[size=xx-small]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.[/size]
[size=xx-small]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. [/size]
[size=xx-small]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? [/size]
Dear ssedlmayr,
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.
Regards,
Ronald