The local variable d_samp is initialized, but never actually used.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7227 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* malloc() => xmalloc() for error checking
* strncpy() replaced with memcpy(),
memcpy appears perfectly safe here and mpd
does not ever use strncpy() (see r4491)
git-svn-id: https://svn.musicpd.org/mpd/trunk@7211 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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
