A Sound Server For X
An anonymous reader writes "X.org, the organization that governs the evolution of the X11R6 specifications, has released a sound server for X, called MAS. According to their site: 'MAS integrates with a compatible X11 server on your desktop. It processes graphic information locally, alleviating the need for network transmission of uncompressed graphical content. Graphic events are easily synchronized with audio events for professional-quality multimedia and accessibility-enabled applications.'" The X.org site describes MAS as an "affiliated technology" rather than "official," but it is released under the same license. "MAS" stands for "Media Application Server," and it's developed by Shiman Associates.
XMAS
Ahh.... how sweet. How gay.
Any practical applications for this?
God is real unless declared integer.
With the sound and graphic capabilities, it sounds like a great start for a full blown MULTIMEDIA server
There are already a bunch of remote sound servers. I happen to like rplay, but there is also artsd, nas, esd, etc... Was the problem simply NIH syndrome?
I'm all for this. Anybody who's dealt with esound knows that XMAS or anything else would be an improvement.
From all the experience I've had with X applications, a sound server is the first thing that they need.
I've used a lot of X apps that have crashed due to bad requests (especially those dealing with unsupported colourmaps), and it makes sense that someone would shore up the current state of the code and work on stability. But why has it taken so long?
At any rate, this is a good step forward--hopefully we'll see more X-based apps improved as to stability and speed (X isn't the speediest thing in the world).
Karma: Excellent Birds (mostly as a result of listening to Laurie Anderson)
Can those knowledgeable with the aRts sound server stand up and tell us what this means for Linux and KDE. Is this a better approach or is aRTs better?
Wait a few days and you'll see RIAA trying to sue them :o)
ESound? Asd? ARTs? It seems a little different in concept, but I just can't get it. If it is, so cheers to the guys that made it... Linux at least (as I understand this is for X, so *BSDs and other *nix should benefit too) need a more standardized sound architecture (Yeah, I know about ALSA, but I mean something more higher level - like DirectSound)
It processes graphic information locally, alleviating the need for network transmission of uncompressed graphical content.
Since it relies on X11 I suspect the bandwidth requirements are going to be really high. X11 over the network is a bandwidth hog, that's all there is to it. Now they're adding sound?
X11 needs a new protocol. Graphical applications run across the network consume ridiculous amounts of bandwidth. If you want to do a test try running the XMMS gui across the network via X11. The last time I did it, XMMS was using 11 megabits per second. It would really suck to try that over a modem or a 64K frame-relay link.
Perhaps this will kick some audio hardware makers into releasing some drivers for the Penguin.
I really wish I could run my Aardvark Q10 (awesome hardware) under something other than MS - they were working on Beos drivers before the death of Be. Those are probably sitting on a floppy in a file cabinet. :(
Great, all the people who think X is too bloated as it is will now be crying "No mas!"
Hey, is it just me, or has anyone else noticed: X-MAS = Christmas?
Yawn.
This will save us from having to configure each programs sound device. =) /me waves @ dsp/esd/arts
Pixels keep you awake!
Lots of neat applications, actually. One of my favorites is a network monitoring room... For instance: network monitoring apps sample your network traffic once a second... while bandwidth and processor utilization of your servers is within preset values, an audio file of a creek is played over your X-enabled speaker system. When congestion occurs, you here a new audio file played (which is, of course, mixed into the original creek audio-file) of a herd of cows drinking water, or something... When a router or server goes down, an alarm is triggered, and a flock of crows start caw-ing; or an elephant trumpets, or whatever...
The point is, when everything is going fine, the audio environment of your network control room sounds like a peacefull woodland setting, or something. When something goes wrong, you hear the animals going crazy. It's a really, really great way to monitor a large network, without being glued to a network monitor all the time.
Typically, an X-server would alert you to each of these above mentioned alerts and statistics by displaying a video-file of some sort, which is displayed on your video monitor (CRT, LCD). With an X-sound-server, you can pipe the same alerts and statistics to audio-files which are 'displayed' on your sound monitors (speakers).
All my computers run BSD and X10 itself ran on Ultrix, which was BSD-based. So my most important question is does MAS run on FreeBSD?
Pardon me, but you need to get your TLAs ;-P
corrected - MAS stands for Multi-Agent System
My other Beowulf cluster is... er...
I would not like fuck around with anti-aliasing/recompiling/enabling_byte_code_rende ring with sound files !!
Thats the real question...
Today is a gift. Save the receipt.
What I want to know is how does one obtain a single letter domain. i.e. x ???
"Don't Follow Leaders." Bob Dylan
There is already a MAS associated with audio -- MOTU Audio System plugins.l obal/index .html)
(http://www.motu.com/english/software/g
Perhaps they should have done a bit of name research.
because it keeps /dev/dsp from fucking up, allows simultaneous feeds, and doesn't give me crap.
/dev/dsp replacement be made with the same features?
Couldn't a
You can't judge a book by the way it wears its hair.
A clipboard server! Has anyone else noticed the really unpredictable behavior of the clipboard in X applications?
A sound, solid server will certainly help with all sorts of stability problems I've experienced with X over the years.
... you meant sound as in audio? Oh, nevermind.
Oh wait
The last big push from the Open Group was Broadway, which was an X protocol based plugin for web browsers. Look at how popular it is today! Their XPrint work is just as successful.
I don't want a bloated with an ancient feel for a sound server.
Linux needs to be saved from X and not to use it even more.
God no.
What does the National Institutes of Health have to do with this? :_|
NIH = Not Invented Here. (Implication: So let's ignore it and reinvent this wheel our own way. Like maybe without that obnoxious radial symmetry. Besides, a round wheel might violate some patent. So let's have lots of engineer fun and waste lots of money, instead of pulling an existing design off the shelf, filing off a few rough spots so it will fit, and installing it.)
NIH goes along with management that thinks you need young developers who are constantly creating (and will reinvent the bubble sort), rather than experienced developers who already have the answers in the can (and will pull down their copy of Knuth Vol III and pick the right sort for the job.)
Which is not to say that I agree with the poster's conclusion that they may be ignoring fine solutions in order to construct one of their own. Integrating the video and audio server and protocols - rather than grafting audio onto an existing video server (which is in turn grafted onto something originally designed for more static displays) - is the right solution for synchronized video/audio. And the integration may have different problems than gluing the bag onto the side of the kludge.
Bantam Dominique roosters crow a four-note song. Once you've heard it as "Happy BIRTHday" you can't NOT hear it that way
And for those who are asking what's wrong with existing sound servers: there's no standard mechanism to query whether one of them is running, especially on a remote machine. (No, relying on magic port numbers is wrong.) And not all of these servers run under all Unix variants. I personally have had very hard time trying to make aRTs work under comercial unices. With this stuff you just talk to X server. There's hope big players will support it because X is the industry standard. esd, to put it bluntly, isn't.
Heh. I did a double take just then, when I thought I read Sharman associates instead of Shiman-- Well, p2p is just like Sound servers, no? ^_^
Recursive (adj.): see 'Recursive'
Key features of the Network Audio System include:
- Device-independent audio over the network
- Lots of audio file and data formats
- Can store sounds in server for rapid replay
- Extensive mixing, separating, and manipulation of audio data
- Simultaneous use of audio devices by multiple applications
- Use by a growing number of ISVs
- Small size
- Free! No obnoxious licensing terms
Applications that support NAS natively:- Festival - The Festival Speech Synthesis System.
- mpg123 - a command line MP3 player
- GAIM - a free AOL IM client
- OpenOffice (StarOffice) - the (now opensourced) StarOffice Suite has built-in NAS support for the Solaris and Linux Platforms.
- The Qt Library - from Trolltech supports NAS natively. You will need to pass the '-system-nas-sound' to './configure' before building.
- libSDL - SDL, the Simple DirectMedia Layer library, now has native NAS support thanks to Erik Inge Bols\x{00F8}
- XAnim - the X Animation viewer
- XBoing - a blockout type X game
- XPilot - a multiplayer client/server space warfare game
- Xemacs - the best cross-plaform, cross-language IDE
- Alsaplayer - A NAS Output plugin written by Erik Inge Bols\x{00F8} is now supplied with the Alsaplayer distribution.
- X MultiMedia System (XMMS). A NAS Output plugin written by Willem Monsuwe is available at ftp://ftp.stack.nl/pub/users/willem/
- Wine. A NAS plugin written by Nicolas Escuder is now available with the WINE distrubution.
Not a trollLess is more !
I am a composer/musician using Linux for my needs and from what I've seen so far all of the aforementioned servers esd, asd, artsd etc. are rather rudimentary and incomplete. Furthermore they have monstrous latencies (which is a big problem for any kind of time-sensitive interactive music).
To this date the best audio server on Linux is JACK, but unfortunately just as any other of the current servers, it is still under development. Nonetheless, its advantages over the given bunch of audio servers is immeasurable:
1) absolutely the best latencies (including even other OS's) down to 2.5-3 milliseconds (you need to patch the kernel for low-latency operation though since the vanilla Linux kernel 2.4 and down has some big issues with the scheduler). All this is achievable even with the current version, even though it is under development.
2) integration where the signal could be routed not only to the dsp but also between the apps (so you could theoretically get the signal routed from xmms into your real-time sound processing app and then to the audio recorder and audio output).
3) Relatively easy to implement and allowing for practically unlimited number of connects
4) Relatively low overhead
Unfortunately, as I mentioned before it is still under development and has its own issues that need attention. Furthermore, apps need to be adapted to be compatible with this server (in another words this server is not trying to be compatible with older servers simply because they are inferior, dead-end kinds of implementations). That being said, it is still the best sound server around.
I would really like to see Linux community let the audio programmers/musicians to provide solution to this issue, because they are the ones who know the best how to create the best possible sound server that will suit their highly demanding needs and yet provide a great architecture for the rest of the casual users.
It is unfortunate though that instead of unifying our efforts everyone feels like they need to make a brand new implementation of the same idea instead of contributing to the projects that are already semi-mature and need further assistance in development. Because of this "so-called-diversity" we now have half-dozen sound servers out of which 90% blow chunks, while the other 10% have a great potential but are incomplete.
So, if you don't have artsd or esd, what is taking care of your sound? I would like to know, because I get less than acceptible results at the moment. I have an onboard ac '97 codec (via chipset) that works fine in windows, blows chunks on Linux, because Arts "overloads" or something like that (I can't remember the error message exactly.) Plus, there is a huge lag in sound on games. Example, in gltron, the sound is about 2 seconds behind. I tried switching to ESD, and although it worked better than artsd, there is still some lag at times. Same goes for Tux racer. Meanwhile, the graphics card never skips a beat.
Oh, no, Linux isn't fragmented. Not in the slighest. Say, is KDE the one with the cute footprint icon? I can never remember.
Run!! Bad Puns!!!
A Minesweeper clone that doesn't suck
h0h0h0 Merry XMAS one and all.
Anyone remember AudioFile? It was an audio server whose architecture was roughly similar to X.
Why not just a audio file yelling, "Hey jerkoff! Your router just cacked!"?
I got excited when I read the word "synchronized"
thinking this might be a mixer with SMPTE timecode capabilities. One of the most important distinctions between "pro" A/V and "consumer" A/V is whether the tracks have a standard sync encoding. Without that, the best you can hope for is "close", and that is not acceptable for broadcast or production.
This requirement is serious enough that pro gear has an input for a real hardware clock.
Sync capabilities and clock controlled sampling are a few of the keys to us having pro audio production on inexpensive hardware. Unfortunately, the barriers to entry remain outrageously high.
For most PC users, the sound card is strictly an output device. few will spend more than $100.00 on it, and even fewer will have audio tracks for which the bitrate or noise floor of these cards is a limiting factor. We tend to regard the sound card as a device for playing back audio that is at most, 44.1Khz, and at the deepest, 16 bit.
For the rest, the sound card is an input device. As such, it theoretically could be as good an input device as that found on a $100,000 audio workstation, and there isn't really a good reason why we CAN'T make the poor-man's DAW if we use one of the better sound cards as a starting point.
A timecoded 24-bit audio track which has not been resampled for the finished output is a minimum for "professional" work, and we *almost* have it on consumer gear! Everything except real timecode.
I place this problem in the same category as the absense of colorspace management on The Gimp. Those who understand what it is, realize that it makes the difference of whether they can use it professionally; those who do not understand, don't care, and don't realize the magnitude of the problem.
Slightly offtopic, but I figured I might as well take a shot- Has anyone successfuly piped audio from a Unix machine to a windows machine, across the network? I got it working using esd and a java version of esd, however the sound quality sucked (and java's sound support didn't like my 5.1 sound, I wouldn't mind but it had 0 bass).
I've looked but can not find a native sound server for windows at all, in any form. Even somthing compiled with cygwin would probably work, but no luck there either.
esound is dying.
This major code release is only our first step. We're putting the MAS core out there in order to create a useful open standard. We feel that this code represents a radical departure from prior attempts to manage time-critical data on the desktop and across the network. We see this as a valid and innovative architecture with extensive implications for the management of media of all modes. In particular, one of the important applications of this architecture is to time-critical accessibility tasks which can now be handled in a completely generic fashion.
We encourage close examination of the code and we will do our best in the near term to bring as much of our insight and intention to the open community. We see this as an opportunity for collective and collaborative innovation.
From our website:
MAS will provide a complete mechanism for media support, for all pluri-modal media, for all platforms, for all operating systems, for all window systems.
MAS supports the desktop and, transparently, the network. In particular, MAS will provide complete support for the X Window System, across the network.
MAS is an open system: the complete core will remain under the original MIT ("X") license, equally supporting open and proprietary use and development.
MAS provides mechanisms for structured extension, and will be supported by dedicated testing and certification processes.
We wish explicitly to thank Sun Microsystems and X.org for their generous support.
Leon Shiman for the MAS Development Team
Hmmm... people seem to have liked the other post, so I'll offer some other (potential) practical applications for the sound server. Just my 2 cents...
1. CAVE environments. Anybody who's worked in an X11 CAVE environment knows that X can handle video cube arangements. Maybe not the most elegent way to run a cave, but it's do-able. X-sound-server can then provide 3D sound support to cave applications.
2. PACS environments (terminal services). Do you have a *nix based picture archiving and communication system (PACS)? For example: a hospital or library kiosk system. Now, your PACS is an audio environment as well.
3. Video Jockeying (VJ). If you're running a linux based VJ operation at a nightclub or dance hall, audio support is now available via X. You can now synchronize your video panels and speakers with the same daemon... Check out JMAX for more information...
4. Voice-over-IP kludge. As microphones are basically just speakers operating in reverse, theoretically, the X-sound-server should support microphones at some level... Hack your X11 system to support XVOIP!
I can't comment specifically on arts or NAS, but I know for sure that esound is unusable for network audio. It cannot synchronise audio with video and there is no upper bound on latency. Alan Cox has an amusing quote about esound: something similar to "esound is just about good enough to make a boing noise when you click something, but it's otherwise useless".
I don't think I'd be stretching my luck to say that arts and NAS have similar limitations. I'll be interested to see if MAS is going to solve these problems, or if it's just another half-assed attempt.
At least to my way of thinking, one of the Achilles' heels of Linux has been the lack of a unified sound system. We've got the graphics more or less together, but sound has always been quite spotty. Hopefully, this will be the magic bullet which starts getting sound card manufacturers to give Linux and other *NIXs the support they deserve.
-Erwos
Plausible conjecture should not be misrepresented as proof positive.
from their site:
moreover, X license allows companies like Sun or HP to incorporate MAS into their systems. they are not always happy with GPL.BTW, their web suxx indeed. nasty frames... :/
this will not let XF86 get better!!!!
I am the Alpha and the Omega-3
Ok.. I have to say that this, for some godawful reason, is probably the most intelligent post I've seen on this subject on /.. I've been a musician for 16 years, and one of the things I really kinda missed when I made the move to Linux was the lack of really good sound support.
IMO, If this is as good as others say it could be, and if it can provide kernel-level (maybe, with source patches) timecode and sound driver support, they'd really have something. I'd love to record/mix my own stuff on cheap hardware and wouldn't think twice to put on a second X server to do it.
You are all fartheads.
Well, we looked hard at the Network Audio System (NAS), and at DEC AudioFile, and esound and aRTs. And we looked hard at the efforts of Tice and Welch at the X Consortium whose X Common Audio project was to take the best of the NAS and DEC AudioFile to form the standard solution. And, we looked hard at all kinds of other soundserver-related things out there.
Our conclusion: none of them had the kind of system-pervasive time management one needs to handle sound, let alone video or other time-dependent information. The system that came closest to meeting our timing needs was DEC AudioFile, but it was not extensible enough for our needs, and lacked support for sample rate conversion.
We didn't take the conclusion lightly. As you say, there's all these applications that support the Network Audio System natively. There's all those old NCD X tubes that support it in firmware, too. And, I really have no interest in stepping away from code that has most of its bugs worked out. But, we just did not see how NAS would scale in both performance and in functionality to handle the kind of high-performance multi-media we all want to see on Linux and UNIX.
I think MAS is a great platform to handle the timing. It's young, though. We're working hard now on a soundserver-style API to ride on top of the lower level core API that's in the source distribution. Beyond that, there are a host of security issues to work through, and the X.org standards process. (Also on our to-do list is a more detailed developement roadmap for the website!)
I think there's a ton of cool, useful stuff in the core of MAS now. For instance, we compute a running estimate of your soundcard's actual sampling rate. You can use that estimate to drive the sample rate converter (srate) device to resample audio to the actual rate of your sound card. We've been appalled at how far off some soundcards' crystals have been!
Please be patient if you e-mail us... We're getting a lot of e-mails for some reason.
Thanks,
Mike Andrews
Lead Developer, Media Application Server (MAS) Project
You can improve things further by using LBX (low-bandwidth X), by running "lbxproxy".
It does some X protocol aware compression. In addition, and perhaps more importantly, it caches some server information that applications commonly re-request over and over again unnecessarily.
Basically, all you need to do is:
$ lbxproxy &
$ export DISPLAY=:63
$ xterm&
For instance, we compute a running estimate of your soundcard's actual sampling rate. You can use that estimate to drive the sample rate converter (srate) device to resample audio to the actual rate of your sound card.
Now THAT's hardcore. If only this existed two years ago...
THIS THING CAN TURN ON A DIME, MACROSSZERO STYLE ALSO FUCK BETA, ~NYORON
Likewise, we don't need another network audio codebase, we need a good network audio protocol. It looks like MAS provides that. As a bonus, you also get a reference implementation.
I agree with this, and I think that these are areas that will take Open Source software to the next level.
-- the computer doesn't want any beer, no matter how much you think it does. NEVER, EVER feed your computer beer.
Does anybody know if MAS (or alternatively jack) can handle a driver for 3D audio rendering, not unlike DRI+Mesa in XF86?
Comment removed based on user account deletion
I think this could definitely be a step forward. That is if anyone uses it. If people actually decide to use this system it would be really nice to be able to have remote logins and have sound play over the X11R6...
Two things that really hinder open source software:
1) Developers don't plan it out and the code looks like muck.
2) People don't pay any attention to the project.
One thing you need to consider is bandwidth, if you apply any amount of decent compression to X11 data, it could be reduce bandwidth usage by a huge amount.
Perhaps if someone gets a decent setup, you could have people producing dummy terminals that would login to an X11 applicaiton server. It wouldn't be able to run games or anything, but it would definitely be something to consider. I know several schools that use the m$ solution to that, maybe we could get X11 to do that also? The only thing you run into then is removable disk access, printers, usb devices (usb storage).
I'm no x11 genius, but do the current audio servers run over x11, or is it something that needs to be compiled into kde apps, or gnome apps? and can the x11 protocol be compressed?
That's my $.02
Just a pondering suggestion though. First off I know this is way off topic and so on. But why hasn't there been a "strongly" supported or stable project on developing a GUI that is more of a client based system rather then a server? I think if Linux want to advance more towards the desktop users they need to begin to develop for them.
+++ David Watts 5495 0.0 0.5 1888 884
You only need the Windows Sound System for "gee neato" sound effects from Windows. I'm not aware of any game or program personally that specifically requires Windows Sound System to be active. CDPLAY.EXE for example can output directly to a SoundBlaster using a TSR.
Could someone clear up the situation for some of the newbies.
ALSA is a sound server... MAS is the same? Would they work together, or be distinct choices. What advantages does MAS have over ALSA? What drawbacks?
Thank you for your time,
Anon
For people wanting to use X over low bandwidth connections, the X consortium (?) in their infinite wisdom invented LBX ("low bandwidth X").
Note that I don't know much about LBX except for the above, so if you used it and you think it sucks, it probably does. My point is that the X Consortium hasn't been ignoring bandwidth as you seem to suggest.
Installed the Bubblemon yet?
Why do we need an audio server for local applications, when almost every soundcard is getting hardware mixing drivers?
Why MAS, when there already is MAS? To all non-musicians: MAS (from MotU) is a widely used audio system in the professional audio world.
the name of this is X-MAS?
Then *I* must do it, and so I will!
Wow, X-MAS is early this year. <grin>
So, maybe somewhere around X-MAS, Linux will have X-MAS builtin? <heh heh> <snarf>
So, is it Free as in "free from school/ work with X-MAS", or Free as in "a gift from Santa"? <heh> <yeah yeah> <snarf>
"We can confirm that Debian does *not* ship the version with the trojan horse. Our version predates it." [CA-2002-28]
The problem is that without some level of integration with the video system (i.e. X), synchronizing full audio/video streams via an external daemon mixing process will result in noticeable latency. That latency can be minimized to about 100 ms or so before you run into the problem of "stutterring" audio streams (audio not reaching the mixer fast enough). This latency is irrelevant for "long, continuous" streams (such as mp3's), and barely noticeable for "sporadic" events (window manager squonks, game bleeps, etc.).
Five years ago, when esd, arts, etc. were in design, the majority of the (Linux) desktop environments that were in existence were limited to xanim and realplayer for a/v playback, which only played back a limited amount of the media available at the time. I figured allowing the capability to have ESD release control of the sound card for "incompatible" programs (quake, low latency audio mastering software, etc.) would be sufficient.
And judging by the fact that it's still available on most distributions today, that original feature set (hear messenging and wm beeps while playing mp3s) seems to hold up well. The problem is going to the next level beyond the external audio mixer, into the realm of synchronized A/V streams (much more prevalent today) and for that an integrated solution is necessary. NAS was the closest thing "back then", i.e. 5 years ago, and I couldn't get it working on Linux for more than two seconds, regardless of whose patches I applied. I assume none of the other mixing daemon authors could either...
How's my programming? Call 1-800-DEV-NULL
That is not "insightful."
/dev/dsp, /dev/mixer, /dev/midi, you don't need some special daemon when sound is built into the operating system.
Linux has
Look towards the end of this paper. It describes that "dangling string network monitor".
Let me say this again. X11 is a bandwidth hog. Numerous posts above defend it and try to explain why it uses so much bandwidth. A few posts seem like nothing more than defenses based on emotional attachment to X11. But, the fact remains that it uses too much bandwidth.
Numerous posts blame the high bandwidth on "poorly designed applications". This says that there needs to be a different protocol. It should not matter what the application is or how badly designed it is the display server should be able to reliably and consistantly export the display, regardless of the underlying application. If it shows up on the screen the display server should be able to push it across the network at a reasonable cost(bandwidth).
A few incredulous posts say that the problem is my use of "eye candy". Excuse me???? Remember that we are talking about X. X is a *GUI* display manager. If we cut out the GUI part, then we don't need X at all and can happily rely on SSH. The whole point of X is eye candy. What kind of display manager is it if it chokes on the *display*.
A few posts say that my numbers are way off base. They site their own experiences at 2Mbps and 200kbps. Regarless of my numbers, 200kbps for a static image is WAY too high, let alone 2Mbps or 11Mbps. See below for more acceptable bandwidth numbers.
As is always the case a few people piped up and offered VNC. VNC is a very nice remote control package but, it really can't be compared to X11 and similar products since VNC does not offer the same functionality. VNC does allow you to remotely control a desktop but X11, RDP and ICA do much more than that.
Which brings us to examples of protocols that offer similar X11 functionality at a MUCH lower bandwidth. VNC is relativley low bandwidth but, as stated above, is less functional than X. RDP consumes about the same amount of bandwidth, up to 200Kbps, but provides much greater graphics quality and far greater responsiveness than VNC or X. But, Citrix ICA is the hands down winner. It offers very similar functioinality to X much better performance than X, VNC or RDP and it is the lowest consumer of bandwidth.
A standard desktop application(not too graphics intensive) can consume as little as 14Kbps. It is possible to have 32 bit color depth with animated graphics(even a movie) and sound running on Citrix ICA at 200kbps or less. The average ICA session consumes 30kbps.
All this means that it *IS* possible to have a networkable display that consumes a minimal amount of bandwidth. X11 using 200kbps to 11Mbps is WAY too high. X needs a new protocol.
and get the better drivers now!
At some point you will come across that needed driver that your distro's kernel doesn't have; until then, enjoy not worrying that you forgot to reload your bootloader after compiling.
You can't judge a book by the way it wears its hair.
As if there weren't already a problem of too many existing solutions to *nix audio... We just need to pick something and make it a standard. Think about the success of X. Sure, there are much better tools out there interacting with users in a graphical manner, but X has worked so well because the unix community adopted it as a standard.
Think about how much code gets re-written. arts, esound, jack, and now mas. We really should just pick one and go for it. If you think it's lacking functionality, then change it, for crying out loud! That's what open source is all about!
At a certain point when everyone's running around like chickens with their heads cut off, it begins to appear that a poor standard would be better than no standard at all.