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On Win32, the third setsockopt parameter has type (char *) while on POSIX
systems it is (void *). However, given that it is a no-op cast to go from a
char pointer to a void pointer, we can cast to a char pointer (with a
possible const modifier) on all platforms and satisfy the compiler.
Signed-off-by: Dan McGee <dan@archlinux.org>
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Wrap close(), use closesocket() on WIN32/WinSock.
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If a connected host disappears without our knowledge, as can happen over
wireless or a hibernating machine, we continue to hold the port open waiting
for messages. Because we never try to send anything down this now-broken
pipe, the connection will sit idle taking up a slot in our allowed incoming
connections list.
If enough of these happen, an unintended Denial of Service takes place,
where all connection slots are filled with now-broken, never ending
connections. Setting the TCP keepalive option at least allows these to time
out after the default two hours, which is sufficient in the non-malicious
case.
Signed-off-by: Dan McGee <dan@archlinux.org>
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In the winsock headers, the setsockopt() argument is declared as
"const char *", not "const void *".
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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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Changed the wrappers for pipe(), socket(), accept(). On WIN32, this
does not work for pipe().
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Added the "fd_util" library, which attempts to use the new thread-safe
Linux system calls pipe2(), accept4() and the options O_CLOEXEC,
SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is
not thread safe.
This is particularly important for the "pipe" output plugin (and
others, such as JACK/PulseAudio), because we were heavily leaking file
descriptors to child processes.
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g_set_error() is a printf-like function, and expects a format string.
Using the return value of gai_strerror() is unsafe.
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g_strerror() is more portable, and guarantees that the returned string
is UTF-8 encoded.
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Moved code from listen_add_address() (listen.c) to socket_util.c.
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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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On some systems, the macro IN6_IS_ADDR_V4MAPPED() is not available.
Don't try to convert IPv6 to their IPV4 equivalents in this case.
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Windows doesn't have the standard headers sys/socket.h and netdb.h.
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Unpack IPv4 addresses which are packed inside an IPv6 address,
i.e. return "127.0.0.1" rather than "::ffff:127.0.0.1".
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Create the socket_util.c library, the first function is
sockaddr_to_string(): it converts a sockaddr struct to a string
containing the IP address in a human-readable form.
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This new LAME encoder plugin is based on the existing shout_mp3.c
plugin.
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This new vorbis encoder plugin is based on the existing shout_ogg.c
plugin.
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The new generic encoder API will replace shout's custom encoder API.
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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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redirect_stdin() is a daemonization function, and disconnecting from
the standard input is always a good idea for MPD.
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Prepare for the migration to the GLib option parser, which uses
gboolean for flags.
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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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When there are standardized headers, use these instead of the bloated
os_compat.h.
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Similar to decoder_control.c, output_control.c will provide functions
for controlling the output thread (which will be implemented later).
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This function is declared, but is neither used nor implemented.
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Instead of having to register each output plugin, store them
statically in an array. This eliminates the need for the List library
here, and saves some small allocations during startup.
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Also rename AudioOutputPlugin to struct audio_output_plugin, and use
forward declarations to reduce include dependencies.
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Just like decoder_api.h, output_api.h provides the audio output API
which is used by the plugins.
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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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Getting rid of CamelCase; not having typedefs also allows us to
forward-declare the structures.
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And again, convert arguments to const.
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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@7345 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7294 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The audio output plugins should get a constant pointer, because they
must not modify the buffer. Since the size is a non-negative buffer
size in bytes, we should change its type to size_t.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7293 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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[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
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Signed-off-by: Eric Wong <normalperson@yhbt.net>
git-svn-id: https://svn.musicpd.org/mpd/trunk@7142 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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except that it now uses a timer for throttling.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6621 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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outputs, which is actually desired behaviour. This way if the shout server
takes a while to respond, the shout output can block until connected
without messing up other audio outputs.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6554 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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reducing the likelyhood of it blocking other outputs.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6543 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6393 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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