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Old 12-31-2007, 11:42 AM   #1
lin_myworld
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Post Sound quality in linux


Hi friends,

just dont know where to post it...

One of my friends said that

Quote:
]he is not satisfied with the quality of sound in Linux(fedora7).

All of u try this : play the same song simultaneously on a linux &
windows machine, u will notice difference in sound quality.

While looking at the sound settings, he found that Fedora uses generic
ALSA sound drivers. He think that the quality is not good because the
drivers are generic, they are not specific.
I want your opinions please.

Last edited by lin_myworld; 12-31-2007 at 12:58 PM.
 
Old 12-31-2007, 12:41 PM   #2
Acron_0248
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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
 
Old 12-31-2007, 01:10 PM   #3
H_TeXMeX_H
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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").
 
Old 12-31-2007, 01:38 PM   #4
matthewg42
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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!
 
Old 05-29-2008, 07:52 AM   #5
Ry12
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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.

Last edited by Ry12; 05-29-2008 at 07:53 AM.
 
Old 05-29-2008, 10:24 AM   #6
jiml8
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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.
 
Old 05-29-2008, 10:47 AM   #7
dv502
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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.

Last edited by dv502; 05-29-2008 at 11:33 AM.
 
Old 05-29-2008, 11:05 AM   #8
Emerson
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What generic sound drivers? There is no generic sound driver in ALSA.
Quote:
All of u try this : play the same song simultaneously on a linux &
windows machine, u will notice difference in sound quality.
How often we see this.
---
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.
 
Old 05-29-2008, 01:50 PM   #9
matthewg42
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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.
 
Old 10-11-2010, 11:02 AM   #10
matt_thumper
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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

Last edited by matt_thumper; 10-11-2010 at 11:05 AM.
 
Old 10-11-2010, 03:13 PM   #11
jf.argentino
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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)...

Last edited by jf.argentino; 10-11-2010 at 03:21 PM. Reason: some precisions and spelling fix
 
Old 10-12-2010, 06:45 AM   #12
matt_thumper
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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
 
Old 10-12-2010, 07:14 AM   #13
jf.argentino
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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.
 
Old 11-04-2010, 11:42 PM   #14
medeirosdez
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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.
 
Old 11-05-2010, 12:14 AM   #15
tommcd
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Quote:
Originally Posted by medeirosdez View Post
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. ...
That could all be implemented on Linux for sure IF there was any sort of interest from Creative's Linux driver people.
Well, that is of course the whole issue. Most hardware manufacturers put all their efforts into developing drivers for Windows. Linux, if it is even considered at all, is an afterthought at best. The companies that fully support linux and open source their drivers (for example Ralink wireless cards) have hardware that works just as well in linux as it does in Windows.

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!

Last edited by tommcd; 11-05-2010 at 12:21 AM.
 
  


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