Re: defaults.pcm.rate_converter not respected?

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Dne Čt 3. ledna 21:50:32, David Kačerek napsal(a):
> Hi,
> I've got Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)
> sound card which support most of sample rates used in my music library so I
> have set my sw players to send audio streams straight to the "hw:0,0"
> device (or to one of virtual devices containing chain of ladspa plugins
> I've added to asound.conf) to prevent pointless and counterproductive
> resampling of dmix or pulseaudio. Only one of my albums is recorded at a
> weird frequency: 88200Hz which has to be resampled to the nearest 96000Hz
> to be able to be accepted by my sound card. Allegedly Alsa uses poor
> resampler by default so I've put "defaults.pcm.rate_converter
> "speexrate_best"" line to asound.conf. Now If I play a 882000Hz track in
> mpd I can see in a performance monitor it uses significantly more CPU when
> "speexrate_best" is set in comparison with the bare "speexrate" so I guess
> mpd respects my "defaults.pcm.rate_converter" value. But when played the
> same track in mplayer the CPU load is the same regardless of the
> "defaults.pcm.rate_converter" value used. Moreover when mplayer is started
> with "-af-adv force=3" parameter to disable internal resampling playing of
> such a track is stopped with an error even though "alsa" or
> "alsa:device=hw=0.0" is set as the audio output. Could it be a wrong setup
> of asound.conf? Maybe I need to specify a specific sample rate to be
> streams converted to but I prefer leave most of sampling frequencies being
> intact (since the sound card supports them) and resample only the
> problematic one which is 88200Hz - how do I do that in asound.conf? Or Is
> it a bug in mplayer? My asound.conf:
> pcm.!default {
> type hw
> card 0
> }
> ctl.!default {
> type hw
> card 0
> }
> 
> defaults.pcm.rate_converter "speexrate_best"
> 
> pcm.loudspeakers {
>     type plug
>     slave.pcm "ladcomp_compressor";
>     hint {
>         show on
>         description "Loudspeakers"
>     }
> }
> 
> pcm.ladcomp_compressor {
>     type ladspa
>     slave.pcm "ladcomp_limiter";
>     path "/usr/lib/ladspa";
>     plugins [{
>         label dysonCompress
>         input {
>             controls [ -30 0.25 0.5 0.5 ]
>         }
>     }]
> }
> 
> pcm.ladcomp_limiter {
>     type ladspa
>     slave.pcm "plughw:0,0";
>     path "/usr/lib/ladspa";
>     plugins [{
>         label fastLookaheadLimiter
>         input {
>             controls [ 0 0 0.01 ]
>         }
>     }]
> }
> 
> pcm.earspeakers {
>     type plug
>     slave.pcm "ladcomp_binaural";
>     hint {
>         show on
>         description "Earspeakers"
>     }
> }
> 
> pcm.ladcomp_binaural {
>     type ladspa
>     slave.pcm "ladcomp_limiter2";
>     path "/usr/lib/ladspa";
>     plugins [{
>         label bs2b
>         input {
>             #1) 700 Hz, 4.5 dB - default.
>             #   This setting is closest to the virtual speaker placement
> with azimuth 30 degrees and the removal of about 3 meters, while listening
> by headphones.
>             #2) 700 Hz, 6 dB - most popular.
>             #   This setting is close to the parameters of a Chu Moy's [3]
> crossfeeder.
>             #3) 650 Hz, 9.5 dB - making the smallest changes in the original
> signal only for relaxing listening by headphones.
>             #   This setting is close to the parameters of a crossfeeder
> implemented in Jan Meier's CORDA amplifiers.
>             controls [ 700 10 ]
>         }
>     }]
> }
> 
> pcm.ladcomp_limiter2 {
>     type ladspa
>     slave.pcm "plughw:0,0";
>     path "/usr/lib/ladspa";
>     plugins [{
>         label fastLookaheadLimiter
>         input {
>             controls [ 0 0 0.01 ]
>         }
>     }]
> }
> 
> OS: Arch Linux x64
> 
> $ aplay -l
> **** List of PLAYBACK Hardware Devices ****
> card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]
>   Subdevices: 0/1
>   Subdevice #0: subdevice #0
> card 0: Intel [HDA Intel], device 1: Conexant Digital [Conexant Digital]
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0
> 
> MPlayer error:
> [AO_ALSA] 0 channels are not supported.
> [AO_ALSA] Format ?? is not supported by hardware, trying default.
> AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample)
> ID_AUDIO_CODEC=ffflac
> [Mixer] No hardware mixing, inserting volume filter.
> [libaf] Automatic filter insertion disabled but formats do not match. Giving
> up.
> Video: no video
> Starting playback...
> MPlayer interrupted by signal 11 in module: decode_audio
> ID_SIGNAL=11
> - MPlayer crashed by bad usage of CPU/FPU/RAM.
>   Recompile MPlayer with --enable-debug and make a 'gdb' backtrace and
>   disassembly. Details in
> DOCS/HTML/en/bugreports_what.html#bugreports_crash. - MPlayer crashed. This
> shouldn't happen.
>   It can be a bug in the MPlayer code _or_ in your drivers _or_ in your
>   gcc version. If you think it's MPlayer's fault, please read
>   DOCS/HTML/en/bugreports.html and follow the instructions there. We can't
> and won't help unless you provide this information when reporting a
> possible bug.
> 
> Thanks!

In case you need it here's the output of the information script:
http://www.alsa-project.org/db/?f=71cac1a2945fb0891b330a51d7a7d72b63818f64


------------------------------------------------------------------------------
Master HTML5, CSS3, ASP.NET, MVC, AJAX, Knockout.js, Web API and
much more. Get web development skills now with LearnDevNow -
350+ hours of step-by-step video tutorials by Microsoft MVPs and experts.
SALE $99.99 this month only -- learn more at:
http://p.sf.net/sfu/learnmore_122812
_______________________________________________
Alsa-user mailing list
Alsa-user@xxxxxxxxxxxxxxxxxxxxx
https://lists.sourceforge.net/lists/listinfo/alsa-user



[Index of Archives]     [ALSA Devel]     [Linux Audio Users]     [Fedora Users]     [Fedora Desktop]     [Fedora SELinux]     [Big List of Linux Books]     [Yosemite News]     [Yosemite Photos]     [KDE Users]     [Fedora Tools]

  Powered by Linux