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Return instead of putting all the code into a if-closure. That saves
one level of indentation.
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adtsParse() always returns 1, and its caller does not use the return
value.
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Since we eliminated the parameters retFileread and retTagsize in all
callers, we can now safely remove it from the function prototype.
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Anonymous code blocks just to declare variables look ugly. Move the
variable declarations up and disband the code block.
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Similar to previous patch: eliminate one variable by using "break".
This also simplifies the code since we can remove one level of indent.
[ew: rewritten to match current API]
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"break" is so much easier than "eof=1; continue;", when "!eof" is the
loop condition.
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Include only headers which are really required. This speeds up
compilation and helps detect cross-layer accesses.
[ew: minor fixups to not break on new core]
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Also enable -Wunused-parameter - this forces us to add the gcc
"unused" attribute to a lot of parameters (mostly library callback
functions), but it's worth it during code refactorizations.
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This has been tested for both playback of streams and
outputting to streams, and seems to work fine with minimal
locking. This reuses the sequence number infrastructure
in OutputBuffer for synchronizing metadata payloads; so
(IMNSHO) should be much more understandable than various
flags being set here and there..
It could still use some cleanup and much testing, but
synchronization issues should be minimal.
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data->muteFrame won't necessarily get cleared when it
enters that block of code, so we don't signal the action
as complete until it is actually cleared.
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We spawned the output buffer thread before daemonizing in
initPlayerData(), which is ultra bad because daemonizes forks
and threads are not preserved on exit. Since playerData has
been stripped bare by this core-rewrite anyways, move this code
into the outputBuffer_* group and drop playerData.[ch]
completely
I completely forgot to test this :<
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This is a huge refactoring of the core mpd process. The
queueing/buffering mechanism is heavily reworked.
The player.c code has been merged into outputBuffer (the actual
ring buffering logic is handled by ringbuf.c); and decode.c
actually handles decoding stuff.
The end result is several hundreds of lines shorter, even though
we still have a lot of DEBUG statements left in there for
tracing and a lot of assertions, too.
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If nothing has been read from the input stream, we don't have to
rewind it.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7397 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The variable "to_read" is never modified except in the last iteration
of the while loop. This means the while condition will never become
false, as the body will break before that may be checked.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7396 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This patch was taken from
http://bugzilla.livna.org/show_bug.cgi?id=1987 and addresses bug
0001693[1]
[1] - http://musicpd.org/mantis/view.php?id=1693
git-svn-id: https://svn.musicpd.org/mpd/trunk@7374 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7370 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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When we are in an input plugin, dc.current_song should already be
set. Use it instead of pc.current_song.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7363 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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We had functions names varied between
outputBufferFoo, fooOutputBuffer, and output_buffer_foo
That was too confusing for my little brain to handle.
And the global variable was somehow named 'cb' instead of
the more obvious 'ob'...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7355 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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All of our main singleton data structures are implicitly shared,
so there's no reason to keep passing them around and around in
the stack and making our internal API harder to deal with.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7354 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This at least makes the argument list to a lot of our plugin
functions shorter and removes a good amount of line nois^W^Wcode,
hopefully making things easier to read and follow.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7353 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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It actually increases our image size a small bit and may even
hurt performance a very small bit, but makes the code less
verbose and easier to manage.
I don't see a reason for mpd to ever support playing multiple
files at the same time (users can run multiple instances of mpd
if they really want to play Zaireeka, but that's such an edge
case it's not worth ever supporting in our code).
git-svn-id: https://svn.musicpd.org/mpd/trunk@7352 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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During my tests, it happened that data->position>newPosition. I have
not yet fully understood why this can happen; for now, replace this
with a run-time check.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7334 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The patch "convert blocks until the buffer is full" did not update
data->chunk_length correctly: it added the number of samples, not the
number of bytes. Multiply that with bytes_per_channel
git-svn-id: https://svn.musicpd.org/mpd/trunk@7332 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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In the patch "special optimized case for 16bit stereo", the check for
"num_channels==2" was missing.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7331 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Not having to loop for every sample byte (depending on a variable
unknown at compile time) saves a lot of CPU cycles. We could consider
reimplementing this function with liboil...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7330 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Read frame->header.channels once, and pass only this integer to
flac_convert().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7329 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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flacWrite() is the only function which sets data->chunk_length. If we
flush the buffer before we return, we can assume that it is always
empty upon entering flacWrite().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7328 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Move the inner loop which converts samples to flac_convert(). There
it is isolated and easier to optimize. This function does not have to
worry about buffer boundaries; the caller (i.e. flacWrite())
calculates how much is left and is responsible for flushing. That
saves a lot of superfluous range checks within the loop.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7327 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Check for flushing the chunk buffer only once per sample, before
iterating over channels and bytes. This saves another 5% CPU cycles.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7326 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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AudioFormat.bits is volatile, and to read it, 3 pointers had to be
deferenced. Calculate this value once. This speeds up this function
by 5%.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7325 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7324 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Try to only include headers which are really needed. We should
particularly check all "headers including other headers". The
long-term goal is to have a manageable, small API for plugins
(decoders, output) without so many mpd internals cluttering the
namespace.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7319 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7300 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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There were some const pointers missing in the previous const-cleanup
patch.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7290 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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libfaad wants uint32_t pointers. Passing a long pointer is bugged on
amd64.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7289 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The patch "Start using song pointers in core data structures" removed
dc->utf8url, and the adaption for wavpack_plugin.c was missing.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7288 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 way more complicated than it should be; and
locking it for thread-safety is too difficult.
[merged r7183 from branches/ew]
git-svn-id: https://svn.musicpd.org/mpd/trunk@7241 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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I initially started to do a heavy rewrite that changed the way processes
communicated, but that was too much to do at once. So this change only
focuses on replacing the player and decode processes with threads and
using condition variables instead of polling in loops; so the changeset
itself is quiet small.
* The shared output buffer variables will still need locking
to guard against race conditions. So in this effect, we're probably
just as buggy as before. The reduced context-switching overhead of
using threads instead of processes may even make bugs show up more or
less often...
* Basic functionality appears to be working for playing local (and NFS)
audio, including:
play, pause, stop, seek, previous, next, and main playlist editing
* I haven't tested HTTP streams yet, they should work.
* I've only tested ALSA and Icecast. ALSA works fine, Icecast
metadata seems to get screwy at times and breaks song
advancement in the playlist at times.
* state file loading works, too (after some last-minute hacks with
non-blocking wakeup functions)
* The non-blocking (*_nb) variants of the task management functions are
probably overused. They're more lenient and easier to use because
much of our code is still based on our previous polling-based system.
* It currently segfaults on exit. I haven't paid much attention
to the exit/signal-handling routines other than ensuring it
compiles. At least the state file seems to work. We don't
do any cleanups of the threads on exit, yet.
* Update is still done in a child process and not in a thread.
To do this in a thread, we'll need to ensure it does proper
locking and communication with the main thread; but should
require less memory in the end because we'll be updating
the database "in-place" rather than updating a copy and
then bulk-loading when done.
* We're more sensitive to bugs in 3rd party libraries now.
My plan is to eventually use a master process which forks()
and restarts the child when it dies:
locking and communication with the main thread; but should
require less memory in the end because we'll be updating
the database "in-place" rather than updating a copy and
then bulk-loading when done.
* We're more sensitive to bugs in 3rd party libraries now.
My plan is to eventually use a master process which forks()
and restarts the child when it dies:
master - just does waitpid() + fork() in a loop
\- main thread
\- decoder thread
\- player thread
At the beginning of every song, the main thread will set
a dirty flag and update the state file. This way, if we
encounter a song that triggers a segfault killing the
main thread, the master will start the replacement main
on the next song.
* The main thread still wakes up every second on select()
to check for signals; which affects power management.
[merged r7138 from branches/ew]
git-svn-id: https://svn.musicpd.org/mpd/trunk@7240 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The counter variables c_samp and c_chan begin at zero and can never be
negative.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7228 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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* 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
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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
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Terminals are 80 columns and that's a hard limit, no exceptions
git-svn-id: https://svn.musicpd.org/mpd/trunk@7207 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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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
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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
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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
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