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After we've been hit by Large File Support problems several times in
the past week (which only occur on 32 bit platforms, which I don't
have), this is yet another attempt to fix the issue.
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Don't abort the whole MPD process when the conversion fails. This has
been a denial-of-service attack vector for years.
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It is illegal to pass a NULL buffer to pcm_byteswap_X(). The result
of this is that pcm_byteswap_X() never returns NULL.
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Currently, byteswapping is performed on the format_buffer. This can
go wrong when this buffer is used twice during one run. Add a
separate buffer for swapping the byte order.
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Changed "0" to "NULL".
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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 PCM sub libraries have 32 bit support now. Add support to the
glue function pcm_convert().
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The function is unused, since we added the pcm_buffer library.
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Removed yet another superfluous buffer layer: return the PCM buffer
from pcm_convert() instead of copying PCM data into the
caller-supplied buffer.
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gcc doesn't know that g_error() never returns. Work around the gcc
warning.
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dest_size is only used in an assertion. Remove its declaration and
move the formula into the assertion.
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Return a temporary buffer from pcm_resample_*() and pcm_convert().
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Replace a "static" buffer with the PCM buffer library.
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Replace a "static" buffer with the PCM buffer library.
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Free memory allocated by libsamplerate when the output or the decoder
is closed.
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In generic PCM functions allowing all sample formats, pass a void
pointer instead of a char pointer.
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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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The function does not need the full pcm_convert_state struct.
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Moved all code which converts the sample format to a separate library.
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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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Don't use libc's rand() function, because it is slow. Our own trivial
linear congruential generator is good enough for dithering.
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It is illegal to pass an empty audio buffer around. pcm_resample()
sometimes seems to result in 0 samples, maybe related to
libsamplerate. To work around that problem, add special checks after
both pcm_convert() invocations. Removed the pcm_resample()==0 checks
from pcm_convert().
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libsamplerate produces cracks in the sound output when the destination
buffer is too small. This is the case when pcm_convert_size() rounds
down. Use ceil(x) instead of floor(0.5+x) there to prevent a buffer
overrun.
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We want to remove gcc.h eventually. This takes care of all the
G_LIKELY/G_UNLIKELY macros.
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Allow pcm_volume() to increase volume.
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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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24 bit output is as important as 16 bit output. Provide a
pcm_convert() implementation which can convert to 24 bit with as
little quality loss as possible.
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The old pcm_convert_size() ignored most of the destination format,
e.g. it did not check its sample size, and assumed it is 16 bit.
Simplify and universalize it by using audio_format_frame_size().
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pcm_convert() converted only to 16 bit. To be able to support other
sample sizes, move that 16 bit specific code to a separate function.
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Separate code from pcm_utils.c to keep it small and simple.
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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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Splitting a frame between two buffer chunks causes distortion in the
output. MPD used to assume that the chunk size 1020 would never cause
splitted frames, but that isn't the case for 24 bit stereo (127.5
frames), and even less for files with even more channels.
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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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When calculating the conversion buffer size, don't hard-code the
formulas for only mono<->stereo.
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Convert any number of channels to stereo. In fact, this isn't really
stereo, it's rater mono blown up to stereo. This patch should only
make it possible to play 5.1 files at all; "real" conversion to stereo
should be implemented, but for now, this is better than nothing.
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In order to be able to deal with non-trivial conversions,
pcm_convertChannels() needs to know both the input and the output
channel count. Simplify buffer allocation in that function.
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Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2()
and pcm_convert_channels_2_to_1(). Improved the quality of
pcm_convert_channels_2_to_1() by calculating the arithmetic mean value
of both samples.
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Pass int16_t pointers instead of char pointers to functions which can
deal with 16 bit audio only.
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The last bit of CamelCase in audio_format.h. Additionally, rename a
bunch of local variables.
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"bits" and "channels" cannot be negative.
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In the libsamplerate fallback code, a "const" attribute was missing.
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"volume" was passed as an unsigned integer, which is correct. It's
just that when it was multiplied with the sample value, the whole
operation was changed to unsigned, breaking the algorithm (and Qball's
ears). Internally change "volume" to signed.
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