stereo separation and high noise. Not even close on a high end system, except for "vinyl nostalgia".

So I am all digital, all lossless. Mostly I am going with CD and DSD (converted to 192kHz/24 bit) audio, all converted to flac audio and stored on my FreeBSD home server. Playback is over HDMI to the audio processor from a FreeBSD HTPC using bit perfect audio (so the computer has nothing to do but get the digital information unaltered to the processor). The same setup also handles passthrough of all surround formats (up to and including DTS:X and TrueHD/Atmos). Everything is OSS only, Pulseaudio is not necessary or beneficial in this context.

Btw, DSD is not inherintly superior to PCM, it all depends on PCM bitrates and sampling.
 
RetroComputingCollector said:
No-one builds their own Hi-Fi systems with good speakers and amps anymore.
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Wrong.

Current system.
Audio Innovations 300 class-A integrated valve amp, also have cyrus, marantz and NAD amps in the cupboard.
Technics CD player.
Snell Type K speakers (genuine Mk 1 originals) and also have some tannoys and monitor audios. I have some lowther mini-monitors I used to listen to a lot but the drive units need a refurb, the rubber surrounds have cracked/perished, unfortunely rather pricey to get it done properly, I'll get round to it one day.
Several different headphone amps and a few sets of mostly sennheiser headphones.
Experimenting with freebsd on thinkpad and external DAC to play lossless FLACs as a source.
And, yeah, a bunch of mp3's because they are readily available.

And a couple of stupid mistakes I made was selling my nakamichi LX-3 and Revox B760, what an idiot. I should have hung on to those, doh! I've got another tape deck around somewhere, a pioneer or something.
 
A little off topic of this "off topic" but this has been one of the most interesting threads I've participated in on these forums.
Stuff that mattered to me a long time ago, the science of accurately reproducing "frequencies", all the different ways you can get from source to "ear".
Modern production of "music" (hey can you autotune me a bit more?) can that really compare to The Righteous Brothers singing live?

good stuff everyone, thanks.
 
Isoux said:
I build and repair Hi-Fi audio equipment as a hobby.
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Me too, I built all kinds of amps, preamps, effects etc over the years, also got involved in bands PA work for a while many years ago. My very first amp was built with ILP audio modules, namely 2-off HY-5 preamps and 2-off HY50 bipolar power amps, back around 1978. These modules were pretty well known in the UK and you could make a nice amplifier with them. Soon after that I started using the hitatchi power mosfets that had just come out, that gave very good results with a relatively simple circuit (and no thermal compensation!). Well.. I've made lots of other stuff, op-amp audio circuts, you know how it goes. I used to build elektor projects too, they had some nice designs. And of course I studied electronics... :-)

Sadly they don't make these ILP modules any more, I think the firm finally folded up a few years ago, but for many years both RS and Farnell used to stock the updated versions of these modules. For a while they had some nice class A mosfet ones, I seem to remember they were called SMOS, I'm not sure whose mosfets they used.

1765999858512.png
 
mer said:
Modern production of "music" (hey can you autotune me a bit more?) can that really compare to The Righteous Brothers singing live?

good stuff everyone, thanks.
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High end, state of the art reproduction, even of old recordings (restored analog to high bit rate lossless transfer) can sound amazing. For example, I have a SACD box of "The Vienna Philharmonics on Holiday" which was recorded in the late 50s and early 60s. Using modern restoration techniques and state of the art audio engineering of the SACD, this sounds absolutely stunning.

Even great audio systems follow the GIGO principle (Garbage In, Garbage Out), but if there is a really good audio source available for the Righteous Brothers, a high end end audio system should be very, very close to a live performance. May be better than the old guys can still do in person today. ;-)
 
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