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paziu
Tux's lil' helper
Joined: 24 Nov 2006
Posts: 78
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 Posted: Fri Nov 07, 2014 6:20 am Post subject: alsa/mplayer question ( sampling ) |
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I am a bit confused at this moment, is mplayer/mpg123/alsa resampling to 48kHz before the output?
| Code: | ==========================================================================
Opening audio decoder: [mpg123] MPEG 1.0/2.0/2.5 layers I, II, III
AUDIO: 44100 Hz, 2 ch, s16le, 320.0 kbit/22.68% (ratio: 40000->176400)
Selected audio codec: [mpg123] afm: mpg123 (MPEG 1.0/2.0/2.5 layers I, II, III)
==========================================================================
AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample)
Video: no video |
tx
edit: maybe it should be moved to kernel? |
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paziu
Tux's lil' helper
Joined: 24 Nov 2006
Posts: 78
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| Posted: Fri Nov 07, 2014 9:46 am Post subject: |
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| going from 44.1 to 48kHz causes interpolation, basically loss(?) |
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EmaRsk
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Joined: 07 Sep 2004
Posts: 158
Location: Italy
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| Posted: Sat Nov 08, 2014 3:04 pm Post subject: |
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I may be wrong, but it probably means that your sound device's DAC works at 48kHz.
Maybe at http://linuxmusicians.com/ you could find a better answer. |
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EmaRsk
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Joined: 07 Sep 2004
Posts: 158
Location: Italy
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| Posted: Sat Nov 08, 2014 3:18 pm Post subject: |
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| I found that in /proc/asound/card2/codec#0 (substitute "card2" with what's relevant for you) I can see the allowed sample rates for my device (look for "rates"). |
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paziu
Tux's lil' helper
Joined: 24 Nov 2006
Posts: 78
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| Posted: Sun Nov 09, 2014 4:58 am Post subject: |
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Thanks!
then I need to remember to switch it before playing a 48khz sample.
I still think it is better then being stuck in the 8bit world, but.... |
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