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To those of you who are listening to hi-rez audio on Ubuntu and derivatives

121.209.219.128
Posted on November 29, 2010 at 15:17:00
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 Make sure you change your PulseAudio settings.

By default, everything is converted to 44.1kHz 16-bit before sending to output device.

Extracted from my vanilla Ubuntu system:

$ pacmd
Welcome to PulseAudio! Use "help" for usage information.
>>> stat
Memory blocks currently allocated: 1, size: 63.9 KiB.
Memory blocks allocated during the whole lifetime: 28, size: 1.6 MiB.
Memory blocks imported from other processes: 0, size: 0 B.
Memory blocks exported to other processes: 0, size: 0 B.
Total sample cache size: 0 B.
Default sample spec: s16le 2ch 44100Hz
Default channel map: front-left,front-right
Default sink name: alsa_output.pci-0000_00_1b.0.analog-stereo
Default source name: alsa_input.pci-0000_00_1b.0.analog-stereo
Memory blocks of type POOL: 1 allocated/7 accumulated.
Memory blocks of type POOL_EXTERNAL: 0 allocated/0 accumulated.
Memory blocks of type APPENDED: 0 allocated/0 accumulated.
Memory blocks of type USER: 0 allocated/0 accumulated.
Memory blocks of type FIXED: 0 allocated/21 accumulated.
Memory blocks of type IMPORTED: 0 allocated/0 accumulated.

 

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RE: To those of you who are listening to hi-rez audio on Ubuntu and derivatives, posted on November 29, 2010 at 23:15:33
giulio


 
And if you use alsa instead, everything gets resampled to 48k.

In the latter case a line like

pcm!default plughw:X

in your .asoundrc file (where X is the number of your card from the output
of the command "aplay -L") should solve the problem

 

It's even worse than that, posted on November 30, 2010 at 01:20:23
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 ALSA and PulseAudio have a symbiotic relationship in Ubuntu.

ALSA is configured to send audio to PulseAudio as a virtual device, PulseAudio then resamples and converts everything to (by default) 16/44.1 and sends it out to an ALSA device.

By default, ALSA dmix is configured to operate at 48kHz.

So, in the worst possible case, if dmix is enabled (and there are quite a few tutorials on the net suggesting that this be done) ...

app -> ALSA -> dmix (resample to 16/48) -> PulseAudio (resample to 16/44.1) -> ALSA sound driver -> audio device

Or, if JACK is used and configured to 192kHz (again, some naive tutorials recommending this)

app -> JACK (resample to 24/192) -> PulseAudio (resample to 16/44.1) -> ALSA sound driver -> audio device

and in the best possible scenario

app (via gstreamer or JACK etc.) -> PulseAudio (resample to 16/44.1) -> ALSA sound driver -> audio device

Sheesh! Whoever said Linux had a great architecture for audio?

 

Update: PulseAudio currently does not support dynamically changing sample rate, posted on November 30, 2010 at 13:29:50
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 To change the sample rate that every stream is converted to, you need to kill and restart the PulseAudio daemon.

Extract from PulseAudio mailing lists:

No, this is not something we support at present due to the practical
problems that this presents.

Whenever changing the sample rate "mid stream" you'll obviously get some
kind of click/pop as the device is closed, reconfigured and reopened, so
this has to be avoided.

The alternative is to take the first sample rate and stick with it. But
then consider playing an 8kHz effect sound and then playing a nice
196kHz audio track just a moment later... it would be converted to 8kHz
which is almost certainly not what the user would want.

So whichever way you cut it, there is no nice solution to do dynamic
sample rate switching of the hardware and the only logical and practical
solution is to use a fixed sample rate on your system and covert to this.

 

RE: Update: PulseAudio currently does not support dynamically changing sample rate, posted on November 30, 2010 at 20:11:42
John Swenson
Audiophile

Posts: 2422
Location: No. California
Joined: October 13, 2002
 There is another solution, don't use pulseaudio for your high quality music. Configure your player to output directly to your high quality output device (whatever it may be) and if you have to have the clicks beeps etc coming out of the same speaker, configure them to come out of the motherboard audio jacks and mix the two analog streams. Or have good speakers for the good stuff and cheap little "computer" speakers for the beeps.

Trying to use pulseaudio in an audiophile environment is just a recipe for disaster.

I don't even have pulseaudio on any of my linux systems and I get along just fine without it.

John S.

 

RE: It's even worse than that, posted on November 30, 2010 at 20:17:41
John Swenson
Audiophile

Posts: 2422
Location: No. California
Joined: October 13, 2002
 Actually it works very well if you just turn off all the mixer stuff and have the player go directly to the ALSA device. If your players insist on just going to the default device its really simple to reconfigure ALSA so the default device is your high quality music device. Once you do that nothing gets resampled unless the actual hardware can't handle the sample rate in the file.

John S.

 

Not that easy, posted on November 30, 2010 at 21:51:46
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 Most players these days are based on gstreamer, which outputs to pulseaudio.

You can get a JACK output plugin for gstreamer, but JACK has the same problem (single master sample rate, changeable only by killing and restarting the daemon). Not practical for listening to multiple audio formats in one session.

Players not based on gstreamer are severely limited in their support of various audio formats.

And you want to stay away from ALSA. dmix is a poor quality mixing engine, considerably worse in quality compared to pulseaudio and JACK. Which is why Linux distributions are moving away from ALSA as an audio framework.

All in all, audio on Linux is a real nightmare. Gone are the good old days when Linux apps could just change the OSS sample rate on the fly. Now we have OSS emulation on top of ALSA, ALSA emulation on top of OSS, pulseaudio as a virtual audio device for ALSA, JACK on top of pulseaudio, and forget any romantic notion of bit-perfect for anything other than 16/44.1!

 

Not practical, posted on November 30, 2010 at 21:58:32
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 First of all, many players are moving away from supporting ALSA.

Secondly, hogging the ALSA device is anti-social to other apps on Linux that want to output sound. That's the reason why JACK, pulseaudio, ESD and dmix was written in the first place.

Fine if we are talking about a dedicated music only player, but not for day to day use on a general purpose machine.

 

Strange thread!, posted on November 30, 2010 at 23:31:28
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 Really?

Pulse Audio is a far-from-ideal audio vehicle for Linux. I have totally avoided Pulse and will do so until further notice.

I often edit two-channel red-book audio files in Audacity with ALSA on Pinguy (a recent Ubuntu remix). My finished results are seemingly faultless. No clicks, no pops. Not a single issue.

As to so-called "high-rez", if one considers only 24/192 or higher, or DSD, as fit for "superior" audio consumption, then Linux may appear Neanderthal. (I consider recording, mixing and mastering techniques to be far more crucial. Forget sampling rate and format.)

Meanwhile, over the last few years, by developing red book circuits and modding/improving the entire red book audio chain, I seem to have arrived at a point where I have no need for SACD. My red book playback is great. (Most re-hashed SACD reissues I already have on vinyl and CD anyway. Most recent SACD recordings are of no great interest to me.)

[PS: I've been involved with desktop Linux and digital audio production (on Mac and Windows) since the mid-90s. Music, radio broadcasting and high-end audio for 30 years.]

Over 'n out.


-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

RE: Strange thread!, posted on November 30, 2010 at 23:40:41
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 If you have truly been involved in "desktop Linux and digital audio production (on Mac and Windows) since the mid-90s" you will surely realise editing audio files on Audacity is completely different from playing back digital audio, and the necessity to share that audio device across multiple apps.

And we are talking about a vanilla install of Ubuntu here, for use as a general purpose machine, not a dedicated audio player. Sure, it's possible to remove PulseAudio from the system. That will break many applications, including all music players based on gstreamer. Including Rhythmbox, the native player for Ubuntu.

 

Another advantage of Linux....., posted on November 30, 2010 at 23:59:48
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 is customizing the base install via Synaptic or apt-get install on the CLI, CT! ;-) Vanilla and rhythm box do not fit my paradigm. Sorry. Some other vanilla apps in Ubuntu do not thrill me either. They are dispensed with as well.

As to your sneeringly dismissive comment, "If you have truly.....blah blah blah.......", may I humbly suggest to occasionally dismount your high horse for the benefit of progressive discourse?



-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

RE: Another advantage of Linux....., posted on December 1, 2010 at 03:18:11
audioAl
Audiophile

Posts: 1462
Location: So. Texas
Joined: December 16, 2007
 Here! Here!
Vista Ultimate 64 bit/e5300 Intel 45nm cpu/ASRock G41M-LE/Asus Xonar DS R 7.1/YamahaRX-V465 HT receiver/ Infinity RS1001 & Cambridge SoundWorks speakers/Yamaha YST-SW216 subwolf

 

RE: Another advantage of Linux....., posted on December 1, 2010 at 03:57:45
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 Since this thread is about listening to music in standard vanilla Ubuntu, and you don't think that vanilla is your "paradigm", I am at loss to understand why you think this thread is of any relevance to you whatsoever.

Perhaps you should go back to editing your files in Audacity. Forgive the interruption to your productivity, nothing to see here.

 

CT has (yet again) mounted her high horse (boring)...... , posted on December 1, 2010 at 12:54:24
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 CT, your own original thread title specified, and I quote, ".....Ubuntu *and derivatives*...."

So are you suddenly changing your own prescribed subject matter, so as to re-mount your high horse?

Few Ubuntu/Linux users go vanilla. That's the very reason they choose GNU/Open Source over others. Ubuntu/Linux can and should be tinkered with for optimal utilization. End-users fine-tune to suit. Geddit?

So have you installed and tested any "Ubuntu derivatives" such as Mint and all its remixed versions? Ubuntu Studio? Lubuntu? Pinguy? Zeven OS? Puppy Studio? etc etc......?

What were your findings with regard to audio applications? (Just mildly curious, that's all.)


-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

Yes it is, posted on December 1, 2010 at 12:55:16
John Swenson
Audiophile

Posts: 2422
Location: No. California
Joined: October 13, 2002
 If you are using gstreamer its easy, just change the output plugin to alsasink instead of pulsesink and any gstreamer app will go direct to alsa. The easiest way to do this is with the program gstreamer-properties, run it and it pops up a little tool, change the default output sink to ALSA and set the properties to the name of the device you want to send the audio to. Thats it, from then on any gstreamer app will go there unless it has its own UI for setting the output.

On the dmix issue, yep dmix is garbage, why use it? Dmix is an alsa plugin it doesn't HAVE to be there. If you set the alsa device to be the actual hardware device you are guaranteed to bypass dmix.

If you have an app that ONLY outputs to the default device you can always change the asoundrc file so the default device points directly to a specific hardware device rather than going through dmix.

So the setup to make things work nicely for good audio in linux is to disable pulseaudio so it doesn't grab your good sound device. Set gstreamer default sink to the specific alsa device, and change the asoundrc file so the default alsa device is pointing to your specific hardware device.

With that then anything should just work. Nothing gets resampled, the card automatically plays the samples at what ever rate is in the file.

You give up some things by doing this, without a mixer only one program can output to the output device at a time. This means you can't have GUI beeps and stuff come out over the good speakers.

There is one wrinkle here, you might need to setup a "plug" interface on your output device. For example a M-audio audiophile 24/96 ONLY takes 32bit samples. Very few applications actually output 32 bit samples. For this you need to setup a plug layer in front of the hardware driver. This will stuff zeros onto the back of the samples to keep the 24/96 happy. It will also automatically resample IF the hardware does not support the sample rate in the file. If you don't use a plug layer it will NEVER resample in the alsa layer, so if the hardware doesn't support the sample rate you get no sound at all. Many soundcards will automatically take either 16 or 24 bit data so you don't need a plug layer with them. Setting up a plug layer is just a couple lines in the asoundrc file.

It really isn't that hard.

John S.

 

RE: Great post!, posted on December 1, 2010 at 14:04:39
Presto
Audiophile

Posts: 5957
Location: Canada
Joined: November 10, 2004
 I agree all the way.

I used to want to get a multichannel audio interface for active tri-amping that was 24/192 or BUST. But I have very little 24/96 material here, and even less 24/192. I also do not believe in resampling to a higher sample rate if it means more succeptibility to jitter - this is what Dan Lavry believes. He won't build anything past 24/96 out of principle - and he builds some of the best stuff on the planet - with heater-stabilized crystals and the whole enchilada.

In fact, in recent years MOST of my purchases have been 16/44.1!

I can't be bothered with the recent "HD Tracks fiascos" and trying to determine which of my "HD" software came straight off digital masters, is resampled digital, or good old and potentially excellent sampled analog. I guess you need a "under investigation" folder and a "confirmed genuine HD" folder. Got kids - not that kind of time.

Yeah, when you can get RBCD sounding this good, who needs HD audio or even upsampling anyways?

Cheers,
Presto

 

Not sure why you are getting so excited, posted on December 1, 2010 at 16:15:44
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 This is what I said in my original post: "To those of you who are listening to hi-rez audio on Ubuntu and derivatives" ... "Make sure you change your PulseAudio settings."

I was attempting to provide advice to anyone running Ubuntu or any of its derivatives that include PulseAudio in the distribution. Perhaps foolishly, I thought that this advice may perhaps be useful to others in a similar situation.

Obviously, if you don't have PulseAudio in your system, or never listen to hi-rez audio, then this advice does not pertain to you.

 

Not for a general purpose system, posted on December 1, 2010 at 16:24:51
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 *** You give up some things by doing this, without a mixer only one program can output to the output device at a time. This means you can't have GUI beeps and stuff come out over the good speakers. ***

This is not acceptable for a general purpose system, not for me anyway. For a dedicated music player, perhaps. Some of us do use Ubuntu for other things besides listening to music.

 

RE: It's even worse than that, posted on December 1, 2010 at 18:39:43
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 "Actually it works very well if you just turn off all the mixer stuff and have the player go directly to the ALSA device. If your players insist on just going to the default device its really simple to reconfigure ALSA so the default device is your high quality music device. Once you do that nothing gets resampled unless the actual hardware can't handle the sample rate in the file.

John S."

I agree with this, and John's later posts in this thread. In my mpd.conf I have:

audio_output {
type "alsa"
name "Ayre QB-9"
device "hw:0,0" # optional

Nothing else needed and dmix is bypassed. Music Playes Daemon does have support for S24_3LE, so no need for plughw. It's very simple and works beautifully.

All this, of course, on a PC Engines Alix single board computer.

 

RE: Not that easy, posted on December 1, 2010 at 18:50:45
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 "All in all, audio on Linux is a real nightmare. Gone are the good old days when Linux apps could just change the OSS sample rate on the fly. Now we have OSS emulation on top of ALSA, ALSA emulation on top of OSS, pulseaudio as a virtual audio device for ALSA, JACK on top of pulseaudio, and forget any romantic notion of bit-perfect for anything other than 16/44.1!"

No, it really isn't. People do insist on complicating it, though, and for no good reason. Most Linux audio players have the option of talking to the sound card directly by specifying hw;0,0. You can even do it, as I did years ago, with something as ancient as XMMS (http://www.xmms.org/index.html) You get bit-perfect output with on the fly sample rate switching.

I currently get bit-perfect output and on-the-fly sample rate switching from from 44.1 all the way to 192 with nary a hiccup on USB Audio Class 2 hardware. It was that easy even three years ago.

Nick

 

RE: Not that easy, posted on December 1, 2010 at 18:54:37
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 Oops. Typo, I meant "No, it really *is* that easy..."

 

RE: Not that easy, posted on December 1, 2010 at 22:27:00
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 *** People do insist on complicating it, though, and for no good reason. ***

There is a good reason for that complication, at least for me. There is only one audio device on my system, and it needs to be shared across all apps, not just the audio player.

This is not a dedicated system for playing back music - this is a system for doing other things, music playback needs to be mixed in with system notification sounds, web browsing, video playback etc.

I understand that sharing the audio device across multiple apps may not be a requirement for you. However, we all have different requirements.

 

RE: It's even worse than that, posted on December 2, 2010 at 00:46:26
soundchekk
Audiophile

Posts: 2426
Joined: July 11, 2007
 If you're looking after best sound from your MPD setup you should try following:

If you're able to install ecasound on your Voyage Linux you should try the pipe output of mpd into ecasound and then to hw:0,0. Ecasound
comes with one of the best realtime-kernel allgined audio engines under
Linux.

I'm quite sure you'll hear a difference.

Something like below should do:

audio_output {
type "pipe"
name "pipe-eca-hw0-4416"
format "44100:16:2"
command "ecasound -q -B:rt -r:98 -b:64 -f:s16_le,2,44100,i -i:stdin -o:alsahw,0,0 2>/dev/null
enable "no"
}

Would be interesting to see how the Ayre DAC responds to that change.

Cheers

-----------------------------------------------------------------

blog latest >> The Audio Streaming Series - tuning kit pCP

 

RE: To those of you who are listening to hi-rez audio on Ubuntu and derivatives, posted on December 2, 2010 at 02:21:47
soundchekk
Audiophile

Posts: 2426
Joined: July 11, 2007
 Hi Christine.

Audio under Linux is a not that easy to handle. I do 100% agree.
The mixer stuff ( amixer/alsamixer/pulsaudio-mixer/jack-mixer/gnome-mixer/hdsp-mixer/.asoundrc/dmix/asf. asf.) for sure got a lot of space for improvements.

There should be one mixer and all others should refer to that one.

The problem around that audio subject I see, is that there is no central function in Linux land coordinating all this. There is Alsa, there is Gnome, there is the App.. So - who's in charge? I can tell you Alsa won'take the ball. If Ubuntu is not taking the ball, nobody will (can!?!?). That's my guess. From the past I know that Ubuntu is putting Audio not on highest priority. That's probably why e.g. Pulseaudio got that strong footprint into it.

However, usually there is a solution to almost every challenge under
Linux. Your dmix problem obviously belongs to the group of easier
to handle problems. If you go for best sound quality you just pass it by.
If sound quality isn't an issue you just use it. A little dmix finetung can be done inside .asoundrc.

Improvement proposals - which are very welcome - you should issue at the Alsa mailing list.

And don't forget you're in free-of-charge Linux land!
Don't complain - push for a solution by yourself.
Millions of voluntarily hours spent by thousands of unpaid volunteers should be respected.


Cheers

-----------------------------------------------------------------

blog latest >> The Audio Streaming Series - tuning kit pCP

 

RE: Not that easy, posted on December 2, 2010 at 07:20:27
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 "There is a good reason for that complication, at least for me. There is only one audio device on my system, and it needs to be shared across all apps, not just the audio player.

This is not a dedicated system for playing back music - this is a system for doing other things, music playback needs to be mixed in with system notification sounds, web browsing, video playback etc.

I understand that sharing the audio device across multiple apps may not be a requirement for you. However, we all have different requirements. "

Christine, let me see if I understand this: Your setup has nothing to do with an audiophile grade sound card? This is just an Ubuntu desktop that you're using as a personal computer?

 

RE: It's even worse than that, posted on December 2, 2010 at 07:23:48
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 Klaus,

I'll give it a try, but it might not be for a while. I don't understand why ecasound should sound better than mpd, but I'm willing to try the experiment.

Also, why "format "44100:16:2"?


Nick

 

RE: It's even worse than that, posted on December 2, 2010 at 08:04:54
soundchekk
Audiophile

Posts: 2426
Joined: July 11, 2007
 Hi Nick.

Just try it. It'll take you not even 5 minutes to get it going. ;)

>> I don't understand why ecasound should sound better than mpd

You might have followed some of my earlier excursions.

To be able to utilize the rt-kernel potential you need a proper setup
on the system. Just having the kernel in place won't change anything.

1. You as a normal user have to have the permissions to apply rt priorities
2. The application should utilize the rt-scheduler SCHED_FIFO.

Since MPD evolution goes very slow, I doubt that they've done
something about that matter during the last two years ( or since
0.15). Perhaps I'm wrong here.
ecasound is build for lowest latency operation using that SCHED_FIFO
scheduler. And that makes the difference.

>> Also, why "format "44100:16:2"?

That's just an example. There won't be a header with the PCM stream.
As far as I recall you need a fixed setup per sample rate.


Advise: If you got access to the kernel config. try to avoid dynamic bandwidth allocation on USB. That should make an audible difference
on your DAC.


Cheers


-----------------------------------------------------------------

blog latest >> The Audio Streaming Series - tuning kit pCP

 

Yep, posted on December 2, 2010 at 10:17:25
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 Laptop. Motherboard HD Audio. Cheap Sennheiser earbuds.

I used to have a dedicated external USB audio device, headphone amp, expensive phones (Etymotics, Audio Technica) blah blah blah

I looked like an audiophool in the plane and on the train. It took minutes to set up everything for a listen.

Then one day I realised that the sound straight out of the motherboard actually sounded pretty decent.

I haven't used the exotic stuff now for a year (apart from my Sleek Audio - will still use that on a plane), and if anything I enjoy music more because it's less cumbersome to set up. I can stop it and start it anytime.

Of course, I still have the dedicated player in the main system for serious listening. And my studio setup with the DAW etc.

 

Good post, and I agree entirely, posted on December 2, 2010 at 10:25:37
Christine Tham
Reviewer

Posts: 4839
Location: Sydney
Joined: December 29, 2001
 The biggest advantage of Linux is flexibility.

Don't like a particular solution, choose another.

Something not quite configured the way you want? Fix it.

Nothing meets your needs? Write your own.

I was hoping my post could be useful to a newbie to Linux listening to the standard players on a vanilla install. In my case, I left the default sampling rate to 44.1kHz but upped the bit depth to 32 bit floating. Distinct improvement, particularly when listening to 24-bit audio.

All pulseaudio really needs is a way to dynamically change the master rate just like Windows and MacOS. I hope that feature will be implemented soon, the developers say it's on their list of things to do.

 

RE: It's even worse than that, posted on December 2, 2010 at 17:18:35
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 >Just try it. It'll take you not even 5 minutes to get it going. ;)

The Hi-Fi is being moved to a new room, so everything is still in boxes. Once everything is settled in I'll give it a serious listen.

Though, I did wonder today if this all for naught when one has an Asynchronous USB DAC --the computer not having much say in the timing of things.

>Advise: If you got access to the kernel config. try to avoid dynamic bandwidth allocation on USB. That should make an audible difference
on your DAC.

I'll take a look when things are up and running.

Thanks, Klaus. Have a safe holiday.

Sincerely,

Nick

 

RE: It's even worse than that, posted on December 2, 2010 at 23:04:11
soundchekk
Audiophile

Posts: 2426
Joined: July 11, 2007
>> one has an Asynchronous USB DAC --the computer not having much say in the timing of things.

That's what marketing says. Yep.

If marketing would be right with those claims, Mercman wouldn't need Amarra,
Gordon and Steve wouldn't hear a difference between Flac and Wav. There wouldn't be a need for an Ayre player asf., asf.

Enjoy.

Cheers

-----------------------------------------------------------------

blog latest >> The Audio Streaming Series - tuning kit pCP

 

RE: It's even worse than that, posted on December 3, 2010 at 13:08:11
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 >>...one has an Asynchronous USB DAC --the computer not having much say >>in the timing of things.

>That's what marketing says. Yep.

>If marketing would be right with those claims, Mercman wouldn't need >Amarra,
>Gordon and Steve wouldn't hear a difference between Flac and Wav. There >wouldn't be a need for an Ayre player asf., asf.

I _only_ mentioned timing, *not* the fact that OS X can't natively handle sample rate switching --something that Ayre player will do for you. And, even though Gordon and Steve (Empirical Steve?) might hear a difference between the two formats, it doesn't necessarily mean that the format is specifically at fault. Who's to say that a specific type of noise/defect is being generated when it's being decoded on a certain (Gordon loves Mac) platform?? We just don't know.

Also, I just reread Ayer's marketing material and they never claimed that Asynchronous USB solved every computer audio problem. They claim "lower" jitter than previous USB solutions, that's about it. And I've never heard Gordon (vie email and on forums) say his asynch solution was all that was needed to make a DAC sound great --and I say this as someone who finds Gordon someone who is stubborn and difficult to communicate with. Can you point me to specific marketing you find offensive?

Lastly, I am almost certain of 3 things thus far:

1. Different audiophiles have their own stubborn ideas of what sounds "better", and it might not jive with mine.

2.Computer audiophiles will (unconsciously) use different computers/software/DAC's, etc. as exotic "tone controls" to try and rectify sins of omission/commission elsewhere in their systems.

3.Using personal computer hardware and desktop operating systems is _absolutely_ not the right tool for audiophile grade 2 channel playback.

3 succinct points against Gordon/Steve, Amarra, and differences in FLAC/WAV.

sincerely,

nick

 

Computer-derived audio is evolving...., posted on December 3, 2010 at 14:34:26
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 ....slowly, gradually.

BTW thus far I can clearly discern sonic differences between FLAC and WAV files on *all* hardware platforms and operating systems.

Agreed, in general, computers are not totally ideal for optimized audio playback. (There are intrinsic technical reasons that have already been well-elaborated many times on AA.)

Agreed, audio has always been, and shall remain, wholly subjective as the activity/process involves human physiology, perception and psycho-acoustics.

Recently I posted here regarding a surprise discovery in PC mainboard audio hardware:

"Another surprise was a VIA VT1708 sound chipset on a newly-installed ASUS M4A78LTMLE PC mainboard, powering a pair of Sennheiser HD-280 phones, and also stereo analogue outputs for amp and speakers. Entirely detailed, dynamic, unconstrained sound quality with remarkable imaging and separation. First integrated PC mainboard sound codec I've heard that warrants any recommendation.

VIA VT1708 Features
* Supports 44.1K/48K/96K/192KHz DAC Independent Sample Rate
* All ADCs Support 48K/192KHz Independent Sample Rate
* Built in High Quality Headphone Amplifier
Various Output Format
* 4 Stereo DACs Support 24-bit, 192KHz Samples
* DAC with 100dB S/N Ratio
* 2 Stereo ADCs Support 24-bit, 192KHz Samples
* ADC with 95dB S/N Ratio
* 8-Channels of DAC Support 16/20/24-bit PCM Format for 7.1 Audio Added-on Function
* High Quality Differential CD Input
* HPF In ADC Path for DC Removal
* Analog CD Input Path for Compatibility"

Sounds great with Linux. (Though obviously not as 'wunnerful' as my stand-alone audio system.......)


-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

RE: Great post!, posted on December 3, 2010 at 15:04:32
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 Thanks Presto. Glad you can enjoy your hobby to the max.

In general so-called 'high-rez' digital has been abysmally mismanaged by the industry. A great shame really. Like many others I have heard how good native DSD can sound. Industry under-performance usually boils down to white-knuckled DRM, outdated business models, lack of duty of care, and A&R incompetence.

Agreed, one should get on with enjoying the music, not proselytising on format, up-sampling, the latest mega-buck Über-Spieler (digital hardware depreciates at an obscenely gargantuan rate!) and gormlessly subsidising excessive record company profits by 'investing' in the umpteenth re-hashed re-issue of a 'mega-super-premium-audiophile-rare-metal-etched-dynamics-enriched-master-remixed-limited-edition-digipak-with-extras-and-out-takes-classic'. Why not purchase contemporary music on an affordable format that's been well-recorded and artfully mastered?

Some AA posters seem obsessed with 'correcting' our 'archaic, Luddite, retrogressive' habits/preferences. As in, so *all* that is new and foisted upon end-users by audio/music corporations *must* be good for us? Really? Relative to what?


-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

RE: Computer-derived audio is evolving...., posted on December 3, 2010 at 16:57:40
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 >thus far I can clearly discern sonic differences between FLAC and WAV files on *all* hardware platforms and operating systems.

Can you list *all* the hardware platforms and operating systems that you've critically listened to?

 

Windows, Mac and Linux - mostly custom workstations......., posted on December 3, 2010 at 22:32:30
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 ....just about all known hard disk partition and format types utilizing many audio apps. But will that affect your pre-emptive stance? Would you ever respect, or at least take seriously, the findings of another audio listener, when those findings differ from your own perceptions?

Nick stated:
"...Different audiophiles have their own stubborn ideas of what sounds "better", and it might not jive with mine...."

BTW may I ask (no offence intended) have you recorded, mixed and/or mastered live music? Are you a trained musician? A recording engineer? How many years experience do you have in high-end audio? Have you studied psycho-acoustics? Do you have relevant experience or other credentials in the field? (No, I'm not a self-anointed expert. Just curious to understand why you reckon FLAC is perfect and a few of us others are deaf. Did you program for the FLAC coding team perhaps?)


-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

RE: Windows, Mac and Linux - mostly custom workstations......., posted on December 4, 2010 at 09:43:49
nycparamedic
Audiophile

Posts: 111
Joined: December 27, 2008
 But will that affect your pre-emptive stance? Would you ever respect, or at least take seriously, the findings of another audio listener, when those findings differ from your own perceptions?

I never said I didn't take other people's findings serious. It's just very hard to understand (over email and forums) exactly what people mean when they report "better" or "worse" sound.

BTW may I ask (no offence intended) have you recorded, mixed and/or mastered live music?

No.

Are you a trained musician?

No.

A recording engineer?

No.


How many years experience do you have in high-end audio?

About 15. (But I don't think that makes me more of an authority on anything.)


Have you studied psycho-acoustics?

No.

Do you have relevant experience or other credentials in the field?

Linux user who (probably) first used the PC Engines embedded x86 boards in a serious attempt at an audiophile grade USB music server.

(No, I'm not a self-anointed expert. Just curious to understand why you reckon FLAC is perfect and a few of us others are deaf...

Did I say that? I'm rereading my post and wondering where the misunderstanding is. I don't think we're on the same page here. Or maybe a language barrier?

... Did you program for the FLAC coding team perhaps?)

No.

But I was sincere when I asked about all the different hardware platforms you've listened to. Have you listened to any embedded boards? Thanks.

Nick

 

Thanks for your response , posted on December 4, 2010 at 13:37:37
spindrifter
Audiophile

Posts: 375
Joined: February 20, 2002
 Indeed, posting about audio perceptions is approximate at best. The descriptor "better" often refers to an overall impression rather than specifics. I prefer specifics and if other posters describe similar aspects of their listening experiences, then maybe we can progress the discussion.

No, I haven't used "embedded boards" nor seriously considered a real-time USB-based audio signal source for my main system, owing to acknowledged technical shortcomings of USB throughput in a noisy, SMPS-powered PC environment. I feed my PC with a simple external USB HDD "music server". I enjoy good sonics while listening to WAV files with Sennheiser HD280s driven by that VIA VT1708S codec on the ASUS mainboard.

Naturally progress on the USB front is being made. One day I might well take the plunge when USB is ready. Maybe you feel that is feasible right now? But are we talking full-bandwidth, unimpaired, high-fidelity and no glitches whatsoever?

As to your emphatic comments about FLAC and those that can find sonic issues with it, perhaps I misunderstood or over-stated your position? Is FLAC your reference file format?

With HDD and SSD storage costs decreasing monthly is there really any point in ripping compressed music files? Why not stick with the original, uncompressed, "full-bandwidth" format?


-spindrifter-
"We have now sunk to a depth at which the restatement of the (bleeding) obvious is the first duty of intelligent men." — George Orwell

 

RE: Update: PulseAudio currently does not support dynamically changing sample rate, posted on December 6, 2010 at 16:59:24
telackey
Audiophile

Posts: 289
Location: Oklahoma
Joined: January 15, 2006
 I do that on my Windows machine and it is a world better than mixing music and general sounds through the same output. All the beeps and rings go to little dinky speakers; music goes to my "real" speakers.

Besides the sound quality advantage, and perhaps more important, is that I can crank up the music and not have my eardrums explode if I get an IM.

 

RE: Not for a general purpose system, posted on December 6, 2010 at 17:11:13
telackey
Audiophile

Posts: 289
Location: Oklahoma
Joined: January 15, 2006
 He is suggesting that you buy two sets of speakers. Set a little dinky pair of $5 speakers as the default output, and then configure your music player (and only your music player) to output to your real speakers.

That is actually more convenient on a general-use computer because it is highly unlikely you want your music and your general computer sounds (IM/e-mail notices, etc) to be played at the same volume anyway.

 

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