Audiophile options for a HQ external USB DAC

mixer monitor 0
Did the command above didn't solve it?

You can try:
mixer -f /dev/mixer0 -s mix 0
mixer -f /dev/mixer1 -s mix 0
mixer -f /dev/mixer2 -s mix 0
mixer -f /dev/mixer3 -s mix 0

Avoid (don't do!):
mixer -f /dev/mixer4 -s mix 0
 
mixer monitor 0 did not solve

i have:
mixer -f /dev/mixer0 -s mix 0 mixer: unknown device: mix usage: mixer [-f device] [-s | -S] [dev [+|-][voll[:[+|-]volr]] ... mixer [-f device] [-s | -S] recsrc ... mixer [-f device] [-s | -S] {^|+|-|=}rec rdev ... devices: vol, pcm
it looks it is new mixer ( there was rewrite of it )
 
mixer monitor 0 did not solve

i have:
mixer -f /dev/mixer0 -s mix 0 mixer: unknown device: mix usage: mixer [-f device] [-s | -S] [dev [+|-][voll[:[+|-]volr]] ... mixer [-f device] [-s | -S] recsrc ... mixer [-f device] [-s | -S] {^|+|-|=}rec rdev ... devices: vol, pcm
it looks it is new mixer ( there was rewrite of it )
Please do all these commands after each other:

mixer -f /dev/mixer0 0
mixer -f /dev/mixer0 pcm 0
mixer -f /dev/mixer1 0
mixer -f /dev/mixer1 pcm 0
mixer -f /dev/mixer2 0
mixer -f /dev/mixer2 pcm 0
mixer -f /dev/mixer3 0
mixer -f /dev/mixer3 pcm 0
 
this worked
bitperfect also now working. thats quite and achievement! thank you very much!
You're welcome.
question: are these mixer changes permanent?
No, you can always reconfigure them as you wish.
But if you don't change them, they'll be remembered.

My experience with bit-perfect on FreeBSD is that cmus does not play 24-bit files but only (the standard) 16-bit files. (I'm talking about lossless FLAC and APE files)
Audacious can play all file types (also FLAC 24-bit 96kHz) in bit-perfect mode, so I would recommend this audio player.
 
i am quite familiar with audio/musicpd so this is my player choice. I also meant that applied changes to mixer are remembered but used wrong word, sorry. I still shocked this was that simple :)
 
i am quite familiar with audio/musicpd so this is my player choice. I also meant that applied changes to mixer are remembered but used wrong word, sorry. I still shocked this was that simple :)
mpd may also be my favorite app but it takes a little longer to configure. I'm also glad it's resolved for you :)
 
I installed mpd today. I used to use it all the time but hadn't used it for a few years.

So what I noticed is that Audacious gives a little more volume at 100% volume on both players. I then adjusted this by giving 2% more volume to mpd in the mixer.

Then I compared the sound quality. I would think it's the same with bit-perfect settings, but it wasn't. The song I played didn't have that much bass, but the bass had more impact on Audacious than it did on mpd.

Then I played the following song on both players: https://opbeatz.beatstars.com/beat/hourglass-1680975

It sounds better in Chromium and Audacious than on mpd. The bass is less tight and less detailed on MPD. Maybe I've misconfigured mpd and that's why it doesn't sound as good. I'll investigate that further.

It could also be that Audacious has better sound for one reason or another: https://www.reddit.com/r/Ubuntu/comments/ado6kl/audacious_sound_with_a_gnome_music_like_interface/

I'm not an audiophile but I find the quality sound of the audacious music player much better than other music players I tried.
 
Audacious and MPD sound slightly different on my system. I researched it further and compared a few more tracks. Audacious has a bass that is more punchy. But MPD has more purity in the voice and instruments.
 
1. i have to say, sadly to my too early excitement, that disabling of mixers we did with above commands, pcm4 became my main device.
cat /dev/sndstat:
pcm4: (play/rec) default
and i had dev.pcm0 set to bitperfect. now that all other mixers disabled, i set dev.pcm.4.bitperfect=1 and it produces distorted sound, hiss and noise. so i was wrong in my comments, wrongly assuming dev.pcm.0 has worked ( it was main device before disable of other mixer devices )
2. i find it very weird, that in bitperfect mode audacious and musicpd plays differently. it cannot be so, in my understanding. either one or second doing something wrong with sound ( equalizer or similar )
 
Audacious has a more punchy and more emphasized bass, MPD has better vocals. For me it is definitely correctly configured on the right output. They both sound very good but there are subtle differences.

I don't know why it won't work for you, I think a developer will be able to help you better, I don't have detailed knowledge of FreeBSD's audio stack.
 
i set dev.pcm.4.bitperfect=1 and it produces distorted sound, hiss and noise
Bitperfect mode sends an audio stream directly, without any processing, to a device. In your case to pcm4, likely your DAC. If you get hiss and noise, you may be sending a format or a sample rate your DAC cannot handle.
 
how to find out which format/sample which DAC can handle?
Most of all from the manual and specifications of the DAC and its settings, if any. Ffprobe (part of the ffmpeg), by using ffprobe audio.file, gives you sample rate, no. of channels and bit depth of an audio file.
Remember that audio applications may alter audio before sending it to an output device. Command cat /dev/sndstat will show you these data as it was last sent.
 
Audacious has a more punchy and more emphasized bass, MPD has better vocals. For me it is definitely correctly configured on the right output. They both sound very good but there are subtle differences.

I don't know why it won't work for you, I think a developer will be able to help you better, I don't have detailed knowledge of FreeBSD's audio stack.

Im not an expert in sound but me too, I find audacious with better sound, if you are interest I made a patch to have 31 bands on audacious

audacious hack

if you like I cant update the post for the actual version
 
On previous discussion DACs vs onboard; it's not just about the chip, there is an amplifier and analog circuitry. The baseline price for good retail USB DAC one channel (stereo) is about 50e. The price of chip itself is about few euros at volume, which is a lot more pricey than the DAC part of the average mobo soundcard which just has to be sub dollar considering $100 retail mobos have them. Indeed, if you take a look at interfaces utilizing stereo DAC and some additions, you'll come to a baseline price of about 100 euros, for example PiSound which has a Burr Brown with MIDI, microphone input, and two pots.

It would be absurd to claim that a mainboard that costs the same price as stereo interface has the same quality interface on board, and it's also multichannel.

Quality headphones are in audio world what gaming laptops are in gaming world. You pay for the ergonomics, not for the performance. Onboard analog circuitry is just not good enough to drive a proper speaker system. For me it is appropriate for home Hi-Fi tops, and only if you have good DAC on the receiver side so you can just use digital out from the computer. But that still counts as using (and paying for) separate DAC.
 
Good day!
I apologize for my English googletranslate :)
I am new to the FreeBSD community, although I have been working with linux for quite a while.

I want to use my usb dac Topping E50 c MiniPC on FreeBSD-14.1 to play music from spotifyd application. But I got this problem: as soon as I enable the options "dev.pcm.5.bitperfect=1" or "hw.snd.report_soft_formats=0" the spotifyd application (portaudio) crashes.

In this regard, please advise me the optimal sound settings without the options "dev.pcm.5.bitperfect=1" and "hw.snd.report_soft_formats=0", so that the sound passes to usb-dac unchanged and without resampling in the original quality. Very much hope for your help. Thanks!!!

Here are the current data and settings of my system:

dmesg | grep uaudio
uaudio0: <Topping E50, class 239/2, rev 2.00/1.07, addr 3> on usbus0
uaudio0: Play[0]: 384000 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer. (selected)
uaudio0: Play[0]: 352800 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.
uaudio0: Play[0]: 192000 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.
uaudio0: Play[0]: 176400 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.
uaudio0: Play[0]: 96000 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.
uaudio0: Play[0]: 88200 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.
uaudio0: Play[0]: 48000 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.
uaudio0: Play[0]: 44100 Hz, 2 ch, 32-bit S-LE PCM format, 2x2ms buffer.

loader.conf
hw.vga.textmode=0
snd_uaudio_load="YES"
#snd_driver_load="YES"
coretemp_load="YES"
amdtemp_load="YES"
sysctlinfo_load="YES"

device.hints
hint.pcm.5.eq="0"
hint.pcm.5.vpc="0"

/etc/sysctl.conf
hw.snd.verbose=2
hw.snd.default_auto=0
hw.snd.default_unit=5
#dev.pcm.5.play.vchans=0
#dev.pcm.5.bitperfect=1 ### crashed spotifyd
#hw.snd.vpc_0db=100
#kern.timecounter.alloweddeviation=0
hw.usb.uaudio.buffer_ms=2
hw.snd.maxautovchans=0
hw.snd.latency=0
hw.snd.feeder_rate_round=0
#hw.snd.report_soft_formats=0 ### crashed spotifyd
hw.snd.vpc_mixer_bypass=0

cat /dev/sndstat
pcm5: <Topping E50> on uaudio0 (1p:0v/0r:0v) default
snddev flags=0x200000e7<SIMPLEX,AUTOVCHAN,SOFTPCMVOL,BUSY,MPSAFE,REGISTERED,PRIO_WR>
[pcm5:play:dsp5.p0]: spd 44100, fmt 0x00200010/0x00201000, flags 0x0000112c, 0x00000023, pid 1443 (spotifyd)
interrupts 125150, underruns 0, feed 125149, ready 8192 [b:1424/712/2|bs:8192/2048/4]
channel flags=0x112c<RUNNING,TRIGGERED,SLEEPING,BUSY,HAS_SIZE>
{userland} -> feeder_root(0x00200010) -> feeder_format(0x00200010 -> 0x00201000) -> feeder_volume(0x00201000) -> {hardware}
 
The plain and simple, also ugly truth is that you don't need an USB DAC normally, because the DACs which are put on mainboards are more than capable of doing their job nowadays.

Only reasons where it is desireable to get one is this:

1. if the audio coming from mainboard is pestered with hisses and interferences, which you cannot get rid of.
2. if you have got a high impedance set of headphones, like 600 Ohms, while your audio output only can handle let's say 32, which means that volume becomes a problem. Which is not a DAC issue anyway strictly, but of amplification.

It's way more effective to invest the money, which you would use for a DAC for, into your headphones.
I've had very good results using the motherboard chipset (actually a bog-standard realtek ALC269) in bitperfect mode, with an external amplifier to drive high impedance headphones (https://www.canford.co.uk/CANFORD-PC-HEADPHONE-AMPLIFIERS) and some nice sennheisers. That little amp is based on a TI TS924 bicmos quad op-amp https://www.st.com/resource/en/datasheet/ts924.pdf and can drive headphones with impedance ranging from 25 to 2k ohms. Whether this qualifies as 'audiophile grade' is another question ... in point of fact it probably does not :), but it still sounds very nice to my ears for not a lot of money. Of course there are probably better headphone amps out there. I did try a mate's expensive external usb soundcard for a weekend out of curiosity but it didn't sound any better to my ears, put it this way, I didn't rush out and buy the usb dac.

Interesting article here https://www.tomshardware.com/reviews/high-end-pc-audio,3733-19.html
Quote of the day: "A $2 Codec Sounds (to us) like a $2000 Device"

"Using world-class headphones, a $2 Realtek integrated audio codec could not be reliably distinguished from the $2000 Benchmark DAC2 HGC in a four-device round-up. "

As is usual with hi-fi, the toms article produced a long discussion on ASR here :-
 
I've had very good results using the motherboard chipset (actually a bog-standard realtek ALC269) in bitperfect mode, with an external amplifier to drive high impedance headphones (https://www.canford.co.uk/CANFORD-PC-HEADPHONE-AMPLIFIERS) and some nice sennheisers. That little amp is based on a TI TS924 bicmos quad op-amp https://www.st.com/resource/en/datasheet/ts924.pdf and can drive headphones with impedance ranging from 25 to 2k ohms. Whether this qualifies as 'audiophile grade' is another question ... in point of fact it probably does not :), but it still sounds very nice to my ears for not a lot of money. Of course there are probably better headphone amps out there. I did try a mate's expensive external usb soundcard for a weekend out of curiosity but it didn't sound any better to my ears, put it this way, I didn't rush out and buy the usb dac.

Interesting article here https://www.tomshardware.com/reviews/high-end-pc-audio,3733-19.html
Quote of the day: "A $2 Codec Sounds (to us) like a $2000 Device"

"Using world-class headphones, a $2 Realtek integrated audio codec could not be reliably distinguished from the $2000 Benchmark DAC2 HGC in a four-device round-up. "

As is usual with hi-fi, the toms article produced a long discussion on ASR here :-
what are details on your setup for headphone amp? i.e how the sound goes from your motherboard chipset to the headphone? what headphones do you use?
Speaking of audiophile world. I also find it to be way too crazy. I have thinkpad t480 with realtek acl257 chip and it works just fine in bitperfect mode. I use simple cheap wired earbuds. The sound is amazing and i can hear details in music that i never able to hear on some other setups. Mind that bitperfect setup on FreeBSD is not a straightforward task as many audiophile-grade ( or so they call it ) devices do not actually provide needed options or making some adjustments that are not documented. I had Schiit Fulla 2, tiny usb dac and i struggled to death making it working in bitperfect mode. Apparently their USB setup is not fully transparent and FreeBSD did the right thing refusing to work in bitperfect mode. What is really comes to high grade quailty is:
- correct power management ( is where amplifier can help ) and
- good headphones.
That's all. All modern onboard chips are able to provide very good quality sound beyond human ear capabilities.
I plan doing upgrade in near future and getting good headphones and headphone amplifier. Nothing other than that, chip will be onboard one.
 
Sound goes from mobo line out (NOT the headphone socket) to headphone amp input. I've got a few sets of heaphones, my best ones are massdrop sennheiser HD6XX, I've also got 560S, 599SE, a couple of other older pairs. The 6XX's do sound the best, but even the cheaper sets sound pretty good. You can connect the thinkpad headphone output into that headphone amp input too, I've done it that way with a thinkpad, unfortunately there's no line out on the thinkpad although the codec chip does support it. Of course line out is preferable so that you bypass the on-board power amp and get the output of the DAC directly.

An external headphone amp that can drive high impedance headphones is a nice addition, usually the mobo onboard audio amp is pretty low power and designed for cheap low-imp headphones or earbuds, in some realtek codecs the amp is integrated on chip, if you actually look at the tiny size of the SMD realtek chip on the mobo you can imagine that the output transistors and output power are going to be pretty small(!) no matter what the datasheet claims, so an external headphone amp gives you much better drive capability and a volume knob! :) That canford one is just an example, there are lots of small headphone amps available. I know the canford one was designed for broadcast studio use (BBC, apparantly) so it should be pretty good, but I'm sure there are better ones out there. ASR is a good site, they do real engineering analysis of amplifiers and show you the results along with build teardowns, have a look at this page to get some ideas


Just don't buy something like this!


I've also used a thinkpad headphone output as a hifi source with bitperfect mode and FLAC audio, going into a valve amp and some good spkrs. It's not bad, although I think my nakamichi LX5 playing some good recordings taken from a linn turntable and shure cartridge beats it (analog... :))

As I said this stuff is unlikely to be up there with true 'audiophile' kit but it sounds pretty good to me for the money spent. And as the Tom's article said, when you do blind listening tests, even true audiophiles can't always tell the expensive kit from the onboard dac.
 
One more thing I just remembered, I power my headphone amp from a linear PSU instead of the cheap switch-mode wall wart thing that they usually use; really just to elminate the possibility of high frequency switching noise getting into the headphone amp. I don't know if it's really any better than using the switch-mode psu though. But I had one lying around so I thought I may as well use it.
 
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