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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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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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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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[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@7143 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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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
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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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
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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
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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
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Indent the entire tree, hopefully we can keep
it indented.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4387 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4333 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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the GPL header where necessary
git-svn-id: https://svn.musicpd.org/mpd/trunk@4317 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4132 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@3926 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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(from ticho)
git-svn-id: https://svn.musicpd.org/mpd/trunk@3477 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@2482 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1275 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1266 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1258 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1248 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1237 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1225 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1125 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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but new code for files seeking as well)
git-svn-id: https://svn.musicpd.org/mpd/trunk@1099 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1076 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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be gentle
git-svn-id: https://svn.musicpd.org/mpd/trunk@1051 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@968 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@942 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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implement in other decoders
git-svn-id: https://svn.musicpd.org/mpd/trunk@940 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@750 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@592 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@578 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@443 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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need to work on seeking for AAC
also, don't reset cb->begin on seek to 0,
instead just set cb->end=cb->begin, works much better for
disabling seeking (like ADIF AAC's)
git-svn-id: https://svn.musicpd.org/mpd/trunk@356 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@329 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@271 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@219 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@75 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@60 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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