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Move from PcmClamp().
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Support for conversion from float to 16, 24 and 32 bit integer
samples.
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Support 32 bit samples with software mixer.
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This updates the copyright header to all be the same, which is
pretty much an update of where to mail request for a copy of the GPL
and the years of the MPD project. This also puts all committers under
'The Music Player Project' umbrella. These entries should go
individually in the AUTHORS file, for consistancy.
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All what's left in pcm_utils.h is the pcm_range() utility function,
which is only used internally by pcm_volume and pcm_mix.
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Moved the software mixing code (used by crossfading) to a separate
library.
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Moved the software volume code to a separate library.
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We are going to split the pcm_utils.c library, and pcm_range() will be
useful for several sub libraries.
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It may be desirable to change the range of integer volume levels
(e.g. to 1024, which may utilize shifts instead of expensive integer
divisions). Introduce the constant PCM_VOLUME_1 which describes the
integer value for "100% volume". This is currently 1000.
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"LOG_H" is a macro which is also used by ffmpeg/log.h. This is
ffmpeg's fault, because short macros should be reserved for
applications, but since it's always a good idea to choose prefixed
macro names, even for applications, we are going to do that in MPD.
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Separate the resampling code from the rest of pcm_utils.c. Create two
sub-libraries: pcm_resample_libsamplerate.c and
pcm_resample_fallback.c.
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Copied and adapted code from the mp3 decoder plugin. This library now
replaces the old and low-quality function pcm_convert_24_to_16().
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Instead of manually calling memset(0) on the pcm_convert_state struct,
client code should use a library function from pcm_utils.c. This way,
we can change the semantics of the struct easily.
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Renamed all functions which were still in CamelCase.
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No CamelCase, and a struct instead of a typedef.
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pcm_mix() might overflow the destination buffer if it is smaller than
the second buffer. This is ok because the physical buffer size passed
by cross_fade_apply() is always big enough, but clutters pcm_mix()
with complicated length checks and contains a dangerous buffer
overflow pitfall. Simplify pcm_mix()/pcm_add() and pass only the
smaller buffer size; let cross_fade_apply() do the memcpy().
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Since we use a C99 compiler now, we can assert that the C99 standard
headers are available, no need for complicated compile time checks.
Kill mpd_types.h.
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Seeing the "mpd_" prefix _everywhere_ is mind-numbing as the
mind needs to retrain itself to skip over the first 4 tokens of
a type to get to its meaning. So avoid having extra characters
on my terminal to make it easier to follow code at 2:30 am in
the morning.
Please report any new issues you may come across on Free
toolchains. I realize how difficult it can be to build/maintain
cross-compiling toolchains and I have no intention of forcing
people to upgrade their toolchains to build mpd.
Tested with gcc 2.95.4 and and gcc 4.3.1 on x86-32.
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Get rid of CamelCase, and don't use a typedef, so we can
forward-declare it, and unclutter the include dependencies.
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We want to expose the AudioFormat structure to plugins; remove some
clutter by moving its declaration to a separate header file.
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7360 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7287 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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It is a good practice to constify pointers when their dereferenced
data is not modified within the functions or its descendants.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7234 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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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git-svn-id: https://svn.musicpd.org/mpd/trunk@7042 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7041 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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because lsr may return less than the input buffer size, and the rest of the
audio code needs to know the new size. This fixes the clicking that was
introduced with recent changes to the lsr code. A huge thanks to remiss
for figuring this out.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6273 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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audio at once, so it won't work for us. The old full API code was still
heavily broken, as each call to pcm_convertSampleRate() used the same
state, even if it was processing two streams of audio. The new code keeps
a separate state for each audio stream that's being converted.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6255 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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and samplerate conversion. This makes the code much easier to read, and
fixes a few bugs that were previously there.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6224 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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git-svn-id: https://svn.musicpd.org/mpd/trunk@4470 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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