| Commit message (Collapse) | Author | Age | Files | Lines |
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This will make refactoring features easier, especially now that
pthreads support and larger refactorings are on the horizon.
Hopefully, this will make porting to other platforms (even
non-UNIX-like ones for masochists) easier, too.
os_compat.h will house all the #includes for system headers
considered to be the "core" of MPD. Headers for optional
features will be left to individual source files.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7130 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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DECODE_STATE_STOP is always set as dc->state, and dc->stop
is always cleared. So handle it in decodeStart once rather
than doing it in every plugin.
While we're at it, fix a long-standing (but difficult to
trigger) bug in mpc_decode where we failed to return
if mpc_decoder_initialize() fails.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7122 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Some compilers and linkers aren't smart enough to optimize this,
as global variables are implictly initialized to zero. As a
result, binaries are a bit smaller as more goes in the .bss and
less in the text section.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5254 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This finally fixes a bug from over two years ago playing a wave file
(oprah.wav) with the following characteristics (from sfinfo):
File Format Microsoft RIFF WAVE Format (wave)
Data Format 8-bit integer (unsigned, little endian)
Audio Data 986827 bytes begins at offset 58 (3a hex)
1 channel, 986827 frames
Sampling Rate 22050.00 Hz
Duration 44.754 seconds
Of course, this has been regression tested with all the files
that the previous commit got working. Thanks to Michael Pruett
(audiofile author) for the hint and shame on me for forgetting
about it for over two years :x
git-svn-id: https://svn.musicpd.org/mpd/trunk@4682 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Use the 'Virtual' variants of afGetSampleFormat, afGetChannels,
afGetVirtualFrameSize in the audiofile library, since it already does
the necessary abstraction for us.
Of course, I've regression tested these changes against my
standard 44100Hz/2ch/16bit wave files and they continue to play
fine.
Files tested:
english.au (Linus Torvalds pronouncing 'Linux' in English)
B01.Red_Bright_Heart.au (Chinese opera, sounds correct to me even though
I don't actually understand the words)
git-svn-id: https://svn.musicpd.org/mpd/trunk@4681 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Add a few new options for indent to try to make
things a bit cleaner
git-svn-id: https://svn.musicpd.org/mpd/trunk@4411 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Indent the entire tree, hopefully we can keep
it indented.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4387 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Nothing here is ever exported for linkage besides the
InputPlugin structure, so mark them static to save a few bytes.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4382 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4333 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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the GPL header where necessary
git-svn-id: https://svn.musicpd.org/mpd/trunk@4317 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4132 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@3926 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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(from ticho)
git-svn-id: https://svn.musicpd.org/mpd/trunk@3477 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@2482 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1275 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1266 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1258 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1248 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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