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Instead of checking the stream_types bit set, we can simply check
whether the methods stream_decode() and file_decode() are implemented.
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Don't pass the "seekable" flag with every decoder_data() invocation.
Since that flag won't change within the file, it is enough to pass it
to decoder_initialized() once per file.
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The strings were constant, but the pointers weren't. C syntax is
somewhat tricky..
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All decoder_plugin structs are initialized at compile time, and must
never change.
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Don't return 0/-1 on success/error, but true/false. Instead of int,
use bool for storing flags.
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A decoder_flush() invocation was missing in the FLAC plugin, resulting
in casual assertion failures due to a wrong assumption about the last
chunk's audio format. It's much easier to remove that decoder_flush()
function and make the decoder thread call ob_flush().
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Call ob_clear() in decoder_command_finished() instead of implementing
that call in every decoder plugin.
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These plugins are not input plugins, they are decoder plugins. No
CamelCase in the directory name.
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Don't compile the sources of disabled decoder plugins at all, and
don't attempt to register these.
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The last bit of CamelCase in audio_format.h. Additionally, rename a
bunch of local variables.
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When there are standardized headers, use these instead of the bloated
os_compat.h.
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The old struct initializers are error prone and don't allow moving
elements around. Since we are going to overhaul some of the APIs
soon, it's easier to have all implementations use C99 initializers.
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Do full C99 integer type conversion in all modules which were not
touched by Eric's merged patch.
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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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This releases several include file dependencies. As a side effect,
"CHUNK_SIZE" isn't defined by decoder_api.h anymore, so we have to
define it directly in the plugins which need it. It just isn't worth
it to add it to the decoder plugin API.
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"decoder plugin" is a better name than "input plugin", since the
plugin does not actually do the input - InputStream does. Also don't
use typedef, so we can forward-declare it if required.
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Provide access to seeking for the decoder plugins; they have to know
where to seek, and they need a way to tell us that seeking has failed.
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Some decoder commands are implemented in the decoder plugins, thus
they need to have an API call to signal that their current command has
been finished. Let them use the new decoder_command_finished()
instead of the internal dc_command_finished().
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Another big patch which hides internal mpd APIs from decoder plugins:
decoder plugins regularly poll dc->command; expose it with a
decoder_api.h function.
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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.
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Similar to the previous patch: pass total_time instead of manipulating
dc->totalTime directly.
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dc->audioFormat is set once by the decoder plugins before invoking
decoder_initialized(); hide dc->audioFormat and let the decoder pass
an AudioFormat pointer to decoder_initialized().
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We are now beginning to remove direct structure accesses from the
decoder plugins. decoder_clear() and decoder_flush() mask two very
common buffer functions.
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Moved all of the player-waiting code to decoder_data(), to make
OutputBuffer more generic.
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decoder_initialized() sets the state to DECODE_STATE_DECODE and wakes
up the player thread. It is called by the decoder plugin after its
internal initialization is finished. More arguments will be added
later to prevent direct accesses to the DecoderControl struct.
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The decoder struct should later be made opaque to the decoder plugin,
because maintaining a stable struct ABI is quite difficult. The ABI
should only consist of a small number of stable functions.
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dc_command_finished() is invoked by the decoder thread when it has
finished a command (sent by the player thread). It resets dc.command
and wakes up the player thread. This combination was used at a lot of
places, and by introducing this function, the code will be more
readable.
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Much of the existing code queries all three variables sequentially.
Since only one of them can be set at a time, this can be optimized and
unified by merging all of them into one enum variable. Later, the
"command" checks can be expressed in a "switch" statement.
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Include only headers which are really required. This speeds up
compilation and helps detect cross-layer accesses.
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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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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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