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Bit Perfect Audio from Linux - Page 11

post #151 of 258

What are the favoite mpd front ends used around here?

 

http://mpd.wikia.com/wiki/Clients#GTK.2B

 

If one was to use mplayer2 for bitperfect gapless playback, what are the recommended GUI's/frontends for mplayer2?

post #152 of 258
Quote:
Originally Posted by wolfson View Post

 

I was just trying (unsuccessfully) to be facetious ;)

 

My understanding is that "bit perfect" audio output means that there ought to be no difference in the data/bits being presented to a DAC, regardless of OS/app (other than jitter magnitude). Once the claim is made that "bit perfect" is achieved at the output of a device (CD player, computer, etc), the output should be identical between the devices.

 

It appeared that the maker of the "Bit Perfect" Mac app was trying to claim otherwise.  I may have been mistaken.

 

I was mocking what I believed to be an example of audiophile-market-snake-oilsmanship

Ah right, it took me a while to get my head around it! Unfortunately there's no paucity of snake-oil in this hobby!

 

 

 

Quote:
Originally Posted by wolfson View Post

What are the favoite mpd front ends used around here?

 

http://mpd.wikia.com/wiki/Clients#GTK.2B

 

If one was to use mplayer2 for bitperfect gapless playback, what are the recommended GUI's/frontends for mplayer2?

 

If I understood the question then I use Sonata from my laptop or MPDroid from my phone to control the daemon which is on my RPi. One of the downfalls of MPDroid for me is that there's no option to view all tracks but Sonata is good for making playlists so I just whack it all in one 'All Tracks' playlist.

 

I think options for gapless/bit depth/sample rate are in the MPDaemon/Alsa config files.


Edited by SamHedges - 6/10/13 at 10:05am
post #153 of 258
Quote:
Originally Posted by wolfson View Post

What are the favoite mpd front ends used around here?

http://mpd.wikia.com/wiki/Clients#GTK.2B

If one was to use mplayer2 for bitperfect gapless playback, what are the recommended GUI's/frontends for mplayer2?

For a graphical front end I like gmpc a lot because it can display all the metadata that mpd is capable of using. This means I can use comment and composer tags as well as the more usual artist, album etc. There are very few graphical mpd clients which take full advantage of mpd's facilities.

For a console front end I love ncmpcpp. It allows full use of metadata, appearance is easily customised and it's easy to use.

gmpc

ncmpcpp


Front end for mplayer2? You could use smplayer2 if you want a GUI. I usually use mplayer2 via scripts, aliases and file manager context menus. The keyboard and mouse controls are good (and can be customised) and I don't find a GUI necessary. I can't claim it's much fun reading the manual and getting it exactly as wanted but once that's done it's done.
post #154 of 258
Quote:
Originally Posted by HPiper View Post

Thanks a ton for this very informative thread. I installed Gmusicbrowser and now FINALLY have bit-perfect playback of my music. Only problem I have is I can't get sound from my speakers, but it works with my dac and headphones. I'll play with it a bit more later, but thanks again.

 

If your speakers are internal or connected to a sound card, then it's expected behavior. In order to set your music player as bit-perfect, you have to set it to output directly to a specific device. This means the audio stream will only go to this device, and not the others.

 

The idea of a software mixer is to fix this problem. A software mixer mixes all incoming streams and sends a copy to every connected device. To get bit perfect, we bypass this mixer, losing this capability. Funny thing is you can still send a different stream to each device. So you can have music out of your DAC while watching a youtube video trough your speakers. I do this all the time.

post #155 of 258
Quote:
Originally Posted by KimLaroux View Post

 

If your speakers are internal or connected to a sound card, then it's expected behavior. In order to set your music player as bit-perfect, you have to set it to output directly to a specific device. This means the audio stream will only go to this device, and not the others.

 

The idea of a software mixer is to fix this problem. A software mixer mixes all incoming streams and sends a copy to every connected device. To get bit perfect, we bypass this mixer, losing this capability. Funny thing is you can still send a different stream to each device. So you can have music out of your DAC while watching a youtube video trough your speakers. I do this all the time.


 It is not big problem, I got it working. I had the wrong numbers in the advanced options I had 2 choices after running the aplay -l command and I just picked the wrong one. It was very clear, after running that command, which my dac was but I wasn't sure about my speakers. Still wondering what that 3rd option is for, I am thinking it is the coax or toslink output from my soundcard. Anyway everything is working perfectly now.

 If I can ever get Linux to do all the stuff I need it to do I fully intend to ditch Windows. This is one large step in that direction.


Edited by HPiper - 6/10/13 at 11:23am
post #156 of 258

In theory, it should be possible to hear the same bitstream on two devices simultaneously, one bit perfect via a USB DAC and duplicated on your PC internal soundcard to its analog output.

 

This would be functionally similar to how any low cost set top CD/DVD player sends bit perfect PCM from a CD out its coax/optical SPDIF jack, plus the same audio out its analog RCA outputs.

 

Using Jack would probably be required to achieve this, perhaps pulseaudio.

 

http://jackaudio.org/

 

http://en.wikipedia.org/wiki/PulseAudio

 

From an abstract point of view, all that is needed is for the bitstream to be duplicated in memory after read/decoded from the source file, then routed to each sound device, a trivial task for a CPU/memory/software, but may be tricky to set up from a practical point of view.

 

You would set multiple sinks (outputs) from the same source (application/audio daemon/bitstream) without mixing/resampling.

 

I have never tried this- just thinking out loud.


Edited by wolfson - 6/10/13 at 12:19pm
post #157 of 258
post #158 of 258

I just removed the resampling line in the output part of mpd.conf and for some reason there's loads of clicking when I send music to my DAC from the Pi. Can anyone tell me why this is and how to solve it?

post #159 of 258
Quote:
Originally Posted by wolfson View Post

In theory, it should be possible to hear the same bitstream on two devices simultaneously, one bit perfect via a USB DAC and duplicated on your PC internal soundcard to its analog output.

 

This would be functionally similar to how any low cost set top CD/DVD player sends bit perfect PCM from a CD out its coax/optical SPDIF jack, plus the same audio out its analog RCA outputs.

 

Using Jack would probably be required to achieve this, perhaps pulseaudio.

 

http://jackaudio.org/

 

http://en.wikipedia.org/wiki/PulseAudio

 

From an abstract point of view, all that is needed is for the bitstream to be duplicated in memory after read/decoded from the source file, then routed to each sound device, a trivial task for a CPU/memory/software, but may be tricky to set up from a practical point of view.

 

You would set multiple sinks (outputs) from the same source (application/audio daemon/bitstream) without mixing/resampling.

 

I have never tried this- just thinking out loud.

 

I believe that the moment you use Jack or Pulse, you lose all bit-perfect output. These things are nothing more than software mixer that plays with the audio stream

 

Quote:
Originally Posted by SamHedges View Post

I just removed the resampling line in the output part of mpd.conf and for some reason there's loads of clicking when I send music to my DAC from the Pi. Can anyone tell me why this is and how to solve it?

 

It's a known bug on the Pi. I'm blaming a flaky USB implementation and the lack of processing power. There is no fix yet as far as I know.

post #160 of 258

Actually not to worry I've just found a fix, it's coming in the next Raspian update. Have a look here.

post #161 of 258

Linux is lacking a bit on the software side of the audiophile world, but makes up for it when you do eventually get a good setup the audio drivers are really awesome. Two of the best words in Linux Audio world is ALSA and Jack, very configurable stuff.

 

I was a linux user for a while, especially when I was stuck with lower end hardware, it's good to play with, still waiting for the day I can truly start using it with all of the games I play.

 

Out of the software itself I'd recommend Amarok, MPD, heck even Rhythmbox is pretty cool and then there's some of the Amarok wannabe's too like Clementine.

 

But I don't understand the bit perfect stuff yet

 

P.S: Linux doesn't seem to be able to do even dsd2pcm which is a shame :( I call DSD superior, in any form, be it native or converted to PCM. To compare with even 24/192 well.. vinyl rips at least, I can clearly hear the instrument separation and at least to me it seems like there's plenty more clarity, and maybe even somewhat of a warmth effect.


Edited by XVampireX - 6/13/13 at 3:50pm
post #162 of 258
Quote:
Originally Posted by XVampireX View Post

I call DSD superior, in any form, be it native or converted to PCM. To compare with even 24/192 well.. vinyl rips at least, I can clearly hear the instrument separation and at least to me it seems like there's plenty more clarity, and maybe even somewhat of a warmth effect.

Well, but DSD could be considered inferior to even 24/96. The aggressive noise shaping adds a lot of HF noise and there's the problem with editing..

There's a thread in sound science about this, about 2 weeks old or so.

post #163 of 258

What's the best way to get USB sound to work with Spotify?

post #164 of 258
Quote:
Originally Posted by XVampireX View Post

 

 

P.S: Linux doesn't seem to be able to do even dsd2pcm which is a shame :( I call DSD superior, in any form, be it native or converted to PCM. To compare with even 24/192 well.. vinyl rips at least, I can clearly hear the instrument separation and at least to me it seems like there's plenty more clarity, and maybe even somewhat of a warmth effect.

 

I think the guys at http://www.hydrogenaudio.org/forums/ might take issue with such "precise" technical jargon :)

 

Given known electrical engineering, acoustics and digital audio science, I'm personally convinced that anything more than 20bit/48Khz PCM is unneeded for playback and cannot be distinguished by most people on most audio equipment, maybe no one in a controlled ABX test.

 

AFAIK, no one has successfully picked out a difference between 16/44.1 and 24/96 in a proper* ABX test.

 

Similarly, I am not aware of any proper ABX test results showing someone can pick out DSD vs PCM

 

Feel free to provide links/data to back up a claim to prove otherwise.

 

*start with a 24 bit 96Khz audio file (FILE1) of your choosing, downsample to 16bit 44.1Khz (file2) then upsample file2 to 24bit 96khz (file3) with accepted best practice resampling software methods (see hydrogenaudio forums), then ABX file1 and file3 on a 24/96 DAC of your choosing.  Foobar with ABX plugin is recommended.


Edited by wolfson - 7/23/13 at 12:46pm
post #165 of 258
Quote:
Originally Posted by wolfson View Post


AFAIK, no one has successfully picked out a difference between 16/44.1 and 24/96 in a proper* ABX test.

http://www.hydrogenaudio.org/forums/index.php?showtopic=81467&st=0

seems to show more than one successful abx of 16/44.1 vs 24/96.

I find statements along the lines of "nobody has ever abx'd x y or z" can be as unreliable as claims of "I can hear the difference but it's so obvious I didn't measure it and I'm too busy to demonstrate it" because I've seen enough claims here and elsewhere that it's supposedly impossible to abx codec Z vs lossless, player A vs player B and so on, and those claims can persist even after being debunked with abx logs and similar. I also saw that one of the most famous "no result" conclusions regarding hi res vs red book derives from a study where some of the "hi res" files were in fact upsampled red book, see item 13 at http://www.head-fi.org/t/486598/testing-audiophile-claims-and-myths so I think it pays to be skeptical of claims of scientifically proven non-difference as well as of the touchy feely impressions. Where people have a stake they also have a motive or, more kindly, anyone can unknowingly make a series of fundamental mistakes that purely coincidentally lead people to where they hoped they would go in the first place.

Can I abx 16/44.1 vs 24/96? I don't have the kind of quiet environment or speakers that would make it worthwhile for me to perform abx of 16/44.1 vs 24/96 where the red book file is a very high quality derivative so I don't buy hi res or worry about it but I can't rule out other people being able to perform an abx that shows a difference.
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