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* outputBuffer: never calculate xfade time if xfade is offEric Wong2008-08-231-3/+5
| | | | | We don't assert on xfade_time > 0 inside any of the xfade calculations since we have no lock around xfade_time.
* outputBuffer: close audio device on stopEric Wong2008-08-231-6/+9
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* outputBuffer: fix buffer_before_play handlingEric Wong2008-08-231-12/+34
| | | | | | | | | | | | | | | | | | buffer_before_play is a prebuffer; always respecting it is almost as good as having no buffer at all. So we only respect it when we haven't played anything. Bugs that were a side effect of this also got fixed: The player would not stop when we got to the end of the last song on non-repeating playlists. The playlist would continuously show the song in the last few seconds of playback, and never move. Having crossfade enabled would also amplify the above effect. So, as a side effect, crossfade now correctly handles end-of-playlist conditions, as well. It will fade out to silence when we're at the end of a playlist.
* outputBuffer: drop buffered audio on new songsEric Wong2008-08-201-0/+1
| | | | Hopefully this fixes the skipping problem Qball reports
* Remove ob_wait_sync and cleanup triggering in playlistEric Wong2008-08-191-28/+17
| | | | | | | | ob_wait_sync was a gross hack anyways. We are one step closer to being able to trigger actions in our worker threads asynchronously. Just need to make input (file opening) in decoder happen _after_ our decoder returns an ACK.
* fix output buffer deadlock when daemonizingEric Wong2008-08-191-22/+1
| | | | | | | | | | | 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 :<
* core rewrite (decode,player,outputBuffer,playlist)Eric Wong2008-08-161-185/+476
| | | | | | | | | | | | | 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.
* added ob_set_lazy()Max Kellermann2008-04-151-1/+7
| | | | | | | | | 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
* Make the OutputBuffer API more consistentEric Wong2008-04-131-65/+65
| | | | | | | | | | 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
* Stop passing our single OutputBuffer object everywhereEric Wong2008-04-131-69/+69
| | | | | | | 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
* Stop passing our single DecoderControl object everywhereEric Wong2008-04-131-15/+13
| | | | | | | 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
* flush after loop in sendDataToOutputBuffer()Max Kellermann2008-04-121-5/+4
| | | | | | Since tailChunk() automatically flushes full buffers, we do not have to check this in every iteration of sendDataToOutputBuffer(). git-svn-id: https://svn.musicpd.org/mpd/trunk@7343 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* yet another migration to size_tMax Kellermann2008-04-121-4/+4
| | | | | We can also get rid of one the two variables. git-svn-id: https://svn.musicpd.org/mpd/trunk@7341 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* check cb->stop in the while loopMax Kellermann2008-04-121-4/+4
| | | | | | Checking dc->stop in the while condition and again after the while loop costs some CPU cycles we should save. git-svn-id: https://svn.musicpd.org/mpd/trunk@7340 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* eliminate OutputBuffer.currentChunkMax Kellermann2008-04-121-29/+51
| | | | | | | | | | | 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
* added output_buffer_expand()Max Kellermann2008-04-121-13/+22
| | | | | | | output_buffer_expand() moves the cb->end to the new position (only its current successor is allowed) and wakes up the player if is waiting for the decoder. This simplifies flushOutputBuffer(). git-svn-id: https://svn.musicpd.org/mpd/trunk@7338 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* initialize all elements in initOutputBuffer()Max Kellermann2008-04-121-0/+2
| | | | | | | The current OutputBuffer object is allocated statically, i.e. it is zeroed. To be safe for other cases in the future, also initialize the other elements. git-svn-id: https://svn.musicpd.org/mpd/trunk@7337 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* wake up player on demandMax Kellermann2008-04-121-1/+9
| | | | | | | | The decoder should not wake up the player when it did not produce a flushed chunk. Move the decoder_wakeup_player() call to flushOutputBuffer() and invoke it only if the buffer was previously empty. git-svn-id: https://svn.musicpd.org/mpd/trunk@7336 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* clean up CPP includesMax Kellermann2008-04-121-5/+0
| | | | | | | | | 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
* pass DecoderControl object to decoder_sleep()Max Kellermann2008-04-121-1/+1
| | | | | | | | | Less global variables: at any invocation of decoder_sleep(), we have a reference to the DecoderControl anyway, so we should pass it. This costs less than having to call getPlayerData() in every tiny function. Maybe some day we will be able to have multiple decoders at the same time... git-svn-id: https://svn.musicpd.org/mpd/trunk@7316 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* added output_buffer_free()Max Kellermann2008-04-121-0/+6
| | | | | | | 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
* don't use short chunk numbersMax Kellermann2008-04-121-2/+2
| | | | | | | | | | | | 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
* added output_buffer_skip()Max Kellermann2008-04-121-0/+6
| | | | | | | | 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
* pass buffered_chunks to initOutputBuffer()Max Kellermann2008-04-121-14/+17
| | | | | | 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
* fix typo in commentMax Kellermann2008-04-121-3/+3
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7305 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* let initOutputBuffer() allocate memoryMax Kellermann2008-04-121-2/+2
| | | | | | 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
* added inline function successor()Max Kellermann2008-04-121-11/+12
| | | | | | The new function successor() can be used to simplify a lot of code lines and saves a lot of "i+>=buffered_chunks" checks. git-svn-id: https://svn.musicpd.org/mpd/trunk@7285 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* moved currentChunk into OutputBufferMax Kellermann2008-04-121-10/+9
| | | | | | | | 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
* added outputBufferShift()Max Kellermann2008-04-121-0/+10
| | | | | Hiding OutputBuffer internals, yet again. Two more assertions. git-svn-id: https://svn.musicpd.org/mpd/trunk@7274 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* use sizeof(chunk.data) instead of CHUNK_SIZEMax Kellermann2008-04-121-2/+2
| | | | | | sizeof() is the more "natural" or "direct" access to the buffer size, instead of a macro happening to be used to the buffer declaration. git-svn-id: https://svn.musicpd.org/mpd/trunk@7270 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* added struct OutputBufferChunkMax Kellermann2008-04-121-17/+15
| | | | | | 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
* added outputBufferChunkData()Max Kellermann2008-04-121-0/+7
| | | | | | 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
* added outputBufferRelative()Max Kellermann2008-04-121-4/+9
| | | | | | | 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
* added outputBufferEmpty()Max Kellermann2008-04-121-0/+5
| | | | | | | 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
* moved check to outputBufferAbsolute()Max Kellermann2008-04-121-0/+17
| | | | | | | 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
* add method availableOutputBuffer()Max Kellermann2008-04-121-0/+8
| | | | | | | | | 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
* moved code to function tailChunk()Max Kellermann2008-04-121-33/+58
| | | | | | This patch removes some clutter from decodeParent() by moving some code out. git-svn-id: https://svn.musicpd.org/mpd/trunk@7247 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* use free()+malloc() instead of realloc()Max Kellermann2008-04-121-1/+3
| | | | | | | realloc() has to copy data to the new buffer. Since convBuffer contains temporary data only, we can safely use free() plus a new malloc(), which saves the mempy(). git-svn-id: https://svn.musicpd.org/mpd/trunk@7246 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* outputBuffer: remove unused variableEric Wong2008-04-121-1/+0
| | | | | [merged r7185 from branches/ew] git-svn-id: https://svn.musicpd.org/mpd/trunk@7243 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Drop metadata updates from HTTP for now (input HTTP, and shout)Eric Wong2008-04-121-76/+2
| | | | | | | | 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
* Initial cut of fork() => pthreads() for decoder and playerEric Wong2008-04-121-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* use unsigned integers in outputBuffer.cMax Kellermann2008-04-121-1/+1
| | | | | The chunk index cannot be negative. git-svn-id: https://svn.musicpd.org/mpd/trunk@7231 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix sign compare warningsMax Kellermann2008-04-121-1/+1
| | | | | | | Do explicit casts before comparing signed with unsigned. The one in log.c actually fixes another warning: in the expanded macro, there may be a check "logLevel>=0", which is always true. git-svn-id: https://svn.musicpd.org/mpd/trunk@7230 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* notify the decoder instead of polling 100hzMax Kellermann2008-03-261-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* moved code to initOutputBuffer()Max Kellermann2008-03-261-0/+17
| | | | | | | 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
* use size_tMax Kellermann2008-03-261-2/+2
| | | | | | | | 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
* fixed -Wshadow warningsMax Kellermann2008-01-261-2/+2
| | | | | Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7143 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Cleanup #includes of standard system headers and put them in one placeEric Wong2008-01-031-3/+1
| | | | | | | | | | | | | 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
* Make pcm_convertAudioFormat return the buffer size. This is necessaryJ. Alexander Treuman2007-05-261-3/+3
| | | | | | | | because lsr may return less than the input buffer size, and the rest of the audio code needs to know the new size. This fixes the clicking that was introduced with recent changes to the lsr code. A huge thanks to remiss for figuring this out. git-svn-id: https://svn.musicpd.org/mpd/trunk@6273 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Reverting to the full lsr API. Turns out the simple API needs all of theJ. Alexander Treuman2007-05-241-1/+2
| | | | | | | | 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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