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| author | J. Alexander Treuman <jat@spatialrift.net> | 2007-05-23 10:31:07 +0000 |
|---|---|---|
| committer | J. Alexander Treuman <jat@spatialrift.net> | 2007-05-23 10:31:07 +0000 |
| commit | fbb3a94f17836aa33de78fd73db0f5d54464379d (patch) | |
| tree | 657629c023edd6eea29bb498d7a94ab003c32219 /src/audioOutputs/audioOutput_ao.c | |
| parent | a70ecdc00c588d824ae87d3084a5155140b70c99 (diff) | |
| download | mpd-fbb3a94f17836aa33de78fd73db0f5d54464379d.tar.gz mpd-fbb3a94f17836aa33de78fd73db0f5d54464379d.tar.xz mpd-fbb3a94f17836aa33de78fd73db0f5d54464379d.zip | |
Switching to the lsr simple API. The problem with the full API is that the
number of channels is specified when the converter state is created.
Previously this was only done once, thus breaking horribly when the input
audio suddenly had a different channel count. A new state could be created
every time the number of channels changes, but this could happen many times
a second if resampling to two different formats at once. The simple API
doesn't have this problem, as channel count is an argument to the
conversion function itself.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6229 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/audioOutputs/audioOutput_ao.c')
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