From: Jonathan Andrews <jon@jonshouse.co.uk>
To: alsa-devel@alsa-project.org
Subject: Trying to understand alsa
Date: Thu, 12 Jan 2012 21:44:11 +0000 [thread overview]
Message-ID: <1326404651.22083.28.camel@jonspc> (raw)
Hi people, i'm trying to understand alsa....
I have an application that works using 512 sample packets of 22050Hz 16
bit mono audio. The 'receiver' takes many audio streams from a network
via UDP, at the moment it pipes them into pulse.
Can alsa buffer audio. At the moment every time I and set an audio
buffer size I get a negative response from
snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the
units alsa uses ...
What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x
1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K
Byte, 512 samples per buffer) in for playback. The packets are arriving
at roughly the correct rate, I just need a buffer to iron out any
jitter in network transmit, do I have to do this myself ?
Can somebody help by telling me which numbers I push into which places
to make it work ?
At the moment I get i keep getting a broken pipe, if I underrun how can
I make it just wait for me ?
I only code in plain old C at the moment.
Many thanks for any help/advice anyone can offer.
Jon
next reply other threads:[~2012-01-12 21:40 UTC|newest]
Thread overview: 8+ messages / expand[flat|nested] mbox.gz Atom feed top
2012-01-12 21:44 Jonathan Andrews [this message]
2012-01-12 21:50 ` Trying to understand alsa Fons Adriaensen
2012-01-12 22:12 ` Andrew Eikum
[not found] ` <1326410207.23310.17.camel@jonspc>
2012-01-13 15:10 ` Andrew Eikum
-- strict thread matches above, loose matches on Subject: below --
2012-01-12 23:36 Jonathan Andrews
2012-01-12 23:36 Jonathan Andrews
2012-01-12 23:45 Jonathan Andrews
2012-01-13 1:25 ` Jonathan Andrews
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