[Anda]

Quote:

So two options left: a newer PC, or a USB2SPDIF device, I guess.

 
Are you located in the US? Go get the V-link192 while the price is still $199:
http://www.amazon.com/Musical-Fidelity-V-LINK192-SPDIF-Converter/dp/B00718X2TU
http://www.audioadvisor.com/prodinfo.asp?number=MFVL192
 
I have the Hiface Two and the XMOS chip works great in Linux. In addition V-link192 has galvanically isolated USB input.

 

[qayws]

Well, no, in  Switzerland. Here the Vlink 192 cost almost double!! But anyway I need the Vlink-II since the DAC won't accept more than 96kHz anyway, and this one is cheaper (I will then have a trouble playing 192k files then! More headaches to come! ) ... or the M2Tech, that's an option too.
 
Thanks again for everything!

 

[Anda]

Quote:

Well, no, in  Switzerland. Here the Vlink 192 cost almost double!! But anyway I need the Vlink-II since the DAC won't accept more than 96kHz anyway, and this one is cheaper (I will then have a trouble playing 192k files then! More headaches to come! ) ... or the M2Tech, that's an option too.
 
Thanks again for everything!

 
Yes, I know. I live in Denmark. You should consider something with XMOS as this is maybe the best USB chip at the moment. I have a DAC with the TE8802 which should also be USB audio class 2, but it has annoying glitches so I use the Hiface2 instead (I use redbook material and sometimes up to max. 96KHz). Here's a comprehensive list of USB audio class 2 devices:
http://www.forum-audiophile.fr/musique-demat/les-dac-transport-usb-audio-class-2-t16248.html
 
Have fun!

 

[analogue 3]

Can anyone recommend a linux music player with bit perfect audio AND folder browsing?
 
I've been using mediamonkey with asio/wasapi for sometime and really enjoy the option of browsing by folder when making custom playlists etc. Something with a similar GUI would be nice too.
 
I would try mediamonkey over wine however I guess I'm more likely to get bit perfect with something native to linux...

 

[Anda]

Music Player Daemon (backend). As frontend I am using ncmpcpp and Sonata (see pics below). Takes a bit of work to set up, but works beautifully.
http://mpd.wikia.com
https://wiki.archlinux.org/index.php/MPD
 

 

 

[Puranti]

After having tested many players (deadbeef, quodlibet, sonata, gmusicbrowser, rythmbox, amarok, etc).
I've discovered recently QMMP, and it's the closest player to winamp you can find.
It has bit perfect playback, great plugins (bs2b, ladspa, etc).
And its UI is very tweakable. I'll settle with this one.
My audinirvana is near

 

[Migou67]

QMMP look very nice, I will give it a try this evening, using Clementine for the moment, Thanks.

 

[Zodac]

Hi there,
 
I'm new into Unix bit perfect sound. I'm using ubuntu server (no GUI) on my HP N40L and my Asus Xonar Essence One via USB with mpd and alsa.
 
I have a strange problem with this setup. Playing files works, but if I change from a "normal" mp3 file with 44.1 khz to a file with 48 khz and vice versa the sound plays too fast / too slow. I have to stop and play the file again and then it's fixed.
 
The first mp3 file after a reboot sounds very corrupted and the Essence One shows 192 khz playback. Stopping and starting fixes this problem too. Any one with an idea to fix this?
 
My Config:
 
ALSA asound.conf
 
pcm.!default {
        type hw
        card One
#       device 0
}

ctl.!default {
        type hw
        card One
}
 
MPD mpd.conf
 
audio_output {
        type            "alsa"
        name            "One"
        device          "hw:0,0" 
        auto_resample   "no"
}
 
 
aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: One [ASUS Xonar Essence One], device 0: USB Audio [USB Audio]
  Subdevices: 0/1
  Subdevice #0: subdevice #0
 
 
 aplay -L
null
    Discard all samples (playback) or generate zero samples (capture)
sysdefault:CARD=One
    ASUS Xonar Essence One, USB Audio
    Default Audio Device
front:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    Front speakers
surround40:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    4.0 Surround output to Front and Rear speakers
surround41:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    IEC958 (S/PDIF) Digital Audio Output
dmix:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    Direct sample mixing device
dsnoop:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    Direct sample snooping device
hw:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    Direct hardware device without any conversions
plughw:CARD=One,DEV=0
    ASUS Xonar Essence One, USB Audio
    Hardware device with all software conversions
 
on a correct playing file:
 
 cat /proc/asound/card0/stream0
ASUS ASUS Xonar Essence One at usb-0000:00:13.2-1, high speed : USB Audio

Playback:
  Status: Running
    Interface = 1
    Altset = 1
    URBs = 8 [ 8 8 8 8 8 8 8 8 ]
    Packet Size = 104
    Momentary freq = 44109 Hz (0x5.8380)
    Feedback Format = 16.16
  Interface 1
    Altset 1
    Format: S16_LE
    Channels: 2
    Endpoint: 6 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000
    Data packet interval: 125 us
  Interface 1
    Altset 2
    Format: S32_LE
    Channels: 2
    Endpoint: 6 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000
    Data packet interval: 125 us
 
 
on a corrupt sounding flac 192 khz file:
 
 cat /proc/asound/card0/stream0
ASUS ASUS Xonar Essence One at usb-0000:00:13.2-1, high speed : USB Audio

Playback:
  Status: Running
    Interface = 1
    Altset = 2
    URBs = 8 [ 8 8 8 8 8 8 8 8 ]
    Packet Size = 208
    Momentary freq = 176438 Hz (0x16.0e00)
    Feedback Format = 18.14
  Interface 1
    Altset 1
    Format: S16_LE
    Channels: 2
    Endpoint: 6 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000
    Data packet interval: 125 us
  Interface 1
    Altset 2
    Format: S32_LE
    Channels: 2
    Endpoint: 6 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000
    Data packet interval: 125 us
 

 

[KimLaroux]

Weird, you're not the first in this thread reporting this issue. Have you looked for bug reports?
 
Others had the same problems using different DACs and different music players. I'm guessing it's a problem with ALSA. It's as if ALSA forgets to change the output sample rate to match the input sample rate.
 
Does your server has a sound card? What happens if you unplug the One and replace it with the on-board sound card in all the configuration files?

 

[Zodac]

I haven't checked bug reports, but I did a google search and didn't came up with anything regarding this issue. I guess I'll try out the One on my windows system via USB (it's connected through coax actually) and check if it's an issue with the One (updated the firmware last week).
 
My HP N40L doesn't have onboard sound so I can't try that.

 

[TommyNavara]

HI guys!!!!!
I'm a ubuntu user.
I use DeadBeef with a FiiO e07k usb dac and it works selecting both IEC...spdif and Direct to hardware without modification.
The sound is very clear and detailed.
I have two question:
is there a way to verify that the audio is really bit-perfect?
Ia possible to use the bit perfect stream playing for example spotify or mog through the flash player (I've tried Jack but selecting the usb dac is not able to start)

 

[KimLaroux]

Quote:

I haven't checked bug reports, but I did a google search and didn't came up with anything regarding this issue. I guess I'll try out the One on my windows system via USB (it's connected through coax actually) and check if it's an issue with the One (updated the firmware last week).
 
My HP N40L doesn't have onboard sound so I can't try that.

 
First, the problem is clearly in Linux, so testing in in Windows isn't gonna tell you much. It's gonna work.
 
Second, testing it trough COAX would not be a representative test. When you connect a DAC trough SPDIF, be it coaxial or optical, to a computer, you're not using the DAC as an audio device. You're using the computer's sound card as an audio device. The whole signal path is so different, it would be almost impossible that you have the same problem with both. It would have to be a problem with the DAC's DAC chip itself, or in userland apps.
 
Have you tried using a different app? A simple test would be to create a folder containing tracks you know when played one after the other will replicate the bug. Then play the folder using aplay or something similar.

 

[bcschmerker4]

Thanks for a Thread for tracking issues with ALSA™ 1.0.2x w/r/t bit-perfect transmission and reception.  I am currently considering several PCI 2.x audio cards as part of planning for a LinUX-box rebuild, including, among others, the Asus® XONAR® Essence™ ST (C-Media® CMI-8788; snd-virtuoso) and E-MU®/Creative® EM3961 1010m (Creative® CA0102; snd-emu10k1-fpga).  (I found my current Creative Laboratories® SB0350 Sound Blaster® Audigy2™ ZS™ to pick up excessive noise from the balance of the 'puter.)  So far, which card has had the least latency from the ALSA™ drivers? the best sound on stock op amps (through beyerdynamic® DT880 600Ω Pro's)?  These will be factors in deciding on an SB0350 replacement.
 
(The Advanced LinUX Sound Architecture Project™ has been unable to develop a fully-functional snd-ctxfi for the Creative Laboratories® CA20K2 DSP, required for the SB0880 Sound Blaster® X-Fi® Titanium™, SB0885 Sound Blaster® X-Fi® Titanium™ FATAL1TY® Professional™, SB0886 Sound Blaster® X-Fi® Titanium™ FATAL1TY® Championship™, and SB1270 Sound Blaster® X-Fi® Titanium™ HD™.  Furthermore, the ALSA™ Project has yet to receive one of the brand-new Sound Blaster® RECON3D® PCI-Express audio cards for checking development work.)

 

[Zodac]

Hi,
 
I found this http://www.spinics.net/linux/fedora/alsa-user/msg10475.html. This guy has exactly the same problem.
 
I tried:
 
cat /dev/urandom | aplay -D hw:2,0 -f S16_LE -c2 -r192000
cat /dev/urandom | aplay -D hw:2,0 -f S16_LE -c2 -r44100
cat /dev/urandom | aplay -D hw:2,0 -f S16_LE -c2 -r48000
 
and each time it changed from (a multiple of) 44100 or 48000 the sound was played wrong so its clearly a problem with alsa. I'm running on Version 1.0.24.

 

[Anda]

Does anyone have experience with using SoX resampler in MusicPlayerDaemon? Can you give an example configuration, e.g. if I wanted to upsample everything to 24/96?