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No CamelCase in the file name. The output_buffer struct is going to
be renamed to music_pipe. There are so many buffer levels in MPD, and
calling this one "output buffer" is wrong, because it's not the last
buffer before the music reaches the output devices.
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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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The function ob_is_empty() is called very often. It's worth it to
convert it to an inline function.
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"bool" should be used in C99 programs for boolean values.
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Typedefs shouldn't be used, use the bare struct names instead.
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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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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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(Ab)use the decoder_command enumeration, which has nearly the same
values and the same meaning.
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Since we moved all PCM conversions to decoder_data(), the attribute
convState isn't being used anymore by the OutputBuffer code. Move it
to struct decoder.
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OutputBuffer should be a more generic low-level library, without
dependencies to the other headers. This patch adds the field
"notify", which is used to signal the player thread. It is passed in
the constructor, and removes the need to compile with the decode.h
header.
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Moved all of the player-waiting code to decoder_data(), to make
OutputBuffer more generic.
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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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In lazy mode (previously the default), outputBuffer.c only wakes up
the player when it was previously empty. That caused a deadlock when
the player was waiting for buffered_before_play, since the decoder
wouldn't wake up the player when buffered_before_play was reached. In
non-lazy mode, always wake up the player when a new chunk was decoded.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7364 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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OutputBuffer.currentChunk contains redundant data: it is either -1
when there is no chunk which is currently being written, or it equals
"end". If we always keep chunk[end] in a valid state, we can remove
OutputBuffer.currentChunk.
This patch may look a bit clumsy, especially flushOutputBuffer(), but
that will be fixed later with an major OutputBuffer API overhaul.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7339 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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There is no danger that gcc will optimize access to OutputBufferChunk
properties, since decoder and player work in different chunk objects.
It is safe to remove "volatile" here.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7318 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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To do proper cleanup before exiting, we have to provide a destructor
for OutputBuffer. One day, valgrind will not complain about memory
leaks!
git-svn-id: https://svn.musicpd.org/mpd/trunk@7315 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Don't be mean with integer sizes. Although we will probably never
have more than 32k buffered chunks, we should use 32 bit integers for
addressing them. We do not save very much (some of the saved space is
eaten by alignment anyway), but we save at least one assembler
instruction for converting short to int.
This change requires some more explicit casts, because gcc was less
picky when comparing short with a full int.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7313 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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First patch without camelCase ;)
output_buffer_skip() lets us eliminate advanceOutputBufferTo(), and
removes yet another external OutputBuffer struct access.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7312 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Try to make OutputBuffer self-contained, without depending on a global
variable.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7310 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This is the first patch in a series which removes the shared memory,
and moves all the playerData objects into the normal libc heap.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7304 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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currentChunk is a global variable, which renders the whole output
buffer code non-reentrant. Although this is not a real problem since
there is only one global output buffer currently, we should move it to
the OutputBuffer struct.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7284 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Hiding OutputBuffer internals, yet again. Two more assertions.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7274 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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To make access to OutputBuffer easier, move everything which belongs
to a chunk into its own structure, namely OutputBufferChunk.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7269 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The chunk size should be in outputBuffer.h since the output buffer
code is its primary user.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7268 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Hiding OutputBuffer internals, again. We get an extra assertion in
return.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7267 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The cross-fade check is still very complicated whenever it uses
OutputBuffer internals. Greatly simplify another check by introducing
outputBufferRelative().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7264 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Another "don't use OutputBuffer internals" patch. This ignores the
copied "end" value, but I do not think that has ever been a real
issue.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7263 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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decoderParent() uses a lot of OutputBuffer internals to see whether
cross-fading should be started. Move these checks to outputBuffer.c,
which also simplifies decoderParent().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7262 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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The method availableOutputBuffer() calculates how many chunks are in
use. This simplifies code which needs this information, and it can
run without knowing OutputBuffer internals. The function knows how to
calculate this when begin>end; this might have been a bug in
decodeParent(), which does not.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7250 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7245 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 chunk index cannot be negative.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7231 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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When the decoder process is faster than the player process, all
decodedd buffers are full at some point in time. The decoder has to
wait for buffers to become free (finished playing). It used to do
this by polling the buffer status 100 times a second.
This generates a lot of unnecessary CPU wakeups. This patch adds a
way for the player process to notify the decoder process that it may
continue its work.
We could use pthread_cond for that, unfortunately inter-process
mutexes/conds are not supported by some kernels (Linux), so we cannot
use this light-weight method until mpd moves to using threads instead
of processes. The other method would be semaphores, which
historically are global resources with a unique name; this historic
API is cumbersome, and I wanted to avoid it.
I came up with a quite naive solution for now: I create an anonymous
pipe with pipe(), and the decoder process reads on that pipe. Until
the player process sends data on it as a signal, the decoder process
blocks.
This can be optimized in a number of ways:
- if the decoder process is still working (instead of waiting for
buffers), we could save the write() system call, since there is
nobody waiting for the notification.
[ew: I tried this using a counter in shared memory, didn't help]
- the pipe buffer will be full at some point, when the decoder thread
is too slow. For this reason, the writer side of the pipe is
non-blocking, and mpd can ignore the resulting EWOULDBLOCK.
- since we have shared memory, we could check whether somebody is
actually waiting without a context switch, and we could just not
write the notification byte.
[ew: tried same method/result as first point above]
- if there is already a notification in the pipe, we could also not
write another one.
[ew: tried same method/result as first/third points above]
- the decoder will only consume 64 bytes at a time. If the pipe
buffer is full, this will result in a lot of read() invocations.
This does not hurt badly, but on a heavily loaded system, this might
add a little bit more load. The preceding optimizations however
are able eliminate the this.
- finally, we should use another method for inter process
notifications - maybe kill() or just make mpd use threads, finally.
In spite of all these possibilities to optimize this code further,
this pipe notification trick is faster than the 100 Hz poll. On my
machine, it reduced the number of wakeups to less than 30%.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7215 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This patch moves code which initializes the OutputBuffer struct to
outputBuffer.c. Although this is generally a good idea, it prepares
the following patch.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7206 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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When dealing with in-memory lengths, the standard type "size_t" should
be used. Missing one can be quite dangerous, because an attacker
could provoke an integer under-/overflow, which may provide an attack
vector.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7205 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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git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 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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void * rather than "x char *"
git-svn-id: https://svn.musicpd.org/mpd/trunk@4408 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@2482 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@1439 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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