My 40th Thread...Linux Life

I sort of dropped off away from civilization the past couple of days as been spending much of my time kicking the tires of Manjaro installed on my low powered Dell D620 Laptop. In comparing with my current preferred distro, MX-Linux there is a lot to like of both.

I do like the software management of Manjaro. Seems very initiative. I was able to install a CD/DVD program that worked right from the repo. The same program, months ago with MX-Linux I must have had and older version as had to go into the terminal and use that to install some other parts to get it to work. Also, with Manjaro, seems like their forum is more noobie friendly with their own new folks section. On the other hand, seems to me like with MX-Linux, many of the users there are not so much the new to linux crowd.

As for speed between the two, on my low powered Dell, MX-Linux was definitely quicker. Of course on a more powerful machine the difference may not have been as noticeable.

Playing around, I purposely, messed up the MBR of my Manjaro test computer, then used the MX-Linux tool's Boot Repair from MX-Linux's live CD to fix the MBR. No problem to fix. That's what really shines I think with MX. Swiss army knife.

Anyhow, enough tire kicking for now. I like both distros .
Agree - MX-Linux has quite a comprehensive set of well designed tools and if it wasn't for the sound issue I encountered I would be perfectly happy to use it as my main distro. Also agree with your comment re Manjaro software repos - quite impressive. But I guess that's what I really like about Linux, plenty of choice if one likes to tweak things or straight out of the box with the main distros and best of all - no WIN 10 updates :)
 
Agree - MX-Linux has quite a comprehensive set of well designed tools and if it wasn't for the sound issue I encountered I would be perfectly happy to use it as my main distro. Also agree with your comment re Manjaro software repos - quite impressive. But I guess that's what I really like about Linux, plenty of choice if one likes to tweak things or straight out of the box with the main distros and best of all - no WIN 10 updates :)

How do you feel about the rolling release with Manjaro?

For me, because my main PC is a Windows computer and I don't use Linux that often (only when I want a system to actually work :LOL:), my Linux is probably better with a non-rolling release like MX-Linux. But if I used a Linux machine everyday, going with a rolling with Manjaro is tempting. Kinda of a pay now or pay later thing I suppose as for frequency of updates. Plus I have old hardware and don't want a rolling release update to make my system not work anymore.
 
How do you feel about the rolling release with Manjaro?

For me, because my main PC is a Windows computer and I don't use Linux that often (only when I want a system to actually work :LOL:), my Linux is probably better with a non-rolling release like MX-Linux. But if I used a Linux machine everyday, going with a rolling with Manjaro is tempting. Kinda of a pay now or pay later thing I suppose as for frequency of updates. Plus I have old hardware and don't want a rolling release update to make my system not work anymore.
I've only been using Manjaro for about a month and so far there has been absolutely no problems with any of the updates. From the reading I've done about the rolling releases it seems most users really don't have any problems. My Laptop is a Dell e6440 with an I5 4th gen from 2014 so somewhat ancient by modern standards. But it works great and it's super fast with Linux so I see no reason to upgrade.



I'm not entirely sure but I think that the Linux kernel does not deprecate old software drivers very often so that really old computers still work just fine. My grandson uses an ancient HP desktop from 2005 with a Pentium D processor (Linux Mint XFCE) and it works great he can even play games like Spore and so forth on it.
 
I've only been using Manjaro for about a month and so far there has been absolutely no problems with any of the updates. From the reading I've done about the rolling releases it seems most users really don't have any problems. My Laptop is a Dell e6440 with an I5 4th gen from 2014 so somewhat ancient by modern standards. But it works great and it's super fast with Linux so I see no reason to upgrade.



I'm not entirely sure but I think that the Linux kernel does not deprecate old software drivers very often so that really old computers still work just fine. My grandson uses an ancient HP desktop from 2005 with a Pentium D processor (Linux Mint XFCE) and it works great he can even play games like Spore and so forth on it.

Glad to see that you are not having any problems so far. I wonder if the reports of problems are mainly from folks who are a bit too adventurous using programs outside of the stable branch.

Looks like you have a good laptop that should last you awhile.
 
20-20,000 Hz

I'm reminded of something my Father said to me a long time ago. We must have been discussing audio frequency ranges of amplifiers and speakers, and he said something like, "I can't even hear those higher ranges." Eventually I got to the time of life when sounds beyond 12,000 Hz were not heard either.

But hats off to those seeking bit-perfect audio (https://www.thewelltemperedcomputer.com/KB/BitPerfect.htm).

I'm on the edge of respectability (I hope) with regard to music playback and streaming. I'm listening to Spotify (highly recommend Release Radar channel there). Their audio is streaming at something called Very high quality, “Equivalent to approximately 320kbit/s.” Okay, that's a careful dodge, and I realize the stream is not true CD quality.

This music is also affected by many local features. For instance this is a partial listing from the command line $ inxi -Fxz
Code:
Audio:
  Device-1: Intel 82801I HD Audio vendor: Hewlett-Packard 
  driver: snd_hda_intel v: kernel bus ID: 00:1b.0 
  Device-2: AMD RV710/730 HDMI Audio [Radeon HD 4000 series] 
  vendor: Hewlett-Packard driver: snd_hda_intel v: kernel bus ID: 01:00.1 
  Sound Server: ALSA v: k5.3.0-46-generic
That explains some of the audio handling in a PC or notebook. My devices 1 and 2 have specifications I'll skip over for now. The sound server Advanced Linux Sound Architecture (ALSA) is "... a software framework and part of the Linux kernel that provides an application programming interface (API) for sound card device drivers." Ref: https://en.wikipedia.org/wiki/Advanced_Linux_Sound_Architecture

The wiki reference explains more about the interaction of ALSA and the sound card (or chip). It also mentions two applications using ALSA:
“The sound servers PulseAudio and JACK (low-latency professional-grade audio editing and mixing), the higher-level abstraction APIs OpenAL, SDL audio, etc. work on top of ALSA and implemented sound card device drivers.”
And all of that just gets the system ready to accept an input source such as MP3 file, instrument, or streaming.

My source, Spotify, has settings that impact the audio delivery. For example, the Music Quality can be changed from Auto to one of the following. The default Auto setting hopefully takes me to Very high quality on a wired or WiFi connection. I switched from Auto to Very high with no noticeable change.Low – Equivalent to approximately 24kbit/s
Normal – Equivalent to approximately 96kbit/s
High – Equivalent to approximately 160kbit/s
Very high (Premium only) - Equivalent to approximately 320kbit/s
Another Spotify setting is Normalize. I turned this off since I'm using PulseAudio as my software equalizer. I mentioned earlier that I'm using one PulseAudio Equalizer pre-set called ziyad_perfecteq. It works for me.

From PulseAudio sound leaves the notebook's audio output jack and connects to my PC multimedia speaker system (Altec Lansing ACS340). That's 2.1-channel with a subwoofer. Output is 40 Watt with Frequency Response 30 - 20000 Hz. I leave the dial controls at neutral settings.

My setup is different than what's been discussed by true audiophiles above. Their approach strives for bit-perfect-ness, and uses an application (like deadbeef) which passes the raw quality of sound file to external devices such as DAC (Digital-to-Audio Converter), amplifier, and high-quality speakers. Or at least I'm hoping that's how things work!
 
Their approach strives for bit-perfect-ness, and uses an application (like deadbeef) which passes the raw quality of sound file to external devices such as DAC (Digital-to-Audio Converter), amplifier, and high-quality speakers. Or at least I'm hoping that's how things work!

Yes, you are correct :)
 
.. Their approach strives for bit-perfect-ness, and uses an application (like deadbeef) which passes the raw quality of sound file to external devices such as DAC (Digital-to-Audio Converter), amplifier, and high-quality speakers. Or at least I'm hoping that's how things work!

I'm not so sure you need to go to any extremes to achieve this. Seems to me, in a basic set up, the app passes the data to the DAC, the DAC processes it. As long as you are not inserting an equalizer or other effect, why wouldn't it be "bit perfect"? Seems it would take effort (or an error - like the CPU not being able to keep up with the data stream) to mangle it.

I question their idea that you can record the output and compare recording to source bit-for-bit. Unless your DAC is perfect (at least to that bit level), quantization errors will creep in. And dithering will be applied somewhere in the process, and since that involves somewhat random noise (it may be filtered), I would not expect bit-for-bit response anyhow. The differences, if all steps are done with good-enough quality, will not be apparent to most (if any).

I'm pretty picky (mostly because it's easy these days) - I record everything in FLAC (lossless). With storage so cheap, why bother even taking a chance that the sound is degraded?

-ERD50
 
A couple weeks ago, I decided that I was going to do something with the old Supermicro CSE-512 dual Xeon pizza box I've had sitting in my basement for maybe 8 years doing nothing. It had 4GB ECC memory (2x2GB). I originally intended to install a Windows OS on it, but it was not cooperating whatsoever. So instead I ended up installing Ubuntu with Microsoft's free SQL Server for Linux. I pulled out the 2x2GB memory, and picked up 64GB ECC (8x8GB) on Ebay for under $30 - it was cheaper to take 64GB rather than just 6x8GB to fill the 6 slots the board has.

Ubuntu works just great on the box. No issues whatsoever. Handles the multi-cpu just fine and it might be almost impossible to utilize the 48GB memory even if I pinned the database in memory.

The free Linux version of MS SQL Server is also pretty sweet. I have old SQL Server 2008 client tools and homegrown apps on my Windows desktop machines and they interface just fine with the brand new Linux SQL Server - no modifications necessary whatsoever - just point it at the new server. Additionally, databases on other Windows versions of MS SQL Server were easily migrated on to the Linux version with a couple button clicks, no compatibility issues whatsoever.

Now as far as the pizza box - it's got these dual 12,000 RPM 50 decibel counter-rotating fans in it that sound like jet engines. Thankfully the machine is in the basement and I really don't care. However, DW has her sewing room down there so I have a feeling that we're going to have to do some timesharing when she's down there or I'm going to have to install some sound dampening panels around the rack.
 
I'm not so sure you need to go to any extremes to achieve this. Seems to me, in a basic set up, the app passes the data to the DAC, the DAC processes it. As long as you are not inserting an equalizer or other effect, why wouldn't it be "bit perfect"? Seems it would take effort (or an error - like the CPU not being able to keep up with the data stream) to mangle it.

-ERD50

Linux has something called pulseaudio, think of it as a mixer, it mixes sounds from web browser(s), music app(s), linux sounds (chimes, notification chimes) etc... then sends to ALSA driver, pulseaudio does up sampling or down sampling depends on the pulseaudio setting for your audio interface, very often it is set for the max sampling rate your hardware is capable of (for example, 24bit at 192khz).

If you play FLAC file CD quality 16bit at 44.1khz, pusleaudio will up sampling it to 24bit at 192khz before it sends to the ALSA layer. This is not bit perfect, once could argue so what? it's still good. However, the up sampling process does add noise and cracking sounds on some system (old hardware). Therefore, people look for the way to by pass the mixer (pulseaudio), and go straight to the ALSA layer -> the DAC. Some music app allows you to by pass pulseaudio(deadbeff is one), most don't.

An easy test if you have a DAC that can display the sampling rate, you can play song with 44.1khz sampling rate, song with 96khz sampling rate, song with 192khz sampling rate. If your DAC shows the correct sampling rate, you are playing with bit-perfect. If your DAC is staying at the same rate, you are not, pusleaudio is up or down sampling your audio to a fix pulseaudio setting.

Same thing under Windows, it does have mixer that mixes all the sounds then sends to driver. And people have to do something similar (WASAPI exclusive mode)

If you are not picky, then just ignore bit-perfect as long as you are happy. If you can tell the different between mp3 and CD and super audio, I think you should try to explore bit-perfect. I am happy with my bit-perfect (ALSA->DAC->Amp) and with an old weak laptop.
 
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An easy test if you have a DAC that can display the sampling rate, you can play song with 44.1khz sampling rate, song with 96khz sampling rate, song with 192khz sampling rate. If your DAC shows the correct sampling rate, you are playing with bit-perfect. If your DAC is staying at the same rate, you are not, pusleaudio is up or down sampling your audio to a fix pulseaudio setting.
PulseAudio is an application I had to add in mint. Might be included with other distros.

Below in the app title bar you can see what's going on...
 

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Linux has something called pulseaudio, think of it as a mixer, it mixes sounds from web browser(s), music app(s), linux sounds (chimes, notification chimes) etc... then sends to ALSA driver, pulseaudio does up sampling or down sampling depends on the pulseaudio setting for your audio interface, very often it is set for the max sampling rate your hardware is capable of (for example, 24bit at 192khz).

If you play FLAC file CD quality 16bit at 44.1khz, pusleaudio will up sampling it to 24bit at 192khz before it sends to the ALSA layer. This is not bit perfect, once could argue so what? it's still good. However, the up sampling process does add noise and cracking sounds on some system (old hardware). Therefore, people look for the way to by pass the mixer (pulseaudio), and go straight to the ALSA layer -> the DAC. Some music app allows you to by pass pulseaudio(deadbeff is one), most don't.

An easy test if you have a DAC that can display the sampling rate, you can play song with 44.1khz sampling rate, song with 96khz sampling rate, song with 192khz sampling rate. If your DAC shows the correct sampling rate, you are playing with bit-perfect. If your DAC is staying at the same rate, you are not, pusleaudio is up or down sampling your audio to a fix pulseaudio setting.

Same thing under Windows, it does have mixer that mixes all the sounds then sends to driver. And people have to do something similar (WASAPI exclusive mode)

If you are not picky, then just ignore bit-perfect as long as you are happy. If you can tell the different between mp3 and CD and super audio, I think you should try to explore bit-perfect. I am happy with my bit-perfect (ALSA->DAC->Amp) and with an old weak laptop.

Where do I find these settings for Pulseaudio, I don't see them in the interface? It seems odd to me that it would attempt to upscale a 44.1k/16b file. Why would it do that? I suppose if it sees you have a 192k/24b DACs it might go ahead and interpolate those intermediate values, which may or may not improve the audio experience.

My DAC (NuForce uDAC) is an older model, specs say USB native bit rate 32 - 48 KHz, 16 bit. Max 48KHz, 24 bit resolution (I assume they mean accuracy?). No support for higher rates, so I can't query it in the way you describe anyhow (unless you think my system might be upscaling from 44.1 to 48?).

But I really doubt any of this would make any difference at all anyhow. Any artifacts from upscaling a 44.1k/16b to 192k/24 would be extremely minor. Far, far less than using mp3 compression - that's a whole different ball game, not a valid comparison at all.

If you are getting "crackling" or pauses in the output, that's a limitation in your system somewhere, and has nothing to do with any quantization effects from upscaling/ Though upscaling, if it is being done, could over-tax an already taxed system to the point of failure like this. In that case, it really should be turned off. Totally different from any "bit perfect" comparisons. That's "bit broken".

PS - I did find this in /etc/pulse/daemon.conf :

; resample-method = speex-float-1
; default-sample-format = s16le
; default-sample-rate = 44100
; default-sample-format = s16le
; default-sample-rate = 44100
; alternate-sample-rate = 48000

That's on my main system, I actually use a very old 2009 ASUS netbook to drive the DAC on my sound system. Maybe I'll boot that up and look later.


-ERD50
 
PulseAudio is an application I had to add in mint. Might be included with other distros.

Below in the app title bar you can see what's going on...

According to your image, your pulseaudio is set to 16 bit 44.1khz, and pulseaudio will down sampling hi-res music (24 bit 192 khz or higher) to 16 bit 44.1khz. You would not care if you don't play any hi-res music.
 
PulseAudio is an application I had to add in mint. Might be included with other distros.

Below in the app title bar you can see what's going on...

Wait a minute - you are concerned about "bit perfect" and then turn on the Equalizer? That equalizer is messing with the bits far, far more than any up/down re-sampling would.

-ERD50
 
According to your image, your pulseaudio is set to 16 bit 44.1khz, and pulseaudio will down sampling hi-res music (24 bit 192 khz or higher) to 16 bit 44.1khz. You would not care if you don't play any hi-res music.
After my comment I played a CD to see what, if anything, would change in the PulseAudio title bar. This is helping me mentally flow chart what's going on.

16LE (bit depth) is straight from the PA conf. It can be changed.

Possible entries for the sample format are: u8, s16le, s16be, s24le, s24be, s24-32le, s24-32be, s32le, s32be float32le, float32be, ulaw, alaw
Ref: https://askubuntu.com/questions/138611/how-to-change-audio-bit-depth-and-sampling-rate

Unless one changes this in the conf, PA uses s16le as audio bit depth.
 
Wait a minute - you are concerned about "bit perfect" and then turn on the Equalizer? That equalizer is messing with the bits far, far more than any up/down re-sampling would.

-ERD50
You're looking for argument where there is none.

I'm not concerned about bit depth, just responding to what an audiophile has mentioned in this thread. It's all good--another opportunity for me to learn.
 
You can check your current playing sampling rate with this cmd (=

I am playing a CD quality song 16 bit 44.1khz

$ sudo cat /proc/asound/card*/stream0
Focusrite Scarlett 2i2 USB at usb-0000:00:14.0-6, high speed : USB Audio

Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 72
Momentary freq = 44100 Hz (0x5.8333) <=======
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (SYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24

....

playing a hi-res song 24bit 192khz

$ sudo cat /proc/asound/card*/stream0
Focusrite Scarlett 2i2 USB at usb-0000:00:14.0-6, high speed : USB Audio

Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 200
Momentary freq = 192000 Hz (0x18.0000) <=====
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (SYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
 
I'm playing an audio CD with rhythmbox. Nothing else "involved."

$ sudo cat /proc/asound/card*
cat: /proc/asound/card0: Is a directory
cat: /proc/asound/card1: Is a directory
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xda100000 irq 33
1 [HDMI ]: HDA-Intel - HDA ATI HDMI
HDA ATI HDMI at 0xda010000 irq 34

Using your command gives file not found.
 
I'm playing an audio CD with rhythmbox. Nothing else "involved."

$ sudo cat /proc/asound/card*
cat: /proc/asound/card0: Is a directory
cat: /proc/asound/card1: Is a directory
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xda100000 irq 33
1 [HDMI ]: HDA-Intel - HDA ATI HDMI
HDA ATI HDMI at 0xda010000 irq 34

Using your command gives file not found.

Can you try? while playing something


sudo ls /proc/asound/card*

or

sudo find /proc/asound/card*/ -name stream*
 
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$ sudo cat /proc/asound/card*/stream0

cat: '/proc/asound/card*/stream0': No such file or directory


$ sudo ls /proc/asound/card*
/proc/asound/cards

/proc/asound/card0:
codec#0 id pcm0c pcm0p pcm1p

/proc/asound/card1:
codec#0 eld#0.0 id pcm3p
 
Looks like your audio interface does not provide stream0

Let's try this while playing something

sudo cat /proc/asound/card0/pcm0p/sub0/hw_params
 
Looks like your audio interface does not provide stream0

Let's try this while playing something

sudo cat /proc/asound/card0/pcm0p/sub0/hw_params

$ sudo cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 8192
buffer_size: 16384
 
$ sudo cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 8192
buffer_size: 16384

With your current setting, I think it will stay at the 44100 rate even if you are playing a hi-res song. However, with the new version of pulseaudio (> v 11) you now could disable pulseaudio resampling (this is new to me too :)

"It is possible to disable resampling by simply adding the avoid-resampling = yes line in the /etc/pulse/daemon.conf file if you have PulseAudio 11.0 installed. "

https://www.soundphilereview.com/ne...allows for,you have PulseAudio 11.0 installed.

You can download the test audio files here to test

2L High Resolution Music .:. free TEST BENCH
 
You're looking for argument where there is none.

I'm not concerned about bit depth, just responding to what an audiophile has mentioned in this thread. It's all good--another opportunity for me to learn.

OK, I guess I misunderstood where you were going with this. I'm also curious, so I'll hang in and try to learn as well.

The kind of funny thing is, the Linux machine I use for my main sound system is a 2009 ASUS netbook that I think I paid ~ $250 for back then. It has such limited memory and SSD space, I don't dare try to update it. (though I'm sure I could - but the music player app works, so why bother?). Since it's been ages since security updates have been issued, I just turned off all the networking. I'm just curious to see how long this thing will go on doing its thing.

-ERD50
 
With your current setting, I think it will stay at the 44100 rate even if you are playing a hi-res song. However, with the new version of pulseaudio (> v 11) you now could disable pulseaudio resampling (this is new to me too :)

"It is possible to disable resampling by simply adding the avoid-resampling = yes line in the /etc/pulse/daemon.conf file if you have PulseAudio 11.0 installed. "

https://www.soundphilereview.com/ne...allows for,you have PulseAudio 11.0 installed.

You can download the test audio files here to test

2L High Resolution Music .:. free TEST BENCH
I've been confusing pulseaudio with pulseeffects...I was thinking pulseaudio was turned off when I quit the app pulseeffects.

  • ALSA is the kernel based system.
  • PulseAudio sits on top of ALSA and extends it with additional modules.
  • PulseEffects is an app which adds filters and manipulation to the sound system in use.
Found my pulseaudio version:
$ pulseaudio --version
pulseaudio 11.1

Found my pulseaudio package:
$ dpkg -l pulseaudio
Desired=Unknown/Install/Remove/Purge/Hold
| Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend
|/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad)
||/ Name Version Architecture Description
+++-==============-============-============-
ii pulseaudio 1:11.1-1ubun amd64 PulseAudio sound server

So I flunked the pop quiz, but know I can pass the test later in the semester.
:(

I'm gonna disable re-sampling in pulseaudio on a clearer day.
 
Manjaro makes it easy to try out different Desktop environments and choose one at login. KDE, Gnome 3, Mate, XFCE, LDXE, and others are available with a few pacman commands.

My favorite is Mate (really Gnome 2) with Ubuntu fonts and themes. Add in some Compiz for wobbly windows. Import the Win 10 fonts so they are available.
 
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