What features/changes would you like to see in future Slackware?
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Based on the frequency of stable kernel updates lately it would certainly be quite a lot of extra work for Pat to build (let alone test) each stable kernel update. I manage to do that myself but it does seem endless at times...
Based on the frequency of stable kernel updates lately it would certainly be quite a lot of extra work for Pat to build (let alone test) each stable kernel update. I manage to do that myself but it does seem endless at times...
The frequency of Ben's updates to 3.2 haven't been too onerous, but I agree: Greg seems to have been pushing one out every 3 days or so, of late, and that's getting a bit silly.
Distribution: Slackware64-current with "True Multilib" and KDE4Town.
Posts: 9,125
Rep:
Quote:
...Also install the alsa-plugins package if you want high quality resampling, upmixing/downmixing and other advanced features....
...High quality resampling
When software mixing is enabled, ALSA is forced to resample everything to the same frequency (48 kHz by default when supported). By default, it will try to use the speexrate converter to do so, and fallback to low-quality linear interpolation if it is not available[3]. Thus, if you are getting poor sound quality due to bad resampling, the problem can be solved by simply installing alsa-plugins.
For even higher quality resampling, you can change the default rate converter to speexrate_medium or speexrate_best. Both perform well enough that in practice it does not matter which one you choose, so using the best converter is usually not worth the extra CPU cycles it requires.
To change the default converter place the following contents in your ~/.asoundrc or /etc/asound.conf:
/etc/asound.conf
defaults.pcm.rate_converter "speexrate_medium"
Note: It is also possible to use libsamplerate converters, which are only about half as fast as the speexrate converters but do not achieve much greater performance. See discussion.
Note: Some applications (like MPlayer and its forks) do their own resampling by default because some ALSA drivers have incorrect delay reporting when resampling is enabled (hence leading to AV desynchronization), so changing this setting will not have any effect unless you configure them to use ALSA resampling...
When software mixing is enabled, ALSA is forced to resample everything to the same frequency (48 kHz by default when supported). By default, it will try to use the speexrate converter to do so, and fallback to low-quality linear interpolation if it is not available[3]. Thus, if you are getting poor sound quality due to bad resampling, the problem can be solved by simply installing alsa-plugins.
It is good, that HD audio codecs ("Azalia") introduced in 2004 support hardware mixing. The AC97 crap which required software mixing and resampling to 48 kHz is a thing of the past now.
Just a thought for a feature, but what about an ncurses script to help setup an asoundrc per user with ALSA with at least dmix and some other preincluded options for sample rates, etc?
Just a thought for a feature, but what about an ncurses script to help setup an asoundrc per user with ALSA with at least dmix and some other preincluded options for sample rates, etc?
The default asoundrc file is /usr/share/alsa/alsa.conf, which has DMix.
As for the sample rate stuff, I find it really unlikely that it would make any real-world difference.
Quote:
Originally Posted by nitecrawler
a dependency manager could be one option...
Assuming you were serious, one of the ideas behind Slackware is that you don't need a dependency manager if you do a full install.
That said, Slackware used to include swaret (in /extra), and slapt-get is a standard part of most forks.
It is good, that HD audio codecs ("Azalia") introduced in 2004 support hardware mixing. The AC97 crap which required software mixing and resampling to 48 kHz is a thing of the past now.
Any of the onboard chips usually are still Software Mixing only. I only know of a few onboard devices that actually do(did) hardware mixing and usually those chips are the same ones found on actual sound cards such as the Sound Blaster Emu*-series, C-Media OxygenHD, some older AC97 stuff like the Aureal 8810. Just about everything that is a CODEC style chip HDA or AC97 is software mixing only.
i don't see hardware resampling noted there
maybe it's in the driver
A High Definition audio codec supports up to 15 streams with varying bit depths and sample rates (up to 192 kHz) with up to 16 channels per stream (only those channels must have the same parameters). So there is no need for software mixing nor resampling.
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