Tools like "sparse" check for missing downcasts, since implicit cast
may be dangerous. Although that does not change the compiler result,
it may make the code more readable (IMHO), because you always see when
there may be data cut off.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7196 09075e82-0dd4-0310-85a5-a0d7c8717e4f
[ew: cleaned up the dirty union hack a bit]
Signed-off-by: Eric Wong <normalperson@yhbt.net>
git-svn-id: https://svn.musicpd.org/mpd/trunk@7180 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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
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
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
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
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
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
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
