mpd/src/outputBuffer.c

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/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "outputBuffer.h"
#include "pcm_utils.h"
#include "playerData.h"
#include "utils.h"
#include "log.h"
#include "normalize.h"
#include "conf.h"
#include "os_compat.h"
static mpd_sint16 currentChunk = -1;
void initOutputBuffer(OutputBuffer * cb, OutputBufferChunk * chunks)
{
memset(&cb->convState, 0, sizeof(ConvState));
cb->chunks = chunks;
}
void clearOutputBuffer(OutputBuffer * cb)
{
cb->end = cb->begin;
currentChunk = -1;
}
void flushOutputBuffer(OutputBuffer * cb)
{
if (currentChunk == cb->end) {
if (((unsigned)cb->end + 1) >= buffered_chunks) {
cb->end = 0;
}
else cb->end++;
currentChunk = -1;
}
}
int outputBufferEmpty(const OutputBuffer * cb)
{
return cb->begin == cb->end;
}
void outputBufferShift(OutputBuffer * cb)
{
assert(cb->begin != cb->end);
assert(cb->begin < buffered_chunks);
++cb->begin;
if (cb->begin >= buffered_chunks)
cb->begin = 0;
}
unsigned int outputBufferRelative(const OutputBuffer * cb, unsigned i)
{
if (i >= cb->begin)
return i - cb->begin;
else
return i + buffered_chunks - cb->begin;
}
unsigned availableOutputBuffer(const OutputBuffer * cb)
{
return outputBufferRelative(cb, cb->end);
}
int outputBufferAbsolute(const OutputBuffer * cb, unsigned relative)
{
unsigned i, max;
max = cb->end;
if (max < cb->begin)
max += buffered_chunks;
i = (unsigned)cb->begin + relative;
if (i >= max)
return -1;
if (i >= buffered_chunks)
i -= buffered_chunks;
return (int)i;
}
OutputBufferChunk * outputBufferGetChunk(const OutputBuffer * cb, unsigned i)
{
assert(i < buffered_chunks);
return &cb->chunks[i];
}
/**
* Return the tail chunk has room for additional data. If there is no
* room in the queue, this function blocks until the player thread has
* finished playing its current chunk.
*
* @return the positive index of the new chunk; OUTPUT_BUFFER_DC_SEEK
* if another thread requested seeking; OUTPUT_BUFFER_DC_STOP if
* another thread requested stopping the decoder.
*/
static int tailChunk(OutputBuffer * cb, InputStream * inStream,
DecoderControl * dc, int seekable,
float data_time, mpd_uint16 bitRate)
{
unsigned int next;
OutputBufferChunk *chunk;
if (currentChunk == cb->end)
return currentChunk;
next = cb->end + 1;
if (next >= buffered_chunks) {
next = 0;
}
while (cb->begin == next && !dc->stop) {
if (dc->seek) {
if (seekable) {
return OUTPUT_BUFFER_DC_SEEK;
} else {
dc->seekError = 1;
dc->seek = 0;
decoder_wakeup_player();
}
}
if (!inStream ||
bufferInputStream(inStream) <= 0) {
decoder_sleep();
}
}
if (dc->stop)
return OUTPUT_BUFFER_DC_STOP;
currentChunk = cb->end;
chunk = outputBufferGetChunk(cb, currentChunk);
chunk->chunkSize = 0;
chunk->bitRate = bitRate;
chunk->times = data_time;
return currentChunk;
}
int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
DecoderControl * dc, int seekable, void *dataIn,
size_t dataInLen, float data_time, mpd_uint16 bitRate,
ReplayGainInfo * replayGainInfo)
{
mpd_uint16 dataToSend;
mpd_uint16 chunkLeft;
char *data;
size_t datalen;
static char *convBuffer;
static size_t convBufferLen;
if (cmpAudioFormat(&(cb->audioFormat), &(dc->audioFormat)) == 0) {
data = dataIn;
datalen = dataInLen;
} else {
datalen = pcm_sizeOfConvBuffer(&(dc->audioFormat), dataInLen,
&(cb->audioFormat));
if (datalen > convBufferLen) {
if (convBuffer != NULL)
free(convBuffer);
convBuffer = xmalloc(datalen);
convBufferLen = datalen;
}
data = convBuffer;
datalen = pcm_convertAudioFormat(&(dc->audioFormat), dataIn,
dataInLen, &(cb->audioFormat),
data, &(cb->convState));
}
if (replayGainInfo && (replayGainState != REPLAYGAIN_OFF))
doReplayGain(replayGainInfo, data, datalen, &cb->audioFormat);
else if (normalizationEnabled)
normalizeData(data, datalen, &cb->audioFormat);
while (datalen) {
OutputBufferChunk *chunk;
int chunk_index = tailChunk(cb, inStream,
dc, seekable,
data_time, bitRate);
if (chunk_index < 0)
return chunk_index;
chunk = outputBufferGetChunk(cb, chunk_index);
chunkLeft = sizeof(chunk->data) - chunk->chunkSize;
dataToSend = datalen > chunkLeft ? chunkLeft : datalen;
memcpy(chunk->data + chunk->chunkSize, data, dataToSend);
chunk->chunkSize += dataToSend;
datalen -= dataToSend;
data += dataToSend;
if (chunk->chunkSize == sizeof(chunk->data)) {
flushOutputBuffer(cb);
}
}
Initial cut of fork() => pthreads() for decoder and player 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
2008-04-12 06:08:00 +02:00
decoder_wakeup_player();
return 0;
}