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Sound quality in linux
Hi friends,
just dont know where to post it... One of my friends said that Quote:
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Hi,
Well, I could say that might be another factors in play, codecs, encoding quality, sound daemon (esd, arts, etc...), sound card, and so on... From my own experience, I think that sound quality is kinda equal between OSes, however, I enjoy much the sound from linux that from windows when I'd use it. Normally in my distributions, I always use the alsa kernel drivers as modules and I can't complaing about the quality, the only thing that I first noted using linux was that I could have higher volume by configuring alsamixer xD, but as 'quality' I didn't see much difference. I might be wrong in this, I'm not a sound encoder/decoder expert so maybe it's true, maybe it's not, as I said, from my own experience, I don't see much difference, in fact, I'll put alsa sound quality, regardless you're using generic drivers or not, a little on top of Windows sound system. Regards |
I think it depends on both the drivers and the codecs (and sometimes also program used). It's best to use native codecs if possible, but, this may not be possible. Some programs also seem to handle music better than others, for example mplayer, you can also choose which audio driver to use in the options. ALSA is NOT always the best driver to use. I've heard JACK is one of the better drivers to use (it's actually a "low-latency audio server").
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Firstly, please post the type of audio card you have in your machine. I believe some chipsets do have driver quality issues with Linux, and it would be helpful to try to confirm if your card is one of these. The program lshw (run as root) should help you determine this, or maybe just checking the output of dmesg shortrly after boot.
Probably you've already tried it, but just in case, make sure the mixer settings are sane... The output volume is usually decided by a combination of PCM and Master levels. If PCM is at 100%, I find on my cheapy integrated audio that I get a lot of clipping and thus horrible sound quality. I get best results with PCM at about 80-90% and then use the Master setting to change the actual volume. A too-small PCM level will lead to low output volume, and a lot of hiss. Also, if you have a Microphone input, make sure the Mic is muted, or at least that mic-input-through-to-output switch is off. You might also want to try turning off input/output mixing options. If any of these suggestions, or another mixer setting improves the sound quality, please post here what you did (in addition to your sound card type, as requested above) so others might profit from your solution. Good luck! |
i have an x-fi sound card and I can definitely notice a difference in sound quality when using it under linux. The fidelity decreases somewhat.
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"quality" is a term that is so generic that it is meaningless when used without substantial qualification.
I personally note no difference between windows and linux sound quality using the same hardware. I also have noticed that with linux driver software I can inadvertantly set up my sound output to clip, which immediately results in a noticeable degradation of sound quality. If there is a quality problem it could easily be there. The solution varies from soundcard to soundcard but generally consists of something like turning down the PCM output level and controlling the output using the master control. |
In my personal experience, I rather use dedicated hardware than the onboard ones.
I don't know about you, but I did notice a big improvement in audio quality when I switched from the onboard realtek soundcard to an PCI Sound Blaster card. |
What generic sound drivers? There is no generic sound driver in ALSA.
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--- I formatted to X filesystem, now it runs much faster than with Y! Q: How much faster, how did you measure it? A: ... [silence]. --- Hell, Z operating system loads so much faster than ... --- Now. If quality is lower then there must be distortion. Luckily enough, bot linear and non-linear distortion we encounter in audio world are well measurable. So, how this person who's quoted above measured the distortion? Show us the numbers please, or all this is old ladies talk. |
It is possible that the windows drivers are doing some audio processing - noise reduction or other filtering, or perhaps some mixer settings to provide something like a bass boost or other enhancement.
I noticed that the windows driver for my ICH4/ICH4-L/ICH4-M chipset built-in laptop soundcard does just this. It probably is reasonable to assume that such a soundcard will have crappy speakers attached and so such a feature of the driver could well be desirable in most cases. I find it possible to get a similar effect by enabling the equalizer in amarok and boosting the bass and mid a little. Perhaps someone knows of some equalizer / effects application which can sit in the background and post-process output to the audio card? I found ecamegapedal, but I'm unsure about how to get it working with anything but the mic input. |
Still Lousy Linux Sound.
Well in 2010 and Fedora 13, sound is still a problem with Fedora.
Despite all the fluff in the responses above about this being the possible cause and that being the possible cause - what also is possible is that it actually *is* a problem! This older Dell computer played wav songs just fine on its soundblaster card when it was Windows about a year ago. And it played them using the exact same Java software I am using now that it is a Fedora 13 box. Additionally, (and not to put too fine a point on it, but what the heck) I can turn to my left and use my new Dell (Windows 7) and use the exact same Java code (recompiled of course) - AND the exact same .wav files and it sounds just fine. So in summary: SoundBlaster sound card: FINE .wav files: FINE Java software: FINE Fedora sound: LOUSY Matt |
Quick response:
Trust me, Fedora 13 is OK with sound (has any distros from many years now), what is the JAVA player you're using? Have you tested another different player? Are you sure the sound api wrapper used by the JAVA application is compatible with PulseAudio (the standard way to access the sound card on FEDORA from something like three years now)?... Why I'm sure that FEDORA 13 is OK because, in normal condition, and with same software audio-processing chain, sound quality has nothing to deal with software environment! Only hardware has influence! (the long response:) Two software problems can be the main cause of sound quality degradation: -buffer overrun or underrun, for sure a buggy driver can cause these, but when it occurs, it's certainly due to a performance problem, the system load is too high for the sound app to be able to feed the output buffer in time. -overrange thus the data is clipped, or worst the overranged part is "wrapped" on the other dynamic side, could be due to a driver too since most of sound card has hardware mixer... BUT ALSA drivers (ALSA is not a generic driver, its a drivers collection with the same interface) are of really good quality, their are certainly bugs inside (as in any complex piece of code), but certainly not easily audible ones, or for really exotic hardware... To finish with sound drivers under linux, there are only two: ALSA and OSS (this last is obsolete now), JACK, PulseAudio, aRts... are sound servers, some glue between the driver and the apps. Some players can use ALSA directly, but this is less and less the case, and I'm not sure if it could introduce some sound troubles due to hardware access conflict (buffer and mixer). Still there, let's go for some more details. A sound application have to feed a buffer with sound samples at a given sampling rate, and with a given quantification. For example a sampling rate of 44.1kHz means there's 44100 values per second and per channel to "describe" the sound, and a 16bits quantification mean than each sample can take an integer value between -32768 and +32767, which represents the maximum instantaneous sound level, voltage level the sound card can push. For what I know (I'm using JACK but it could be different with PulseAudio, or with applications that drive ALSA directly), the sampling rate and the quantification is set once for the sound card, and playing a song with a different sampling rate and / or quantification need software tricks (downsampling or upsampling, dithering...). These tricks can degrade the sound quality if badly coded, but once again, the ones we can found under linux are of great quality. One more important point: until now i was talking as if the sound information is recorded samples per samples in the sound file. This is true for wav files and some other format (flac, au...), but not for lossy format like mp3 or ogg. For what I know, only the encoding part is critical for sound quality, because the point is to remove data that aren't essential to the _HUMAN_ sound perception, and a dumb encoder could remove data more heard-able than an intelligent one. But the decoding part that need to transform sound back (a sound card only understand PCM data) doesn't do any "intelligent" task, thus, if there's no major bugs, couldn't degrade quality. But you can increase quality with some post processing trick (but do not expect miracle). I'm playing with fedora until the core 1, and I can't live without music, so trust me if there were any problem with sounds under linux, I wouldn't have leaved WINDOWS definitly, and I've just tested my gift (a RME Multiface) with oscilloscop, spectrum analyser (I'm working in an electronic company) and the results are great... due to the hardware of great quality, not due to FEDORA (of great quality too)... |
Quick reply to that very Quick Response:
Trust me, Fedora 13 is OK with sound (has any distros from many years now), what is the JAVA player you're using? Have you tested another different player? Are you sure the sound api wrapper used by the JAVA application is compatible with PulseAudio (the standard way to access the sound card on FEDORA from something like three years now)?... > As indicated extremely concisely in my post, there is no need to test anything. I have three cases to extrapolate and draw conclusions from. The broken component clearly, and concisely, is Fedora. Fedora is an overwhelming example of extremely good coding (that is, in case you don't understand, software). My exposure to their talented development community leads me to believe that *they* are interested in correcting weak points in that fantastic software - and even bugs when they are found. *They* are, even if *you* prefer not to hear about them. Continue on with your blinders, by all means. Why I'm sure that FEDORA 13 is OK because, in normal condition, and with same software audio-processing chain, sound quality has nothing to deal with software environment! Only hardware has influence! (the long response > That's so funny! re-read my post you silly thing you. The hardware has NOT changed. You silly thing you. Two software problems can be the main cause of sound quality degradation: -buffer overrun or underrun, for sure a buggy driver can cause these, but when it occurs, it's certainly due to a performance problem, the system load is too high for the sound app to be able to feed the output buffer in time. -overrange thus the data is clipped, or worst the overranged part is "wrapped" on the other dynamic side, could be due to a driver too since most of sound card has hardware mixer... > Well that's ...uh, well that's *not* interesting. The software is just fine, thank you very much. I have run it for years and years and years on several platforms. By the way, RedHat was one of those platforms. But you go on thinking it's the software. ....blinders.... BUT ALSA drivers (ALSA is not a generic driver, its a drivers collection with the same interface) are of really good quality, their are certainly bugs inside (as in any complex piece of code), but certainly not easily audible ones, or for really exotic hardware... To finish with sound drivers under linux, there are only two: ALSA and OSS (this last is obsolete now), JACK, PulseAudio, aRts... are sound servers, some glue between the driver and the apps. Some players can use ALSA directly, but this is less and less the case, and I'm not sure if it could introduce some sound troubles due to hardware access conflict (buffer and mixer). > Well, Fedora's drivers might be the problem. Windows drivers aren't. RedHat drivers aren't; but Fedora's drivers (which, by the way, are SOFTWARE) might be a problem. Still there, let's go for some more details. A sound application have to feed a buffer with sound samples at a given sampling rate, and with a given quantification. For example a sampling rate of 44.1kHz means there's 44100 values per second and per channel to "describe" the sound, and a 16bits quantification mean than each sample can take an integer value between -32768 and +32767, which represents the maximum instantaneous sound level, voltage level the sound card can push. For what I know (I'm using JACK but it could be different with PulseAudio, or with applications that drive ALSA directly), the sampling rate and the quantification is set once for the sound card, and playing a song with a different sampling rate and / or quantification need software tricks (downsampling or upsampling, dithering...). These tricks can degrade the sound quality if badly coded, but once again, the ones we can found under linux are of great quality. > I don't care to even understand the meaning of all that hob-glob. Now, I've got *my* blinders on. One more important point: until now i was talking as if the sound information is recorded samples per samples in the sound file. This is true for wav files and some other format (flac, au...), but not for lossy format like mp3 or ogg. For what I know, only the encoding part is critical for sound quality, because the point is to remove data that aren't essential to the _HUMAN_ sound perception, and a dumb encoder could remove data more heard-able than an intelligent one. But the decoding part that need to transform sound back (a sound card only understand PCM data) doesn't do any "intelligent" task, thus, if there's no major bugs, couldn't degrade quality. But you can increase quality with some post processing trick (but do not expect miracle). I'm playing with fedora until the core 1, and I can't live without music, so trust me if there were any problem with sounds under linux, I wouldn't have leaved WINDOWS definitly, and I've just tested my gift (a RME Multiface) with oscilloscop, spectrum analyser (I'm working in an electronic company) and the results are great... due to the hardware of great quality, not due to FEDORA (of great quality too)... > Ditto my last remark. My goodness, if you spent as much time working with the Fedora community on correcting their issues as you do adamantly decrying that they have no issues - my goodness - it would be an *even* better product than it is now. That's OK, I continue to like Fedora better than other Linux distros that I've used. I continue to like Fedora better than Windoze (yes, that's really true!). I just don't like the Fedora sound quality on my Dell SoundBlaster card - which is lousy. Matt |
I don't like how sounds your post. OK I'm blind (deaf is more appropriate there) and whatever other adjective you prefer like silly or dumb...
My post wasn't aggressive, its purpose was to explain to you why your conclusion is false... Try another player found on fedora repository (audacious xmms...), if there's still a problem try to figure from where it can come from. As a conclusion, sound under fedora works great for most people out of the box, but I won't waste my time with aggressive people for which it doesn't. |
Let's keep it simple
I don't know precisely what problem you guys are talking about, anyway, I want to leave here the comment of mine on this subject.
I own a Creative SoundBlaster X-Fi XtremeMusic. On Windows, Creative's drivers offer you technologies fully implemented in hardware like Crystalizer and CMSS-3D. Crystalizer stands for a really dynamic equalizer which adjusts itself according to the quality of the sound the card is fed with. For example: if I'm playing a 96kbps MP3 file, with considerable loss of high frequencies, the card is able to play with its own equalizer, the one specifically called Crystalizer, so you can kind of restore the lost frequencies (that's the illusion you have). CMSS-3D stands for a multi purpose surround sound mixer. For example: if you have a 5.1 home theater speakers setup and want to play a stereo sound, the CMSS-3D technology is able to identify instruments on the sound the card is fed with and dynamically distribute it through all the available speakers. It's a totally active design, that's why I say "dynamically". If on the other hand, all you have is a pair of speakers or headphones, the card is able to mix into stereo channels up to 8 surround sound channels with Creative's proprietary HRTF technology. It immerses you into your DVD / BD movies or 3D games. That could all be implemented on Linux for sure IF there was any sort of interest from Creative's Linux driver people. Anyone knows for sure it'll never happen anymore, thus, we're left with a complete 2D and raw sound experience with Linux or any other OS except Windows with Creative's sound cards. I often test new Linux distributions but can never stay with anyone of them because of the lack of such important technologies. I have 3 headsets, high fidelity ones (12 to 28 KHz frequency range and near 20 bit sound quality), and I very frequently watch 5.1 CH videos and play 5.1 CH games. I just couldn't live without CMSS-3D. Besides that, X-Fi Crystalizer is just so amazing I have no pleasure on listening to anything without it. Linux, OS X and Windows do have almost no difference in sound quality. Up to XP, with such a good card as the X-Fi, Windows was the best out of the three. Having them all properly set up not to make any sort of resamplings, they're all the same. Each, of course, deal with latency a different way. I'm not quite familiar with other OS'es but at least Windows has WASAPI and works admirably with ASIO 2.0 and OpenAL (hardware accelerated, opposed to software rendered on Linux and OS X). These APIs are quite good on latency, specially ASIO. Anyway, without resampling and not taking into consideration latency, any OS should be the same. Now, when it comes to improvements implemented in hardware / driver, no doubt Windows wins (on Creative's side, specifically). I'll just keep using Linux once in a while with its 100% software rendered, mixed, resampled audio, quite flat and 2 dimensional. For the best quality and user experience and full use of my hardware capabilities, I'll unfortunately have to stay with Windows. By the way, have any of you ever realized that Linux just doesn't grow on the "hardware accelerated" industry? The most it gets is OpenGL. There are no solutions to fully decode in hardware video streams, like MPEG2 or H.264, or to mix audio streams from different sources, or to process MIDI, or EAX effects, or audio resempling, downmixing, or TCP offload, or whatever. Everything, except OpenGL is software implemented. I shame indeed! Sorry for the bad English, I'm a self taught English student, from Brazil. May G-d's peace be with you who seek for it. |
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For those who swear that sound is "better" in Windows, perhaps try a blind comparison. Install Windows and linux on the same computer. Have a friend boot either Windows or linux randomly and play some music. See if you can consistently tell whether Windows or linux is playing the music. You may be surprised at what you find. Even the (supposedly golden eared) high end audio reviewers at Stereophile magazine dare not do blind comparisons of audio hardware. Medeirosdez, Welcome to the LQ forums! |
Res to Tommcd
Hello there Tommcd! I just wanted to thank you for your kindly welcome!
I totally agree with you on what you've said. There is no difference. To make clear once and for all, people should keep in mind we're talking about **digital signals** here, processed in a lossless (PCM) manner. If there are no resamplings or any other effects applied to the PCM stream, any OS is pretty much the same! It works just like this: *You open the application that will play your songs. *The application uses the appropriate demuxer (to deal with .mp3, .wma, .ogg, .m4a... extensions) to get the encoded stream (wma, mp3, aac, flac, ogg...). *Then there is a decoder that will convert back the encoded stream into a PCM stream [the application could, then, make use of some DSPs to resample or up / downmix by itself]. *The PCM stream, most likely to be untouched by the application, is sent to your sound system. Now, when it comes to the PCM stream and the sound system, things are just like this: *If you have a sound server, like Pulse Audio, the PCM stream is sent to the Pulse server. Pulse up / downsamples the PCM stream if it is different of its own settings (you can change Pulse default sampling rate by editing the "/etc/pulse/daemon.conf" file, it also up / downmixes the PCM file if its told to, *and, at last, the processed (or maybe untouched) PCM stream is sent to ALSA system. I'm not going to get into deeper details on how things work under Windows or OS X, for example, because things are almost the same with all of them. The point is: the **true** maximum audio quality is achieved but getting the PCM stream untouched to the sound card. That ANY operating system can do, including Linux. ALSA is able to do the resampling stuff on its own, but it's not as configurable as Pulse Audio. I myself like to set Pulse to 44,1 KHz sampling and S32_LE resolution all the time, except when I have to deal with 48 KHz audio samplings. I know for sure my sound card works only at 48, 96 or 192 KHz on Linux, but its audio processor (X-Fi) has an excellent resampler using advanced techniques. So, I prefer to always let the sound card to the processing. If you don't have such hardware capabilities, I guess you'd really want to consider taking a look at this website. Up and downmixing on Linux, anyway, is flat and 2 dimensional as I've said. I've already tried ALSA "vdownmix" plugin, it works but it's not yet as well implemented as Creative's own HRTF technology. Amarok has a built in equalizer that does its job, specially when set to "Rock" if I'm not wrong. aTunes also has its own equalizer but it sounds horrible and is not worth a try. So, quality is the same on every OS if you know how to set them up. Here I've left the tips of mine to help anyone who would occasionally be interested. > Preferably, don't do any resamplings. > Tell Pulse and / or ALSA to always use the highest definition for your sound card (mine is 24 bit, so I use the S32_LE instruction both under Pulse and ALSA). > Avoid multichannel if all you have is a pair of speakers or headphones, or try the not so well implemented "vdownmix" plugin for ALSA. > Try Amarok equalizer (probabily set to "Rock") to help improve a bit the perceived sound quality. Any questions, I'm fully available at "medeirosdez@yahoo.com.br". Best wishes to everyone here and to Linux in the hope that it continues to improve and to be a more and more appealing alternative to MS Windows or Apple OS X. |
To medeirosdez,
Hi there! Here's my prob. The songs(.mp3) and the music videos (mpeg4) have similar (good) audibility and quality when played on both Linux(Ubuntu 10.10) and Win XP,but when i compare the sound in DVD movies( DivX) in the two OSes, there is a distinct difference,louder in Win. and low in Linux! For the movies( tested 4-5 movies) i have MPlayer and GnomeM player and the sound output is low in both. In Win. i have used the Win.Media Player and VLC. I have Realtek HD Audio Drivers installed. :confused: Can you kindly explain and show a solution to this odd difference in sound volume between Music Videos and Movies? Consider me a semi-newbie to linux! :scratch: warm regards, vyver. |
Hey vyver!
That's nice you're counting on me to help you. I'm pretty sure I have the right answer to you! The "problem" you talk about exists on Windows too. We're actually talking about a Dolby Digital issue. It's not exactly a problem though, nothing that has got to be fixed. Dolby Digital has a lower audio range, output, volume, whatever they call it, meaning it sounds lower than other formats, like DTS (which happens to be better concerning audio quality). Windows 7, for example, has a built-in Dolby Digital decoder. If you right-click the video screen (on Media Player) and go to "Enhancements" (I'm not sure that's what it's like on the English version of the software since I use the Br-Portuguese version) you can click "Dolby Digital settings" and it will give you three options for you to choose: "Normal", "Night" and "Cinema". If you choose "Normal" you're going to have a boost on low volumes and a decrease on the excessively high ones. If you choose "Night", you'll have the same softened output but a slight increase on the dialogs (I guess this one sounds better for onboard sound solutions). Now, if you choose "Cinema" mode, then you'll have the same exact output that the DVD was intended to have, called full dynamic. At times the sound will be low, and at times high, it'll vary dynamically. Most people won't like it if they don't have a good sound card, capable of high dynamics in audio, and a good speaker system or headphones too. I don't know which Dolby Digital decoder you've chosen under Windows XP, since it doesn't have a built-in one, I'd recommend, by the way, AC3Filter, it's one of the bests out there. Anyway, it seems your decoder under Windows is set to something like the "Night" mode, giving you pretty much the same (high enough) volume experience throughout the whole video. Linux, on the other hand, doesn't have the same configurability of Windows 7 and other decoders like AC3Filter, and instead of choosing the "Night" mode equivalent as does your Windows XP decoder, it chooses the "Cinema" mode for quality purposes. As I've said, preferably, the encoded, compressed sound should be decoded and sent to your audio system **untouched**. If you have your system sound output set to stereo there is surely some processing done with the decoded DVD audio in order to fit it to 2CH outputs. Anyway, the Linux decoder tries to convert the compressed audio stream to PCM **as it is**, with the same output volume as is in the DVD. As said, too, Dolby Digital has lower volume if compared to other compression formats, like DTS, MP3, etc.. Thus, you have the impression that Linux sounds worse, which isn't true. Actually, it sounds better since it doesn't play with audio volume. I don't know any solution to this but trying to increase both the PCM and Master volumes under ALSA Mixer (type on terminal "alsamix"). I myself can't stand watching DVD's or any multichannel audio under Linux. Firstly because it has no advanced audio downmixing with well implemented HRTF techniques, as Creative's CMSS-3D. Secondly, it has no hardware accelerated MPEG-2 decoding. There are good [paid] choices for Windows out there, but for Linux the best you're gonna have is VLC which has plenty of choices for deinterlacing. Anyway, none of them have come even near to the quality of my hardware dedicated decoding under Windows. My ATI Radeon HD 3850 deinterlaces using vector adaptive algorithm with "pulldown" detection, that makes my DVD's look gorgeous! Let alone if I had a better card, like the 4000 series, with upscaling techniques!!! I like most Linux Mint and Ubuntu. I have made extensive tests with Mint 10 RC and Ubuntu 10.10. I can say that Totem DVD player, or Gnome Mplayer look just like Apple's DVD player on Mac OS X. Sound for me is not an issue on DVD's because of my better sound card. I hope to have helped make things clearer to you! Feel free to ask anything else if you need, I'll be keeping up with this.. thread here. Best wishes! Paulo Henrique, from Brazil!! |
I have Realtek HD drivers as well. I also have a decent set of speakers (Creative) to blast out whatever I am doing with sound. My son-in-law is an avid gamer and he lusts after my speakers. The point is, there is more to sound than just a driver.
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Surely mlangdn, there's more to sound than just a driver. But what if all you have are headphones? See what the problem is? You've got to have good technology to help you make the best out of the thing you have. That's the point of my concern with HRTF techniques.
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Yes - but that decent set of speakers cost less than $30 US. Cheap doesn't mean bad. That said, it may not be concert-hall like either. It really depends upon your own needs. It impresses the heck out of my grandkids when they play their games.
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I think I get what you mean! Totally agreed!
Anyway, speakers normally don't support high frequency range. I've been looking at the specs of a few systems and they rarely go beyond 20KHz maximum. CD audio exceeds that limit going up to 22,050KHz. For instance, I have a Tool (band) HDCD (44,1KHz sampling / 20 bit) and I was taking a look at the frequency spectrum. There are parts of some songs that reach something like 21,5 KHz. That is surely lost on speaker systems. I know there's a whole discussion on whether there's any difference on supporting frequencies beyond human threshold, and I'm not putting that into question. Thing is: the highest the frequency range, the better. It's impracticable to me to have speakers where I live. The house is not mine and people here would put me out if I started listening to anything out loud. That restricts me to headphones, which is not an issue to me, at all! I have a Phillips one with a frequency range of 15 to 28KHz, and a 106dB sensitivity. My sound card has a 109dB SNR, which makes it almost pair with my headphone's sensitivity. CD's, at 16 bit, can go up to 96dB, so, I'm doing even better here towards 24 bit (the best cards out there can barely make it to 20 bit anyway). What I mean is, headphones are a very attracting alternative to speakers! Especially if you're dealing with high quality audio, like DVD / BlueRay ones! The point is: how to get the best surround sound experience with 2 channel headphones? Technology comes in here! Let me a bit out of my own scenario here and take a look at mlangdn's: they've got speakers out there which seem to give them a very good experience with sound. If their set of speakers are two channel only, they'd certainly need the same technology that a headphones user needs. Now, what if they have multichannel speakers, like 5.1? What if all they had at times was a two channel sound to play on that setting? Technology comes in too! A card capable of actively, intelligently, dynamically distributing sounds through the available speakers would be awesome! It'd be an amazing virtual surround sound from a stereo source, quite different of just copying the left channel to all left side speakers and right channel to all right side speakers. The subject of this thread is "sound sounds bad on Linux". In fact, good sound card paired with good speakers is quite enough for majority of people, but even those who confidently defend they don't need any improvement would be astonished if they were introduced to what technology can make to take the PC user experience to a whole new level of perceived quality and entertainment. mlangdn, would you mind telling me what platform is used when you talk about gaming? Is it Windows or is it Linux. If Linux, I'm anxious to know what games specifically your grandkids play, that's very interesting! :-) I hope you now understand better my point on defending technology in parallel with a good sound card and good speakers or headphones. It's pretty clear to me none of them live well without one another. Best wishes! |
Medeirosdez,
Do you know why pulse-audio in Ubuntu uses so much more CPU resources than alsa? And if so, do you know how to make pulse-audio use less CPU resources? There are a great many people complaining about this on the ubuntuforums.org. I have always removed pusle-audio because of this. I don't notice any difference in sound with or without pulse-audio, but I have never tried to tweak pulse-audio either. |
As far as I'm concerned this squabbling over sound cards is no more than another "Linux vs Windoze"thread and we all know where this ends up!but however earlier someone stated to try the music on THE SAME HARDWARE in a blind test and I doubt very much if you could EVER tell the difference!!!
QUOTE: For those who swear that sound is "better" in Windows, perhaps try a blind comparison. Install Windows and linux on the same computer. Have a friend boot either Windows or linux randomly and play some music. See if you can consistently tell whether Windows or linux is playing the music. You may be surprised at what you find. Even the (supposedly golden eared) high end audio reviewers at Stereophile magazine dare not do blind comparisons of audio hardware.END QUOTE! |
tommcd,
I guess I know a way to help you, but I'm not sure this is going to solve your problem. At this link you can find an easy way to customize Pulse settings. An specific line on the daemon.conf file may dramatically increase your CPU usage in certain scenarios. Ubuntu and other distributions based on it set Pulse to work at the CD default 44,1 KHz. Any audio at a different sampling rate will be resampled to 44,1 KHz. That's not exactly what you'd want unless your hardware is stuck at 48KHz sampling and is incapable of decent high quality resampling on its own. The line I was talking about on daemon.conf file is "resample-method=". My advice is to set it to "speex-float-X", where "X" is a number varying from 1 to 10, default being 3. The highest the number, the better the resampling at the cost of more processing power needed. I had it set to 10 just because I want the highest quality possible, but that would usually eat up 25% of my CPU when resampling stuff from 48KHz to 44,1 or vice-versa. For reference, I have a Core 2 Extreme X6800 @ 2,93GHz. You can have it to its default 3 and it'll probabily be same if not a bit better than Windows 7 resampling algorithm. It's a real important thing to understand that the best you can do both for performance and quality is accordingly change the default sampling rate on that file everytime you deal with stuff sampled at a different rate. That makes Pulse use about 1 to 2% CPU when I'm listening to my songs. There is an issue, anyway, that is **not** related to how much CPU power Pulse needs. If I have a fairly heavy page loaded on my browser and start scrolling it up and down while listening anything through Pulse, or if I make any heavy use of my CPU, I have a lot of pops and distortions on the sound. That things disappoints me quite a lot and I could find no solution for it anywhere. Instead of uninstalling Pulse, I'd recommend configuring ALSA and making your applications access it rather then Pulse. That was simple and as soon as I configured ALSA, Flash audio was automatically direct to it rather than Pulse. A great deal which fixes the pops and distortions issue with Pulse. And the best of it: I can keep Pulse since without it Ubuntu is just loses its system related audio functions. Create a file called asound.conf at /etc/ and put this on it: pcm.!default { type hw card 0 format S16_LE rate 44100 } ctl.!default { type hw card 0 } Change S16_LE to the highest your sound card can deal with. Mine is 24 bit capable, S24_LE is not available (I forgot the command to list the available formats), but S32_LE is, then I use S32_LE rather than S16_LE. The "rate" I choose varies according to the sound I'm dealing with. It's important to tell ALSA what rate to use because it will **always** default to 48KHz whenever DSP is needed. After saving the file, restart ALSA: sudo /sbin/alsa-utils stop It will automatically restart (it has with me). If not do: sudo /sbin/alsa-utils start After changing /etc/pulse/daemon.conf you also have to restart Pulse: sudo killall pulse $ pulse If I remember correctly, the command pulse as normal user, starts up Pulse after it's been killed. I hope to have helped once more. |
tiredofbilkyyaforallican,
I'd recommend that you don't say this again. I'm *sooo* excited I'm being able to talk to someone about what I have experienced with Linux and Windows! This is **not really** a which is better. I'm afraid people start saying that whenever someone says Linux does not an specific thing that Windows does just fine. I like Linux and I think it's a *great* OS, especially if you come to think it costs you nothing and has comparable speeds to Windows, which costs *so much*. One would expect Windows to be way better because of its price, but Linux has managed to be compared to Windows and be found almost if not the same. Let's please don't take this subject any longer. It's going to be a way to keep peace in here, ok? Anyway, I've said plenty of times: there is *no* difference between *any* OS, except when you start talking about latency, resampling, up and downmixing, etc.. Those are features that could be perfectly implemented on Linux *if* there was any sort of interest from the industry. There isn't, anyway. There is no interest not even from Linux developers around the globe as I have seen. They'd do a great job, buuuut... ALSA has got the thing in their hands with their plugins, like dmix and vdownmix. There has got to be development, just that. And, one thing has got to be said: people are not worried about audio on computers. Anything is going to be OK for them if it can hardly be differentiated from one thing or the other. Our people and our modern industry is mostly concerned on **graphics**, that's it. |
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Dear Paulo Henrique,
I have installed the alsamixer gui and just upped the volumes to the max. there(window of it) and voila, the movies sound is an absolute blast with no compromise in quality! Thanks and thanks again! Planning to compete with BOSE? warm regards, vyver. |
Medeirosdez,
Thanks a lot for all the helpful info. I managed to learn a few things from following this thread. |
Thank you vyver, thank you tommcd for your kind support. I'm glad my small knowledge helped you guys out there!
Sound is not really a concern for people but I think it should be. Anything you need, I'm available at "medeirosdez@yahoo.com.br". May G-d's peace be with you who seek for it! Best wishes! |
Try different Media Players.
I am not disappointed with VLC 1.0.4. The skin does not come with it so you have to add that as well. If you go to open it with out the skin you will be looking at nothing.I think VLC is One of the top applications period.
Linux users are not into high fidelity like a windows user.A coworker of mine has a twenty thousand dollar computer that he can't afford to insure for the full amount. He is a Media addict. I only have XP pro for a windows box but VLC has a better sound quality than media player easily on either system. This is just driving speakers that are built into a monitor. I intuitively don't believe sound quality is a decoder/codec issue or drivers. From my own background I believe that the programs are mimicking hardware with software which is not the same as having a quality amplifier circuit. It's only worth having a certain quality level on a computer before going to a real home system with a power amp and so on.If something is built for a specific purpose it does a better job. I assume that there are amps that you can connect to a PC but have never looked. You should probably try a few Media Players like Banshee to see which one is the best. Media Players have a lot of dependencies so if your Package Manager does not have it,it will be a HUGE pain to install it from RPMs. RPMs are in single files so you might need sixty of them or so. |
I think the quality under xine might be better than xmms.It's sometimes a little difficult to distinguish crappy decoding from crappy encoding. A significant factor might be that xine uses the ALSA interface directly, instead of going through the OSS emulation interface.
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I think gstreamer is bit more popular nowadays, or is becoming more popular at least (because of Ubuntu), than xine, don't you? And it's decoding seems excellent to me.
theKbStockpiler, I think VLC is the best video player for Linux out there. I'm yet to find any that has more features than that. |
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regards, vyver. |
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