It depends on USB chip in the DAC: some DACs recognized on fly, some with problems, some wth workarounds.
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Gmusicbrowser also has verbose and debug flags, if you just force it to crash the regular output is less than useful.
I installed it and tried to replicate the bug but to no avail.
There are no commonly reported bugs or tickets on anything like this issue so strace would be useful as well in case it's something wrong with your system.
Distro? Version? Gstreamer version? If you replicate the activity on a different player will a similar behaviour occur?
It's probably facing some issues being on a Distro that doesn't fully support it, or perhaps an older version of Gstreamer. I'm running the latest version of Bodhi Linux, and my most common problems are the aforementioned crash issue and a glitched display of time passed and time remaining in a song when I pause it. If I pause halfway through a 4-minute song, it might display 122 hours of play in the song, and two minutes left, with the time slider all the way to the right.
You're right, jack isn't bit perfect and isn't supposed to be. Jack its "middle ware" connecting the input and output of "user applications" to those of "backend" libraries and drivers, like alsa, oss and portaudio.
Jacks reason for being is (low-latency) recording and playback and (sample rate and format) synchronization between the in and output streams applications and the backend software (like hardware drivers) produce or consume.
jackd (the master daemon process) glues all those different streams together to a fixed sample rate and bit depth. The combined output channels are just one of those streams, which jacks synchronizes and reformats.
Since then I have acquired a good Linux computer and I am running Ubuntu 12.10 with a Sound Blaster SB400 and Grado SR325 headphones. Using your info I think I have achieved 24/192 output. I am using DeadBeef for the program. I chose the output to be set at SB Audigy 2 Value (400), ADC Capture/Standard PCM device Playback Direct Hardware without any conversions. Also set the secret Rabbit Code to automatic samplerate and a target sample rate of 192,200, and quality algorithm to since_best_quality. For the ADplug I set it to prefer Ken emu over Satoh, and then set the ALSA output plugin to no ALSA resampling and release device while stopped. Preferred buffer size is 20000 and preferred period size is 1024.
I do not know what some of these do, but the 24/192 recordings I do have on my hard disk do sound pretty good.
And the last question, if I do the above, would it benefit me to use the output of the DAC to go into a tube headphone amplifier? Would I gain anything?
It's sad to read about your unfortunate crash. Nice though that you're endeavors led to the queen having a proper sound system, I guess it's rather hard to fill up her not too modest surroundings with majestic sound.
Maybe this sheds some light on your questions.
Your software configuration is not geared towards "bit perfect" audio playback. Bit perfect is about getting rid of any influence of the computers operating system and user applications on (the playback of) digital source files. The source files are supposed to have arrived at your music playback computer in a (bit) perfect manner. This could easily be assessed by comparing the bits that make up the files, both at the sender and your computer, if the sender is willing to provide such information. In the computer world this normally is done using checksums, for example using the popular md5sum program. No confusion on the "perfectness" is possible.
The part of getting the (pristine) source music files from your computer (or network device) to a playback system like your SB Audagy in a (bit) perfect manner is a little more complex. That is because all consumer operating systems (like Linux, Mac and Windows) are default configured towards usability and convenience, and not audio playback. That explains the existence of threads like this one. According to the description of your configuration, in which deadbeef sends it's output to pulseaudio, which re-samples and re-formats the incoming audio, before handing it over to alsa's libraries and hardware driver for the SB Audagy, it is not bit perfect. It alters the digital audio before handing it over the DA converter. Many audio enthusiasts therefore prefer "bit perfect" output of their music playback software.
BTW, a DA converter can't be bit perfect, as such a device has both a digital and an analog end (no "bits" there) and the D-to-A conversion involves filters that influence the analog audio in the audible domain.
With your current hardware and Ubuntu, it's perfectly possible to get this kind of transparent transfer of digital audio. On Ubuntu, it's all about bypassing pulseaudio or other audio altering software and using alsa's hardware interfaces instead (using "hw:x,y"). Just read about it in this thread, yay101's Newbie Guide on bit perfect playback or the articles on my website (which are geared towards using Music Player Daemon / mpd).
You could then really concentrate on getting good sounding source files and comparing different audio components, like a separate DA converter and/or a high quality headphone amplifier, without having to worry about the applications or the operating system influencing the sound quality.
Bifrosf have USB C-Media CM6631A chip. It also in Modi.
I found out this link:
May be it will help.
The BiFrost SHOULD be plug and play, the chip it uses for usb is supported on linux in usb1.0 mode. Usb 2.0 mode will be supported as soon as CMedia get around to it.
Always be on the watch for devices that have supported hardware but then use a joint usb mode for connections such as using other chips on input to lower or remove jitter, these often have adverse affects without the specific driver from the manufacturer.
Managed to get CMUS bitperfect as far as I understand via ALSA (no dmix, no software volume control unlike deadbeef, no DSP, no funny business), as CMUS is ncurses based you can also run outside of X, which is nice.
Open CMUS, then press 7, scroll down to "output_plugin" make sure it's "alsa".
Scroll up to "dsp.alsa.device" make sure it's "plughw:0", 0 worked for me, could be 0,0 or 1,0 etc. If plughw doesn't work try "hw:0" or 0,0 etc.
Scroll down to change "mixer.alsa.channel" delete "Master" so the setting is empty, also empty "mixer.alsa.device".
Lastly make sure "softvol" is "false".
Changing volume via "[ & ]" in CMUS produces: "Error: can't change volume: mixer is not open".
"cat /proc/asound/card0/pcm0p/sub0/hw_params" output from a 44.1Hz FLAC file is:
rate: 44100 (44100/1)