This patch does the following:
-enables WVC support for streams as well,
-improves MPD inputStream <=> WavPack stream connector,
-fixes two compile warnings (which were caused by MPD API change).
Mantis #1660 <http://musicpd.org/mantis/view.php?id=1660>
git-svn-id: https://svn.musicpd.org/mpd/trunk@7210 09075e82-0dd4-0310-85a5-a0d7c8717e4f
The local variable eof can actually be replaced with a simple "break".
With a negative ret, the value of chunkpos can be invalidated, I am
not sure if this might have been a bug.
[ew: no, a negative ret will correspond to ret == OV_HOLE and ret
will be reset to zero leaving chunkpos untouched (code cleaned up
to make this more obvious]
git-svn-id: https://svn.musicpd.org/mpd/trunk@7202 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Local variables which are never read before the first assignment don't
need initialization. Saves a few bytes of text. Also don't reset
variables which are never read until function return.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7199 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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
From <http://wiki.xiph.org/index.php/MIME_Types_and_File_Extensions>:
> .oga - audio/ogg
>
> * Ogg Audio Profile (audio in Ogg container)
> * Applications supporting .oga, .ogv SHOULD support decoding
> from muxed Ogg streams
> * Covers Ogg FLAC, Ghost, and OggPCM
> * Although they share the same MIME type, Vorbis and Speex
> use different file extensions.
> * SHOULD contain a Skeleton logical bitstream.
> * Vorbis and Speex may use .oga, but it is not the
> prefered(sic) method of distributing these files because of
> backwards-compatibility issues.
Thanks to Qball and Rasi for the patch.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7191 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
the code is inconsistent when FLAC_API_VERSION_CURRENT is not defined:
sometimes version > 7 is assumed, and sometimes version <= 7. solve
this by assuming the version is old when FLAC_API_VERSION_CURRENT is
not defined.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
git-svn-id: https://svn.musicpd.org/mpd/trunk@7144 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
The updated initialize method did not tell the libFLAC to look for the tag containing the replay information.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7075 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Fixing stopping mpd from block when trying to stop a ogg stream that is buffering.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7053 09075e82-0dd4-0310-85a5-a0d7c8717e4f
ogg_stream_type_detect may not be compiled correctly
when compiling FLAC (1.1.4+) without Vorbis
git-svn-id: https://svn.musicpd.org/mpd/trunk@6896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Both mp4 and (ogg)flac inputPlugins got HTTP inputStream support
later in the game, so their calls to sendDataToOutputBuffer()
didn't get updated to support buffering while the outputBuffer
was full. This fixes it.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6873 09075e82-0dd4-0310-85a5-a0d7c8717e4f
the force flag will issue FATAL() if an invalid value is
specified
git-svn-id: https://svn.musicpd.org/mpd/trunk@6857 09075e82-0dd4-0310-85a5-a0d7c8717e4f
For the default: case, just use the error message that libFLAC
provides instead of using something ambiguous. Also, this gets
rid of long lines in the code, making it easier to digest.
Of course, we save ~100 bytes of text space in the process :)
git-svn-id: https://svn.musicpd.org/mpd/trunk@6830 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Parse ReplayGain info in LAME tags and use it if no ID3v2 ReplayGain tags
are found. This is currently a bit unsafe, as apparently some LAME tags
have bogus ReplayGain values. But I'm finding a lot of MP3s with valid
LAME tags that fail the LAME tag CRC check. So until I figure out why
that's happening, it's an unreliable method for checking if the LAME tag is
valid.
A big thanks to tmz for writing the original patch.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6798 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Turns out the fix was as simple as specifying the OPEN_TAGS flag when
opening the file. Thanks again to Kodest for figuring this one out.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6657 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This ReplayGain code is currently disabled because WavpackGetTagItem can't
seem to find replaygain_* fields in APEv2 tags (which is how wvgain stores
ReplayGain values). Additionally, because APEv2 tags are stored at the end
of the file, this code is only implemented for regular files and not HTTP
streams. Using HTTP seeking it *may* be possible to implement it for both.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6656 09075e82-0dd4-0310-85a5-a0d7c8717e4f
have any effect until the aac and mp4 input plugins actually support a
stream decoding API.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6481 09075e82-0dd4-0310-85a5-a0d7c8717e4f
size_t (1.1.3) makes a lot more sense, but older flac used unsigned
here...
git-svn-id: https://svn.musicpd.org/mpd/trunk@5258 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
We'll be dealing with legacy server configurations for a long
time to come.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5253 09075e82-0dd4-0310-85a5-a0d7c8717e4f
sendDataToOutputBuffer returns an int (and always has), and
using the existing 'ret' is fine in mp3Read().
git-svn-id: https://svn.musicpd.org/mpd/trunk@5246 09075e82-0dd4-0310-85a5-a0d7c8717e4f
MP3 playback, thus allowing songs that run longer than the Xing frame
claims (f.e., an MP3 created by catting two MP3s together) to continue
playing past the end.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5157 09075e82-0dd4-0310-85a5-a0d7c8717e4f
assumption that non-seekable streams are live and any gapless info is
incorrect.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5150 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Instead, stop decoding as soon as we've found the frames/samples at the
"end" that we want drop.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5149 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This means that when using libFLAC as a shared object,
OggFLAC support is dependent on the compile-time options of
the libFLAC library loaded.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5112 09075e82-0dd4-0310-85a5-a0d7c8717e4f
We will restore compatibility with the old API in the
next few commits; along with OggFLAC support.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5110 09075e82-0dd4-0310-85a5-a0d7c8717e4f
move flac_decode to the bottom, so we don't have to declare
all of our static functions.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5109 09075e82-0dd4-0310-85a5-a0d7c8717e4f
I'm checking for zero-size allocations and assert()-ing them,
so we can more easily get backtraces and debug problems, but we'll
also allow -DNDEBUG people to live on the edge if they wish.
We do not rely on errno when checking for OOM errors because
some implementations of malloc do not set it, and malloc
is commonly overridden by userspace wrappers.
I've spent some time looking through the source and didn't find any
obvious places where we would explicitly allocate 0 bytes, so we
shouldn't trip any of those assertions.
We also avoid allocating zero bytes because C libraries don't
handle this consistently (some return NULL, some not); and it's
dangerous either way.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4690 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
This bug was NOT introduced in my OggFLAC additions, honest!
As far as I can see, it was introduced way back in r2482, but
nobody ever noticed until the post here:
http://www.musicpd.org/forum/index.php?topic=1152.0
While we're at it, clean up some of the variable typing.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4664 09075e82-0dd4-0310-85a5-a0d7c8717e4f
I'm not using __FUNCTION__ or __func__ because compiler support
for these is still a bit iffy as far as I know...
git-svn-id: https://svn.musicpd.org/mpd/trunk@4662 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Unfortunately there doesn't seem to be an indent switch for this,
but we have find + perl:
find src -name '*.[ch]' | xargs perl -i -p -e \
's/^\s+(\w+):/$1:/g unless /^\s+default:/'
This is a followup to r4605, and there are no actual code
changes in this.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4661 09075e82-0dd4-0310-85a5-a0d7c8717e4f
playerData.c:
proper error checking
directory.c:
properly check myFgets() for errors
(it returns NULL on error)
inputPlugins/mp3_plugin.c
get rid of commas at the end of enums
interface.c:
we weren't using long long, so strtoll isn't needed
get rid of void-pointer arithmetic
sllist.c:
get rid of void-pointer arithmetic
compress.c:
get rid of C++ comments, some compilers don't accept them
Note that I personally like void pointer arithmetic, but some
ancient compilers don't support them :(
git-svn-id: https://svn.musicpd.org/mpd/trunk@4510 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
Both values are compiled to zero, but this is more correct
since we're using the correct enum (in the unlikely case that
the FLAC library breaks compatibility).
git-svn-id: https://svn.musicpd.org/mpd/trunk@4379 09075e82-0dd4-0310-85a5-a0d7c8717e4f
These are just warnings from sparse, but it makes the output
easier to read. I ran this through a quick perl script, but
of course verified the output by looking at the diff and making
sure the thing still compiles.
here's the quick perl script I wrote to generate this patch:
----------- 8< -----------
use Tie::File;
defined(my $pid = open my $fh, '-|') or die $!;
if (!$pid) {
open STDERR, '>&STDOUT' or die $!;
exec 'sparse', @ARGV or die $!;
}
my $na = 'warning: non-ANSI function declaration of function';
while (<$fh>) {
print STDERR $_;
if (/^(.+?\.[ch]):(\d+):(\d+): $na '(\w+)'/o) {
my ($f, $l, $pos, $func) = ($1, $2, $3, $4);
$l--;
tie my @x, 'Tie::File', $f or die "$!: $f";
print '-', $x[$l], "\n";
$x[$l] =~ s/\b($func\s*)\(\s*\)/$1(void)/;
print '+', $x[$l], "\n";
untie @x;
}
}
git-svn-id: https://svn.musicpd.org/mpd/trunk@4378 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Functions that should stay inlined should have an explanation
attached to them.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4355 09075e82-0dd4-0310-85a5-a0d7c8717e4f
the performance optimization was broken for big-endian architectures.
mpd-uclinux is doing using a slightly different optimization to flacWrite()
that we may end up using here in the future.
git-svn-id: https://svn.musicpd.org/mpd/trunk@3753 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* autotools support is included, the floating point Vorbis decoder
remains the default.
* close bug #353
* Thanks to Hannes Reich for the patch
git-svn-id: https://svn.musicpd.org/mpd/trunk@3453 09075e82-0dd4-0310-85a5-a0d7c8717e4f
