[ronalde]

Hi bcschmerker4®,
 
Whether the interface is balanced (IEC 60958 Type I using XLR) or not (IEC 60958 Type II using RCA/coax) is of no importance to the symptoms. Both interfaces carry computer generated SPDIF-signals with all problems related to it. Those subtle problems in sound quality are inherent to the SPDIF-signal, and exist even when there is proper signal transfer without coarse issues like grounding loops, signal decay and/or dropped packets.
 
The latter problems could benefit from dedicated sound hardware inside the PC coupled to balanced connections. On the other hand, if you experience those issues at your home setup, something else is wrong. They are intended for professional use where long distances between devices are required and the prevention of electrical problems (in the rest of the delivery chain) is more important than sound quality. That's also the reason that currently not a single professional audio device exists which aims at bit-perfectness in the whole chain.
 
Regards,
Ronald

 

[bcschmerker4]

Thanks, @ronalde, for the additional details on the problem.  As of August 2014, the Creative Laboratories® SB0350 PCI 2.2 card and SB0250 I/O Drive are still functional in my custom LinUX box; I'm still debating a potential re-build around an Intel® Core Processor™ i5-3500 Series APU (FCLGA 1155), ASRock® Z68 Pro-3M, and up to four new Western Digital® Caviar Red® hard drives, as JACK™ doesn't want to fly at all speeds on the current Advanced Micro Devices® Athlon 64® X2 5600+ (running at 2.9 GHz) and 780G/SB700 chipset.

 

[ronalde]

  Thanks, @ronalde, for the additional details on the problem.  As of August 2014, the Creative Laboratories® SB0350 PCI 2.2 card and SB0250 I/O Drive are still functional in my custom LinUX box; I'm still debating a potential re-build around an Intel® Core Processor™ i5-3500 Series APU (FCLGA 1155), ASRock® Z68 Pro-3M, and up to four new Western Digital® Caviar Red® hard drives, as JACK™ doesn't want to fly at all speeds on the current Advanced Micro Devices® Athlon 64® X2 5600+ (running at 2.9 GHz) and 780G/SB700 chipset.

Jack is middleware aimed at (semi-)professional usage and aims for troublefree and "realtime" routing of multiple audio streams, like pulseaudio is middleware targeted towards desktop users providing them a seamless plug-n-play experience.
 
Those layers settle for one single fixed sample rate and bit depth for all streams by providing automagic resampling, bit depth conversion and/or leveling. In other words bit-imperfectness is their raison-d'être.
 
Regards,
Ronald 

 

[watchpocket]

As one new to the attempt to achieve bit-perfect, I found Rizlaw's post #3 and yay101's post #265 the two single most informative items in this entire thread.  Thank you to both of you! 
 
My goal right now is simply to be able to play DSD files -- .dsf and .dff files -- preferably in DeaDBeef and/or Clementine. 
 
But so far no matter what player I'm in, when I open a .dsf file, nothing happens.  I'm on a 64-bit Ubuntu 12.04 desktop workstation using Gnome Basic (I don't want to know about Unity, thank you).
 
My USB DAC is the Light Harmonic / LH Labs Geek Out 1000 (the "1000" refers to the DAC's power in milliwatts), which definitely handles DSD (via DoP) and all sampling rates.

Should players like DeaDBeef and Clementine be able to play .dsf and .dff files out-of-the-box?  I suspect not.  But what combination of players, plug-ins, codecs and voodoo might be needed?  I've installed every Gstreamer plugin I could find for Rythmbox, I've tried to use the Signalyst hqplayer trial (way complicated), as well as Amorak, Audacious, VLC, SMplayer/mplayer and a few others, to play a .dsf file. 
 
I see that JRiver Media Center, MPD, and Kodi (formerly XBMC) all advertise that they play DSD files, but I'd like (for now at least) to avoid large client-server-type programs such as these, particularly JRiver which is not free and open-source. I may use one of them if necessary, but first I'd like to see if I can get a more lightweight, audio-only player to play DSD files. 
 
I can get the bit-perfect -- there's plenty of explanation here to do that.  But I haven't yet been able to play DSD files in Ubuntu, as I have with foobar2000 on windows (at my work location) with the same DAC.   I can provide output to the standard commands if needed, but wondered, before I do that, if there might be a more fundamental answer. 
 
Thanks for any guidance or pointers.

 

[bcschmerker4]

  As one new to the attempt to achieve bit-perfect, I found Rizlaw's post #3 and yay101's post #265 the two single most informative items in this entire thread.  Thank you to both of you! 
 
My goal right now is simply to be able to play DSD files -- .dsf and .dff files -- preferably in DeaDBeef and/or Clementine. 
 
But so far no matter what player I'm in, when I open a .dsf file, nothing happens.  I'm on a 64-bit Ubuntu 12.04 desktop workstation using Gnome Basic (I don't want to know about Unity, thank you).
 
My USB DAC is the Light Harmonic / LH Labs Geek Out 1000 (the "1000" refers to the DAC's power in milliwatts), which definitely handles DSD (via DoP) and all sampling rates.

Should players like DeaDBeef and Clementine be able to play .dsf and .dff files out-of-the-box?  I suspect not.  But what combination of players, plug-ins, codecs and voodoo might be needed?  I've installed every Gstreamer plugin I could find for Rythmbox, I've tried to use the Signalyst hqplayer trial (way complicated), as well as Amorak, Audacious, VLC, SMplayer/mplayer and a few others, to play a .dsf file. 
 
I see that JRiver Media Center, MPD, and Kodi (formerly XBMC) all advertise that they play DSD files, but I'd like (for now at least) to avoid large client-server-type programs such as these, particularly JRiver which is not free and open-source. I may use one of them if necessary, but first I'd like to see if I can get a more lightweight, audio-only player to play DSD files. 
 
I can get the bit-perfect -- there's plenty of explanation here to do that.  But I haven't yet been able to play DSD files in Ubuntu, as I have with Foobar2000 on windows (at my work location) with the same DAC.   I can provide output to the standard commands if needed, but wondered, before I do that, if there might be a more fundamental answer. 
 
Thanks for any guidance or pointers.

After looking up some possible programs at FileInfo.com, I found that Direct Stream Digital DFFs are handled by foobar2000, which is ported across all major OS's.  OtOH, Delusion Music Files (DMFs) and Delusion Sample Files (DSFs) were developed by Olivier Lapicque for applications originally developed for Microsoft® Windows® 6-up; don't know whether Lapicque has the source code available for download.

 

[watchpocket]

Thanks for your response.  foobar2000, however, is not available on the Ubuntu Linux platform.
 
The .dsf file extension I'm referring to is this: (DSD) Storage Facility files (DSF), a digital audio file format and file extension (.dsf).
 
The .dff extension is the DSD Interchange File Format (DSDIFF) (.dff).
 
They're the only two DSD audio-file extensions that I'm aware of.
 
From DSD-Guide.com:
 
DSD or Direct Stream Digital is a one bit recording format.  DSF and DFF are 'kinds' of audio files that are delivered as downloads or read by software for playback. 
DSF has the ability to hold metadata and DFF doesn't, however, some players only read DFF files.

In PCM recording, DSF and DFF would be similar to having .wav and .aiff files which are both PCM
More complex answers to the above question exists, but for those new to DSD, this is a simple explanation.
 
I don't know from "Delusion Music Files" but I'm pretty sure they don't have anything to do with DSD audio, I could be wrong.
 

 

[stv014]

  Thanks for your response.  foobar2000, however, is not available on the Ubuntu Linux platform.

 
It can be run without any major problems using Wine. If you use the WASAPI plugin in "push" mode, and select the correct audio output device for Wine (that is, one that uses ALSA directly), then software processing issues should be avoided. On the other hand, the last time I checked, the default DirectSound output was not bit perfect.

 

[watchpocket]

Thank you for the tip.  I'm going to hold off on Wine/foobar though until I'm sure no native Ubuntu app does what I'm looking for.

 

[yay101]

At work, can't really reply. But just a heads up at this stage look at mpd for DSD/ DSF.

 

[watchpocket]

Thanks, yay101.  My thinking has already been moving in the direction that mpd might be the most useful item for me right now.

 

[yay101]

MPD is excellent as a daily player, the default limits on playlists can be annoying though. If I had a DSD capable DAC I would write a guide for setting it up, but sadly I do not.

 

[ronalde]

Originally Posted by watchpocket /img/forum/go_quote.gif
 
I've installed every Gstreamer plugin I could find ...
 
I can get the bit-perfect -- there's plenty of explanation here to do that.

Hi watchpocket,
 
gstreamer is never bit-perfect. Stick with alsa with mpd in bit-perfect mode, eg, using hw:x,y alsa output device names in mpd.conf.
 
No voodoo required .. a little prayer around christmas never hurts though.
 
Regards,
Ronald

 

[userser]

   
    DeaDBeef:  (a) Click on the Edit menu, then Preferences.
                       (b) In the Preferences window under "Sound" tab > "Output plugin" = "ALSA"
                                                                                                          "Output device" = "HDA Intel ALC889,
                                                                                                                                      IEC958, S/PDIF" 
 
                            NOTE: the "Output device" choice will depend on the type of sound card in your computer. It may be labeled differently from the one I have.                           
 
                       (c) In the same Preferences window select > Plugins
                            In the left column select "Alsa Output Plugin" and then click the "Configure" button
                            In the configuration window make sure that ALSA resampling is unchecked and that you place a check in box to "Release device when stopped".
 
You are now done setting up DeaDBeeF for bit perfect playback.

 
 
 
I just signed up to tell you guys that this setup still rocks on Ubuntu 14.04.1 LTS; DeadBeef 0.6.2. I found nothing as simple as this.
 
 
 

GYROCOM C&C Co., LTD Audiotrak ProDigy CUBE at usb-0000:00:1d.0-1, full speed : USB Audio
Playback:
  Status: Running
    Interface = 3
    Altset = 2
    Packet Size = 582
    Momentary freq = 96000 Hz (0x60.0000)
  Interface 3
    Altset 1
    Format: S16_LE
    Channels: 2
    Endpoint: 3 OUT (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000
  Interface 3
    Altset 2
    Format: S24_3LE
    Channels: 2
    Endpoint: 3 OUT (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000
Capture:
  Status: Stop
  Interface 2
    Altset 1
    Format: S16_LE
    Channels: 2
    Endpoint: 2 IN (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000
  Interface 2
    Altset 2
    Format: S24_3LE
    Channels: 2
    Endpoint: 2 IN (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000

 
 
 
Peace&Love

 

[bixby]

I hope someone can help with a dropout problem I am having with Deadbeef.  
 
Running HP Laptop with Lubuntu installed and Deadbeef.  Machine has core 2 Duo and 2GB of ram.  Files are located on NAS and 16/44khz Aiff files streamed via wireless to the Linux machine.  Throughput is 100 Mbps to 350 Mbps on wireless.  Deadbeef is feeding USB dac via Alsa plugin with no resampling and buffers currently set at 8290 adn 1024 for the period size.
 
I get very short dropouts about every 5-10 minutes of playback.  Any idea what can be adjusted to remedy?
 
thanks for your ideas

 

[yay101]

5-10 minutes sounds network related to me, I'd set deadbeef to buffer the full file then see if it still happens.