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Thread: Intel On Rebuilding The X.Org Linux Desktop

  1. #41
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    Quote Originally Posted by drag View Post
    Flash sucks. Get over it.

    The reason the "rest of the world" uses it is just illustrates the piss-poor state of Windows default players, not how wonderful or useful flash is.

    People tried for ages to embed quicktime or realplayer or windows media player bullshit into their browsers and that software is so shitty that most users rejected it outright.

    Flash video players exist not because they are so good, but because WMP is so shitty.

    In Linux we don't have this problem. VLC or Mplayer or Totem can handle playing videos just fine.

    Flash is wonderful and useful if you like:
    1. Having videos play using twice the CPU and twice the memory usage
    2. You like having to re-download the same video over and over and over again each time you want to use it.
    3. You like having long waits and really really really really shitty seek performance.

    Hell I can't even figure out how to make this particular flash player seek at all. I guess I have no choice but to sit and watch the stupid thing all the way through and god forbid I want to close my browser.

    Because I LOVE the fact that I have to run proprietary software that has shitty and buggy performance compared to the number of high quality and fast media players.


    Anyways the quality of the video is extremely poor and you can't hear what the person is saying nor what is on the screen. Using mplayer I can make it somewhat bearable and control audio through filters and whatnot to make it hearable, but on this flash player it is impossible.



    Let me see how critically difficult it is to play videos that I have to download.

    Hrmm....

    1. Click download link
    wait
    2. Video plays.

    Your right. That is way too difficult.
    Well said, drag!

    I really hope those running Phoronix consider what you've written as well as others who have echoed your sentiments.

  2. #42
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    Hmm.. I'm not anti-flash, but would definitely prefer a free alternative! I Look forward to the ogg video html inclusion in Firefox 3.1, and it would be only logical for Phoronix to make use of that. On one condition:

    Don't ever post a POS quality video (also the TI mini-projector video) like that again! Seriously, have Phoronix checked the videos before posting them? Assuming you did, where is the "Excuse us for the technical problems, and expect a shock if you wear headphones" note? I appreciate Phoronix trying to deliver videos from conferences, but in these cases it would be more professional to simply remove them. Also, why post videos at all, when all they show (or not show..) is irrelevant? Delivering mp3/ogg of the speeches would make more sense.
    Last edited by numasan; 02-10-2009 at 08:50 PM.

  3. #43
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    Quote Originally Posted by numasan View Post
    Hmm.. I'm not anti-flash, but would definitely prefer a free alternative! I Look forward to the ogg video html inclusion in Firefox 3.1, and it would be only logical for Phoronix to make use of that. On one condition:

    Don't ever post a POS quality video (also the TI mini-projector video) like that again! Seriously, have Phoronix checked the videos before posting them? Assuming you did, where is the "Excuse us for the technical problems, and expect a shock if you wear headphones" note? I appreciate Phoronix trying to deliver videos from conferences, but in these cases it would be more professional to simply remove them. Also, why post videos at all, when all they show (or not show..) is irrelevant? Delivering mp3/ogg of the speeches would make more sense.
    Were you still watching by the last half of the first talk, when keithp got up and was demoing keystone-distortion correction with randr 1.3? You could see to white part of the projector output make the shapes he was talking about reshaping his desktop to... See, the video wasn't useless. But yeah, that was about it for the utility of the video.

    I would have liked phoronix to mention that the quality sucks and you can't see much, so only take the time to listen (and probably not watch) if you're really interested. And seriously, edit the second vid to remove the minutes-long sections of lost audio. But I don't wish that the videos were taken down or never posted. I'm glad I heard the talks. I find it interesting to know something about the personality of the devs, which you can tell from the way they joke about things in their talk, and stuff like that.

  4. #44
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    Quote Originally Posted by KDesk View Post
    It is also possible to change the resampling of dmix in Alsa and force it to 44.1KHz, without the need of the useless and deprecated Open Sound System.
    That just changes to another fixed frequency. I don't want any 48kHz things resampled to 44.1kHz either! Although it's mostly movies, which I play with mplayer which I have configured to use the hw pcm, not the default dmix pcm. They use the TOSLINK passthrough for AC3 & DTS, anyway. Still, there are games that output 48kHz, and stuff.

    Still, probably useful. Thanks.

    Wasn't Intel planning to make a GPU with dedicated video memory? Why they don't make a working GEM/TTM and EXA/UXA for every one, not only for one graphics company...
    EXA does work for everyone. One of the talks, probably Eric's IIRC, referred to EXA as being mostly the radeon accel architecture. As other have said, other drivers can convert to UXA and/or GEM whenever they want.

    It makes a lot of sense to do all this experimental work on just one driver (intel), because they've had to go back and re-write whole chunks of things after getting feedback from the kernel people, for example. So if they'd spent extra time because of more kinds of HW to worry about when they were just getting the first version written, it would have been more wasted work that they had to throw out.

    There's also the fact that several of the key people (keithp and Eric Anholt, for example) are Intel employees paid to make the Intel driver work well. It's great that they're doing it by making the whole infrastructure suck less. (listen to the talks about how much memory copying used to happen, and how mesa used to need to keep a second copy of every texture in case the (lack of) memory manager tossed the old one. I remember vegastrike using huge amounts of memory on g965 a couple years ago. I think it's better now, but I also have 4GB instead of 2GB, and a lot of things have changed in vs (precompressed textures!)...)
    Last edited by llama; 02-11-2009 at 12:08 AM.

  5. #45
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    Quote Originally Posted by drag View Post
    Well it's the audio card driver to know what sound formats the cards can accept and then translate that into something that the card can receive without puking or having high cpu load.
    That's not a driver problem nor sound card problem, that's ALSA's problem.
    http://alsa.opensrc.org/index.php/Dm...und_quality.3F
    Quote Originally Posted by alsa.opensrc.org
    Dmix by default uses 48kHz sample rate. So, if your source is 44.1kHz, it will be upsampled to 48khz.
    To resample from 44.1KHz to 48KHz is bad for the quality and the cpu usage.

    Quote Originally Posted by drag View Post
    So if people are having issues with wrong audio formats being sent to the card then this is a driver issue and if you don't want to deal with it in the future a good way is to file a bug report and hopefully it'll get fixed.
    The problem has always been arts, esd, dmix, and currently it is Dmix or pulseaudio. If you want higher resample quality the CPU usage increases.
    A dmix as a software mixer is bad, but now it's better than it was some years ago.

    But sure in many cases are driver bugs.


    And yes, Flash sucks, even more if it's close sources, it sucks more!

    Quote Originally Posted by RealNC
    I'm using OSS 4, not the deprecated OSS inside the Linux kernel.
    I haven't used OSS 4.x so I can't know if it is complicated to configure or not, but to install and configure OSS without having problems... to many problems... like with pulseaudio.

    Quote Originally Posted by llama
    That just changes to another fixed frequency. I don't want any 48kHz things resampled to 44.1kHz either! Although it's mostly movies, which I play with mplayer which I have configured to use the hw pcm, not the default dmix pcm. To use the passthrough for AC3 & DTS. Still, there are games that output 48kHz, and stuff.
    Yes, you are right, that's a problem, it is fixed. But to mix the sound, it has to be in one fixed frequency, so there is where we have to choose between the common audio CD 44.1KHz or DVD, games, etc.

    I would like to know how Vista or OS X does this of resampling and mixing audio.

  6. #46
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    Quote Originally Posted by KDesk View Post
    Yes, you are right, that's a problem, it is fixed. But to mix the sound, it has to be in one fixed frequency, so there is where we have to choose between the common audio CD 44.1KHz or DVD, games, etc.
    When there's only one stream, why not play it at native rate? Then either keep the HW at the sample rate of the first stream, or switch to your favourite fixed frequency when more than one client has the sound device open. Or when more than one client has the sound dev open & not paused. But switching sample rates probably unavoidably introduces audible glitches and latency (HW HW dependent), so you want to avoid doing it too frequently. Maybe keep resampling for a few seconds after dropping back to 1 stream, in case another second stream appears right away.


    Oh yeah, that's the other downside to dmix: It doesn't support pause, so when you single-step frame-by-frame in mplayer, the audio gets ahead of the video and the video goes in a big burst when you unpause. mplayer could be smarter about this, but it isn't, and only works well with -ao alsa when HW pause is supported.

    dmix could be a lot better with more special-casing for one stream. That would solve both of those complaints (for me, since I don't run any crappy desktop stuff that keeps a sound device open for your whole session. Nor any of the stupid audio players like rhythymbox that only has a pause button, not a stop button. let go of my audio device, you bastard...) That would make dmix a lot less sucky for something that's enabled in the default config.

    SURVEY QUESTION: Do you usually only have a single client for your dmix pcm when you're not actually listening to two things at once? e.g. just watching a movie or listening to some music. Try sudo lsof /dev/snd/* (yes, you need root, because e.g. pulseaudio will have permissions you don't, so you can't access its /proc/pid/fd) If you see more than one process on /dev/snd/pcmC0D0p, you have more than one thing playing on device 0 of card 0, paused or otherwise.

    Does anyone know how pulseaudio compares on any of this? I usually killall pulseaudio, but I haven't gotten around to weeding it out of the default X startup script, and it's part of Ubuntu's default install. I should probably just remove the package, since it's easy enough to reinstall if I want it. But I guess I keep meaning to check it out and see if it's better than dmix. So, is it?

  7. #47
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    Actually, considering that recording took place in room next to one where Keith Packard had talk, quality is not all that bad.

  8. #48
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    Quote Originally Posted by KDesk View Post
    To resample from 44.1KHz to 48KHz is bad for the quality and the cpu usage.
    Quality depends on the resampling algorithm. I have 7 choices here: Fast, low, medium, high, high+, production and "none" which probably lets the sound card handle it if it can do hardware mixing/resampling.

    Neither of them produce any significant CPU load (not even 1%; I'm on Intel Core 2 E6600.)

    Btw, most cards only support 48kHz, so even if you don't resample in software, the card itself will still do it and probably with an inferior algorithm then what ALSA or OSS4 offer, so in these cases it might be better to rather resample to 48kHz to avoid hardware resampling. Of course if you're sure your card doesn't resample, then your concern is valid.

    I haven't used OSS 4.x so I can't know if it is complicated to configure or not, but to install and configure OSS without having problems... to many problems... like with pulseaudio.
    Yeah, OSS4 will probably not solve everyone's problems. I just mentioned it because it happened to solve mine and I'm very happy with it, so YMMV. Probably for you ALSA is a better option.

  9. #49
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    Quote Originally Posted by RealNC View Post
    Quality depends on the resampling algorithm. I have 7 choices here: Fast, low, medium, high, high+, production and "none" which probably lets the sound card handle it if it can do hardware mixing/resampling.

    Neither of them produce any significant CPU load (not even 1%; I'm on Intel Core 2 E6600.)
    That's pretty efficient. I thought good-quality resampling took more CPU than that, but maybe not with well-written asm... One of the dmix links someone posted had someone mentioning that ALSA's dmix took 30% of the CPU on his Atom netbook. So yeah, Atom is where this matters for more than audio quality.

    Btw, most cards only support 48kHz, so even if you don't resample in software, the card itself will still do it and probably with an inferior algorithm then what ALSA or OSS4 offer, so in these cases it might be better to rather resample to 48kHz to avoid hardware resampling. Of course if you're sure your card doesn't resample, then your concern is valid.
    Do you mean most add-in sound cards, like emu10k and descendents from Creative Labs? They support multiple streams in hardware, so don't software mixing either. I thought most AC97 and HDA onboard sound cards supported 44.1KHz. My DG965WH mobo uses a nice Sigmatel STAC9271D codec (http://www.idt.com/?genID=STAC9271&s...cPart_STAC9271, http://www.idt.com/products/getDoc.cfm?docID=18451989) for it's HDA, which supports 44.1, 48, 88.2, 96, 176.4, and 192kHz sample rates, at 16, 20, or 24 bit integers per sample (no hw float support). They claim a 105dB SNR for the DAC, and 95dB for the ADC. That's just for the audio chip, not the whole mobo, but I never hear any hiss when I crank up my speakers and play digital silence. Are other HDA codecs typically worse? I've never been a fan of Realtek's cheap-ass ethernet designs, which burden the CPU, but are their audio codecs actually ok?

    Anyway, isn't it mostly just add-in gamer-targetted cards that are 48kHz-only?

  10. #50
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    Maybe. I've used some cards from Creative over the years ("Live" and "Audigy" models) and OSS always reports their "native" sample rates starting at 48kHz. The only case I've seen 44.1kHz in the list was with an Intel HDA on-board, no idea which chip exactly it used. But an older AC97 on-board codec I had (some RealTek chip I think) was also only able to use 48k and up. So I guess it's not only add-in cards that resample 44.1->48.

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