Re-tagging 0.13.0 release to fix a couple of bugs with the tarball.

git-svn-id: https://svn.musicpd.org/mpd/tags/release-0.13.0@6325 09075e82-0dd4-0310-85a5-a0d7c8717e4f

8 files changed, 3203 insertions, 0 deletions

diff --git a/trunk/src/audioOutputs/audioOutput_alsa.c b/trunk/src/audioOutputs/audioOutput_alsa.c
new file mode 100644
index 000000000..3ade3df46
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_alsa.c
@@ -0,0 +1,427 @@
+/* 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 "../audioOutput.h"
+
+#include <stdlib.h>
+
+#ifdef HAVE_ALSA
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+#define MPD_ALSA_BUFFER_TIME_US 500000
+/* the default period time of xmms is 50 ms, so let's use that as well.
+ * a user can tweak this parameter via the "period_time" config parameter.
+ */
+#define MPD_ALSA_PERIOD_TIME_US 50000
+#define MPD_ALSA_RETRY_NR 5
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <alsa/asoundlib.h>
+
+typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
+					snd_pcm_uframes_t size);
+
+typedef struct _AlsaData {
+	char *device;
+	snd_pcm_t *pcmHandle;
+	alsa_writei_t *writei;
+	unsigned int buffer_time;
+	unsigned int period_time;
+	int sampleSize;
+	int useMmap;
+	int canPause;
+	int canResume;
+} AlsaData;
+
+static AlsaData *newAlsaData(void)
+{
+	AlsaData *ret = xmalloc(sizeof(AlsaData));
+
+	ret->device = NULL;
+	ret->pcmHandle = NULL;
+	ret->writei = snd_pcm_writei;
+	ret->useMmap = 0;
+	ret->buffer_time = MPD_ALSA_BUFFER_TIME_US;
+	ret->period_time = MPD_ALSA_PERIOD_TIME_US;
+
+	return ret;
+}
+
+static void freeAlsaData(AlsaData * ad)
+{
+	if (ad->device)
+		free(ad->device);
+
+	free(ad);
+}
+
+static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+	AlsaData *ad = newAlsaData();
+
+	if (param) {
+		BlockParam *bp = getBlockParam(param, "device");
+		ad->device = bp ? xstrdup(bp->value) : xstrdup("default");
+
+		if ((bp = getBlockParam(param, "use_mmap")) &&
+		    !strcasecmp(bp->value, "yes"))
+			ad->useMmap = 1;
+		if ((bp = getBlockParam(param, "buffer_time")))
+			ad->buffer_time = atoi(bp->value);
+		if ((bp = getBlockParam(param, "period_time")))
+			ad->period_time = atoi(bp->value);
+	} else
+		ad->device = xstrdup("default");
+	audioOutput->data = ad;
+
+	return 0;
+}
+
+static void alsa_finishDriver(AudioOutput * audioOutput)
+{
+	AlsaData *ad = audioOutput->data;
+
+	freeAlsaData(ad);
+}
+
+static int alsa_testDefault(void)
+{
+	snd_pcm_t *handle;
+
+	int ret = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK,
+			       SND_PCM_NONBLOCK);
+	snd_config_update_free_global();
+
+	if (ret) {
+		WARNING("Error opening default alsa device: %s\n",
+			snd_strerror(-ret));
+		return -1;
+	} else
+		snd_pcm_close(handle);
+
+	return 0;
+}
+
+static int alsa_openDevice(AudioOutput * audioOutput)
+{
+	AlsaData *ad = audioOutput->data;
+	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+	snd_pcm_format_t bitformat;
+	snd_pcm_hw_params_t *hwparams;
+	snd_pcm_sw_params_t *swparams;
+	unsigned int sampleRate = audioFormat->sampleRate;
+	unsigned int channels = audioFormat->channels;
+	snd_pcm_uframes_t alsa_buffer_size;
+	snd_pcm_uframes_t alsa_period_size;
+	int err;
+	const char *cmd = NULL;
+	int retry = MPD_ALSA_RETRY_NR;
+	unsigned int period_time, period_time_ro;
+	unsigned int buffer_time;
+
+	switch (audioFormat->bits) {
+	case 8:
+		bitformat = SND_PCM_FORMAT_S8;
+		break;
+	case 16:
+		bitformat = SND_PCM_FORMAT_S16;
+		break;
+	case 24:
+		bitformat = SND_PCM_FORMAT_S24;
+		break;
+	case 32:
+		bitformat = SND_PCM_FORMAT_S32;
+		break;
+	default:
+		ERROR("ALSA device \"%s\" doesn't support %i bit audio\n",
+		      ad->device, audioFormat->bits);
+		return -1;
+	}
+
+	err = snd_pcm_open(&ad->pcmHandle, ad->device,
+			   SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+	snd_config_update_free_global();
+	if (err < 0) {
+		ad->pcmHandle = NULL;
+		goto error;
+	}
+
+	cmd = "snd_pcm_nonblock";
+	err = snd_pcm_nonblock(ad->pcmHandle, 0);
+	if (err < 0)
+		goto error;
+
+	period_time_ro = period_time = ad->period_time;
+configure_hw:
+	/* configure HW params */
+	snd_pcm_hw_params_alloca(&hwparams);
+
+	cmd = "snd_pcm_hw_params_any";
+	err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
+	if (err < 0)
+		goto error;
+
+	if (ad->useMmap) {
+		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
+						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
+		if (err < 0) {
+			ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
+			      " %s\n", ad->device, snd_strerror(-err));
+			ERROR("Falling back to direct write mode\n");
+			ad->useMmap = 0;
+		} else
+			ad->writei = snd_pcm_mmap_writei;
+	}
+
+	if (!ad->useMmap) {
+		cmd = "snd_pcm_hw_params_set_access";
+		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
+						   SND_PCM_ACCESS_RW_INTERLEAVED);
+		if (err < 0)
+			goto error;
+		ad->writei = snd_pcm_writei;
+	}
+
+	err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
+	if (err < 0) {
+		ERROR("ALSA device \"%s\" does not support %i bit audio: "
+		      "%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
+		goto fail;
+	}
+
+	err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
+						  &channels);
+	if (err < 0) {
+		ERROR("ALSA device \"%s\" does not support %i channels: "
+		      "%s\n", ad->device, (int)audioFormat->channels,
+		      snd_strerror(-err));
+		goto fail;
+	}
+	audioFormat->channels = channels;
+
+	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
+					      &sampleRate, NULL);
+	if (err < 0 || sampleRate == 0) {
+		ERROR("ALSA device \"%s\" does not support %i Hz audio\n",
+		      ad->device, (int)audioFormat->sampleRate);
+		goto fail;
+	}
+	audioFormat->sampleRate = sampleRate;
+
+	buffer_time = ad->buffer_time;
+	cmd = "snd_pcm_hw_params_set_buffer_time_near";
+	err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
+						     &buffer_time, NULL);
+	if (err < 0)
+		goto error;
+
+	period_time = period_time_ro;
+	cmd = "snd_pcm_hw_params_set_period_time_near";
+	err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
+						     &period_time, NULL);
+	if (err < 0)
+		goto error;
+
+	cmd = "snd_pcm_hw_params";
+	err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
+	if (err == -EPIPE && --retry > 0) {
+		period_time_ro = period_time_ro >> 1;
+		goto configure_hw;
+	} else if (err < 0)
+		goto error;
+	if (retry != MPD_ALSA_RETRY_NR)
+		DEBUG("ALSA period_time set to %d\n", period_time);
+
+	cmd = "snd_pcm_hw_params_get_buffer_size";
+	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
+	if (err < 0)
+		goto error;
+
+	cmd = "snd_pcm_hw_params_get_period_size";
+	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
+						NULL);
+	if (err < 0)
+		goto error;
+
+	ad->canPause = snd_pcm_hw_params_can_pause(hwparams);
+	ad->canResume = snd_pcm_hw_params_can_resume(hwparams);
+
+	/* configure SW params */
+	snd_pcm_sw_params_alloca(&swparams);
+
+	cmd = "snd_pcm_sw_params_current";
+	err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
+	if (err < 0)
+		goto error;
+
+	cmd = "snd_pcm_sw_params_set_start_threshold";
+	err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
+						    alsa_buffer_size -
+						    alsa_period_size);
+	if (err < 0)
+		goto error;
+
+	cmd = "snd_pcm_sw_params_set_avail_min";
+	err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
+					      alsa_period_size);
+	if (err < 0)
+		goto error;
+
+	cmd = "snd_pcm_sw_params_set_xfer_align";
+	err = snd_pcm_sw_params_set_xfer_align(ad->pcmHandle, swparams, 1);
+	if (err < 0)
+		goto error;
+
+	cmd = "snd_pcm_sw_params";
+	err = snd_pcm_sw_params(ad->pcmHandle, swparams);
+	if (err < 0)
+		goto error;
+
+	ad->sampleSize = (audioFormat->bits / 8) * audioFormat->channels;
+
+	audioOutput->open = 1;
+
+	DEBUG("alsa device \"%s\" will be playing %i bit, %i channel audio at "
+	      "%i Hz\n", ad->device, (int)audioFormat->bits,
+	      channels, sampleRate);
+
+	return 0;
+
+error:
+	if (cmd) {
+		ERROR("Error opening alsa device \"%s\" (%s): %s\n",
+		      ad->device, cmd, snd_strerror(-err));
+	} else {
+		ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
+		      snd_strerror(-err));
+	}
+fail:
+	if (ad->pcmHandle)
+		snd_pcm_close(ad->pcmHandle);
+	ad->pcmHandle = NULL;
+	audioOutput->open = 0;
+	return -1;
+}
+
+static int alsa_errorRecovery(AlsaData * ad, int err)
+{
+	if (err == -EPIPE) {
+		DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
+	} else if (err == -ESTRPIPE) {
+		DEBUG("alsa device \"%s\" was suspended\n", ad->device);
+	}
+
+	switch (snd_pcm_state(ad->pcmHandle)) {
+	case SND_PCM_STATE_PAUSED:
+		err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
+		break;
+	case SND_PCM_STATE_SUSPENDED:
+		err = ad->canResume ?
+		    snd_pcm_resume(ad->pcmHandle) :
+		    snd_pcm_prepare(ad->pcmHandle);
+		break;
+	case SND_PCM_STATE_SETUP:
+	case SND_PCM_STATE_XRUN:
+		err = snd_pcm_prepare(ad->pcmHandle);
+		break;
+	case SND_PCM_STATE_DISCONNECTED:
+		/* so alsa_closeDevice won't try to drain: */
+		snd_pcm_close(ad->pcmHandle);
+		ad->pcmHandle = NULL;
+		break;
+	default:
+		/* unknown state, do nothing */
+		break;
+	}
+
+	return err;
+}
+
+static void alsa_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	AlsaData *ad = audioOutput->data;
+
+	alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
+}
+
+static void alsa_closeDevice(AudioOutput * audioOutput)
+{
+	AlsaData *ad = audioOutput->data;
+
+	if (ad->pcmHandle) {
+		snd_pcm_drain(ad->pcmHandle);
+		snd_pcm_close(ad->pcmHandle);
+		ad->pcmHandle = NULL;
+	}
+
+	audioOutput->open = 0;
+}
+
+static int alsa_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+	AlsaData *ad = audioOutput->data;
+	int ret;
+
+	size /= ad->sampleSize;
+
+	while (size > 0) {
+		ret = ad->writei(ad->pcmHandle, playChunk, size);
+
+		if (ret == -EAGAIN || ret == -EINTR)
+			continue;
+
+		if (ret < 0) {
+			if (alsa_errorRecovery(ad, ret) < 0) {
+				ERROR("closing alsa device \"%s\" due to write "
+				      "error: %s\n", ad->device,
+				      snd_strerror(-errno));
+				alsa_closeDevice(audioOutput);
+				return -1;
+			}
+			continue;
+		}
+
+		playChunk += ret * ad->sampleSize;
+		size -= ret;
+	}
+
+	return 0;
+}
+
+AudioOutputPlugin alsaPlugin = {
+	"alsa",
+	alsa_testDefault,
+	alsa_initDriver,
+	alsa_finishDriver,
+	alsa_openDevice,
+	alsa_playAudio,
+	alsa_dropBufferedAudio,
+	alsa_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else /* HAVE ALSA */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
+#endif /* HAVE_ALSA */
diff --git a/trunk/src/audioOutputs/audioOutput_ao.c b/trunk/src/audioOutputs/audioOutput_ao.c
new file mode 100644
index 000000000..a7f437ef4
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_ao.c
@@ -0,0 +1,246 @@
+/* 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 "../audioOutput.h"
+
+#ifdef HAVE_AO
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <ao/ao.h>
+
+static int driverInitCount;
+
+typedef struct _AoData {
+	int writeSize;
+	int driverId;
+	ao_option *options;
+	ao_device *device;
+} AoData;
+
+static AoData *newAoData(void)
+{
+	AoData *ret = xmalloc(sizeof(AoData));
+	ret->device = NULL;
+	ret->options = NULL;
+
+	return ret;
+}
+
+static void audioOutputAo_error(void)
+{
+	if (errno == AO_ENOTLIVE) {
+		ERROR("not a live ao device\n");
+	} else if (errno == AO_EOPENDEVICE) {
+		ERROR("not able to open audio device\n");
+	} else if (errno == AO_EBADOPTION) {
+		ERROR("bad driver option\n");
+	}
+}
+
+static int audioOutputAo_initDriver(AudioOutput * audioOutput,
+				    ConfigParam * param)
+{
+	ao_info *ai;
+	char *dup;
+	char *stk1;
+	char *stk2;
+	char *n1;
+	char *key;
+	char *value;
+	char *test;
+	AoData *ad = newAoData();
+	BlockParam *blockParam;
+
+	audioOutput->data = ad;
+
+	if ((blockParam = getBlockParam(param, "write_size"))) {
+		ad->writeSize = strtol(blockParam->value, &test, 10);
+		if (*test != '\0') {
+			FATAL("\"%s\" is not a valid write size at line %i\n",
+			      blockParam->value, blockParam->line);
+		}
+	} else
+		ad->writeSize = 1024;
+
+	if (driverInitCount == 0) {
+		ao_initialize();
+	}
+	driverInitCount++;
+
+	blockParam = getBlockParam(param, "driver");
+
+	if (!blockParam || 0 == strcmp(blockParam->value, "default")) {
+		ad->driverId = ao_default_driver_id();
+	} else if ((ad->driverId = ao_driver_id(blockParam->value)) < 0) {
+		FATAL("\"%s\" is not a valid ao driver at line %i\n",
+		      blockParam->value, blockParam->line);
+	}
+
+	if ((ai = ao_driver_info(ad->driverId)) == NULL) {
+		FATAL("problems getting driver info for device defined at line %i\n"
+		      "you may not have permission to the audio device\n", param->line);
+	}
+
+	DEBUG("using ao driver \"%s\" for \"%s\"\n", ai->short_name,
+	      audioOutput->name);
+
+	blockParam = getBlockParam(param, "options");
+
+	if (blockParam) {
+		dup = xstrdup(blockParam->value);
+	} else
+		dup = xstrdup("");
+
+	if (strlen(dup)) {
+		stk1 = NULL;
+		n1 = strtok_r(dup, ";", &stk1);
+		while (n1) {
+			stk2 = NULL;
+			key = strtok_r(n1, "=", &stk2);
+			if (!key)
+				FATAL("problems parsing options \"%s\"\n", n1);
+			/*found = 0;
+			   for(i=0;i<ai->option_count;i++) {
+			   if(strcmp(ai->options[i],key)==0) {
+			   found = 1;
+			   break;
+			   }
+			   }
+			   if(!found) {
+			   FATAL("\"%s\" is not an option for "
+			   "\"%s\" ao driver\n",key,
+			   ai->short_name);
+			   } */
+			value = strtok_r(NULL, "", &stk2);
+			if (!value)
+				FATAL("problems parsing options \"%s\"\n", n1);
+			ao_append_option(&ad->options, key, value);
+			n1 = strtok_r(NULL, ";", &stk1);
+		}
+	}
+	free(dup);
+
+	return 0;
+}
+
+static void freeAoData(AoData * ad)
+{
+	ao_free_options(ad->options);
+	free(ad);
+}
+
+static void audioOutputAo_finishDriver(AudioOutput * audioOutput)
+{
+	AoData *ad = (AoData *) audioOutput->data;
+	freeAoData(ad);
+
+	driverInitCount--;
+
+	if (driverInitCount == 0)
+		ao_shutdown();
+}
+
+static void audioOutputAo_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	/* not supported by libao */
+}
+
+static void audioOutputAo_closeDevice(AudioOutput * audioOutput)
+{
+	AoData *ad = (AoData *) audioOutput->data;
+
+	if (ad->device) {
+		ao_close(ad->device);
+		ad->device = NULL;
+	}
+
+	audioOutput->open = 0;
+}
+
+static int audioOutputAo_openDevice(AudioOutput * audioOutput)
+{
+	ao_sample_format format;
+	AoData *ad = (AoData *) audioOutput->data;
+
+	if (ad->device) {
+		audioOutputAo_closeDevice(audioOutput);
+	}
+
+	format.bits = audioOutput->outAudioFormat.bits;
+	format.rate = audioOutput->outAudioFormat.sampleRate;
+	format.byte_format = AO_FMT_NATIVE;
+	format.channels = audioOutput->outAudioFormat.channels;
+
+	ad->device = ao_open_live(ad->driverId, &format, ad->options);
+
+	if (ad->device == NULL)
+		return -1;
+
+	audioOutput->open = 1;
+
+	return 0;
+}
+
+static int audioOutputAo_play(AudioOutput * audioOutput, char *playChunk,
+			      int size)
+{
+	int send;
+	AoData *ad = (AoData *) audioOutput->data;
+
+	if (ad->device == NULL)
+		return -1;
+
+	while (size > 0) {
+		send = ad->writeSize > size ? size : ad->writeSize;
+
+		if (ao_play(ad->device, playChunk, send) == 0) {
+			audioOutputAo_error();
+			ERROR("closing audio device due to write error\n");
+			audioOutputAo_closeDevice(audioOutput);
+			return -1;
+		}
+
+		playChunk += send;
+		size -= send;
+	}
+
+	return 0;
+}
+
+AudioOutputPlugin aoPlugin = {
+	"ao",
+	NULL,
+	audioOutputAo_initDriver,
+	audioOutputAo_finishDriver,
+	audioOutputAo_openDevice,
+	audioOutputAo_play,
+	audioOutputAo_dropBufferedAudio,
+	audioOutputAo_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else
+
+#include <stdio.h>
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
+#endif
diff --git a/trunk/src/audioOutputs/audioOutput_jack.c b/trunk/src/audioOutputs/audioOutput_jack.c
new file mode 100644
index 000000000..1fdfaf4bb
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_jack.c
@@ -0,0 +1,440 @@
+/* jack plug in for the Music Player Daemon (MPD)
+ * (c)2006 by anarch(anarchsss@gmail.com)
+ *
+ * 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 "../audioOutput.h"
+
+#ifdef HAVE_JACK
+
+#include <stdlib.h>
+#include <errno.h>
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+#include <pthread.h>
+
+#include <jack/jack.h>
+#include <jack/types.h>
+#include <jack/ringbuffer.h>
+
+pthread_mutex_t play_audio_lock = PTHREAD_MUTEX_INITIALIZER;
+pthread_cond_t  play_audio = PTHREAD_COND_INITIALIZER;
+
+/*#include "dmalloc.h"*/
+
+#define MIN(a, b) ((a) < (b) ? (a) : (b))
+/*#define SAMPLE_SIZE  sizeof(jack_default_audio_sample_t);*/
+
+
+static char *name = "mpd";
+static char *output_ports[2];
+static int ringbuf_sz = 32768;
+size_t sample_size = sizeof(jack_default_audio_sample_t);
+
+typedef struct _JackData {
+	jack_port_t *ports[2];
+	jack_client_t *client;
+	jack_ringbuffer_t *ringbuffer[2];
+	int bps;
+	int shutdown;
+} JackData;
+
+/*JackData *jd = NULL;*/
+
+static JackData *newJackData(void)
+{
+	JackData *ret;
+	ret = xcalloc(sizeof(JackData), 1);
+
+	return ret;
+}
+
+static void freeJackData(AudioOutput *audioOutput)
+{
+	JackData *jd = audioOutput->data;
+	if (jd) {
+		if (jd->ringbuffer[0])
+			jack_ringbuffer_free(jd->ringbuffer[0]);
+		if (jd->ringbuffer[1])
+			jack_ringbuffer_free(jd->ringbuffer[1]);
+		free(jd);
+		audioOutput->data = NULL;
+	}
+}
+
+static void jack_finishDriver(AudioOutput *audioOutput)
+{
+	JackData *jd = audioOutput->data;
+	int i;
+
+	if ( jd && jd->client ) {
+		jack_deactivate(jd->client);
+		jack_client_close(jd->client);
+	}
+	DEBUG("disconnect_jack (pid=%d)\n", getpid ());
+
+ 	if ( strcmp(name, "mpd") ) {
+ 		free(name);
+ 		name = "mpd";
+ 	}
+
+	for ( i = ARRAY_SIZE(output_ports); --i >= 0; ) {
+		if (!output_ports[i])
+ 			continue;
+ 		free(output_ports[i]);
+ 		output_ports[i] = NULL;
+ 	}
+
+	freeJackData(audioOutput);
+}
+
+static int srate(jack_nframes_t rate, void *data)
+{
+	JackData *jd = (JackData *) ((AudioOutput*) data)->data;
+ 	AudioFormat *audioFormat = &(((AudioOutput*) data)->outAudioFormat);
+
+ 	audioFormat->sampleRate = (int)jack_get_sample_rate(jd->client);
+
+	return 0;
+}
+
+static int process(jack_nframes_t nframes, void *arg)
+{
+	size_t i;
+	JackData *jd = (JackData *) arg;
+	jack_default_audio_sample_t *out[2];
+	size_t avail_data, avail_frames;
+
+	if ( nframes <= 0 )
+		return 0;
+
+	out[0] = jack_port_get_buffer(jd->ports[0], nframes);
+	out[1] = jack_port_get_buffer(jd->ports[1], nframes);
+
+	while ( nframes ) {
+		avail_data = jack_ringbuffer_read_space(jd->ringbuffer[1]);
+
+		if ( avail_data > 0 ) {
+		    avail_frames = avail_data / sample_size;
+
+		    if (avail_frames > nframes) {
+			avail_frames = nframes;
+			avail_data = nframes*sample_size;
+		    }
+
+		    jack_ringbuffer_read(jd->ringbuffer[0], (char *)out[0],
+					 avail_data);
+		    jack_ringbuffer_read(jd->ringbuffer[1], (char *)out[1],
+					 avail_data);
+
+		    nframes -= avail_frames;
+		    out[0] += avail_data;
+		    out[1] += avail_data;
+		} else {
+		    for (i = 0; i < nframes; i++)
+  			out[0][i] = out[1][i] = 0.0;
+		    nframes = 0;
+		}
+
+		if (pthread_mutex_trylock (&play_audio_lock) == 0) {
+			pthread_cond_signal (&play_audio);
+			pthread_mutex_unlock (&play_audio_lock);
+		}
+	}
+
+
+	/*DEBUG("process (pid=%d)\n", getpid());*/
+	return 0;
+}
+
+static void shutdown_callback(void *arg)
+{
+	JackData *jd = (JackData *) arg;
+	jd->shutdown = 1;
+}
+
+static void set_audioformat(AudioOutput *audioOutput)
+{
+	JackData *jd = audioOutput->data;
+	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+
+	audioFormat->sampleRate = (int) jack_get_sample_rate(jd->client);
+	DEBUG("samplerate = %d\n", audioFormat->sampleRate);
+	audioFormat->channels = 2;
+	audioFormat->bits = 16;
+	jd->bps = audioFormat->channels
+		* sizeof(jack_default_audio_sample_t)
+		* audioFormat->sampleRate;
+}
+
+static void error_callback(const char *msg)
+{
+	ERROR("jack: %s\n", msg);
+}
+
+static int jack_initDriver(AudioOutput *audioOutput, ConfigParam *param)
+{
+	BlockParam *bp;
+	char *endptr;
+	int val;
+ 	char *cp = NULL;
+
+	DEBUG("jack_initDriver (pid=%d)\n", getpid());
+	if ( ! param ) return 0;
+
+	if ( (bp = getBlockParam(param, "ports")) ) {
+		DEBUG("output_ports=%s\n", bp->value);
+
+		if (!(cp = strchr(bp->value, ',')))
+			FATAL("expected comma and a second value for '%s' "
+			      "at line %d: %s\n",
+			      bp->name, bp->line, bp->value);
+
+		*cp = '\0';
+		output_ports[0] = xstrdup(bp->value);
+		*cp++ = ',';
+
+		if (!*cp)
+			FATAL("expected a second value for '%s' at line %d: "
+			      "%s\n", bp->name, bp->line, bp->value);
+
+		output_ports[1] = xstrdup(cp);
+
+		if (strchr(cp,','))
+			FATAL("Only %d values are supported for '%s' "
+			      "at line %d\n", (int)ARRAY_SIZE(output_ports),
+			      bp->name, bp->line);
+	}
+
+	if ( (bp = getBlockParam(param, "ringbuffer_size")) ) {
+		errno = 0;
+		val = strtol(bp->value, &endptr, 10);
+
+		if ( errno == 0 && endptr != bp->value) {
+			ringbuf_sz = val < 32768 ? 32768 : val;
+			DEBUG("ringbuffer_size=%d\n", ringbuf_sz);
+		} else {
+			FATAL("%s is not a number; ringbuf_size=%d\n",
+			      bp->value, ringbuf_sz);
+		}
+	}
+
+	if ( (bp = getBlockParam(param, "name"))
+	     && (strcmp(bp->value, "mpd") != 0) ) {
+		name = xstrdup(bp->value);
+		DEBUG("name=%s\n", name);
+	}
+
+ 	return 0;
+}
+
+static int jack_testDefault(void)
+{
+	return 0;
+}
+
+static int connect_jack(AudioOutput *audioOutput)
+{
+	JackData *jd = audioOutput->data;
+	char **jports;
+	char *port_name;
+
+	if ( (jd->client = jack_client_new(name)) == NULL ) {
+		ERROR("jack server not running?\n");
+		freeJackData(audioOutput);
+		return -1;
+	}
+
+	jack_set_error_function(error_callback);
+	jack_set_process_callback(jd->client, process, (void *)jd);
+	jack_set_sample_rate_callback(jd->client, (JackProcessCallback)srate,
+				      (void *)audioOutput);
+	jack_on_shutdown(jd->client, shutdown_callback, (void *)jd);
+
+	if ( jack_activate(jd->client) ) {
+		ERROR("cannot activate client");
+		freeJackData(audioOutput);
+		return -1;
+	}
+
+	jd->ports[0] = jack_port_register(jd->client, "left",
+					  JACK_DEFAULT_AUDIO_TYPE,
+					  JackPortIsOutput, 0);
+	if ( !jd->ports[0] ) {
+		ERROR("Cannot register left output port.\n");
+		freeJackData(audioOutput);
+		return -1;
+	}
+
+	jd->ports[1] = jack_port_register(jd->client, "right",
+					  JACK_DEFAULT_AUDIO_TYPE,
+					  JackPortIsOutput, 0);
+	if ( !jd->ports[1] ) {
+		ERROR("Cannot register right output port.\n");
+		freeJackData(audioOutput);
+		return -1;
+	}
+
+	/*  hay que buscar que hay  */
+	if ( !output_ports[1]
+	     && (jports = (char **)jack_get_ports(jd->client, NULL, NULL,
+							JackPortIsPhysical|
+							JackPortIsInput)) ) {
+		output_ports[0] = jports[0];
+		output_ports[1] = jports[1] ? jports[1] : jports[0];
+		DEBUG("output_ports: %s %s\n", output_ports[0], output_ports[1]);
+		free(jports);
+	}
+
+	if ( output_ports[1] ) {
+		jd->ringbuffer[0] = jack_ringbuffer_create(ringbuf_sz);
+		jd->ringbuffer[1] = jack_ringbuffer_create(ringbuf_sz);
+		memset(jd->ringbuffer[0]->buf, 0, jd->ringbuffer[0]->size);
+		memset(jd->ringbuffer[1]->buf, 0, jd->ringbuffer[1]->size);
+
+		port_name = xmalloc(sizeof(char)*(7+strlen(name)));
+
+		sprintf(port_name, "%s:left", name);
+		if ( (jack_connect(jd->client, port_name,
+				   output_ports[0])) != 0 ) {
+			ERROR("%s is not a valid Jack Client / Port ",
+			      output_ports[0]);
+			freeJackData(audioOutput);
+			free(port_name);
+			return -1;
+		}
+		sprintf(port_name, "%s:right", name);
+		if ( (jack_connect(jd->client, port_name,
+				   output_ports[1])) != 0 ) {
+			ERROR("%s is not a valid Jack Client / Port ",
+			      output_ports[1]);
+			freeJackData(audioOutput);
+			free(port_name);
+			return -1;
+		}
+		free(port_name);
+	}
+
+	DEBUG("connect_jack (pid=%d)\n", getpid());
+	return 1;
+}
+
+static int jack_openDevice(AudioOutput *audioOutput)
+{
+	JackData *jd = audioOutput->data;
+
+	if ( !jd ) {
+		DEBUG("connect!\n");
+		jd = newJackData();
+		audioOutput->data = jd;
+
+		if (connect_jack(audioOutput) < 0) {
+			freeJackData(audioOutput);
+			audioOutput->open = 0;
+			return -1;
+		}
+	}
+
+	set_audioformat(audioOutput);
+	audioOutput->open = 1;
+
+	DEBUG("jack_openDevice (pid=%d)!\n", getpid ());
+	return 0;
+}
+
+
+static void jack_closeDevice(AudioOutput * audioOutput)
+{
+	/*jack_finishDriver(audioOutput);*/
+	audioOutput->open = 0;
+	DEBUG("jack_closeDevice (pid=%d)\n", getpid());
+}
+
+static void jack_dropBufferedAudio (AudioOutput * audioOutput)
+{
+}
+
+static int jack_playAudio(AudioOutput * audioOutput, char *buff, int size)
+{
+	JackData *jd = audioOutput->data;
+	size_t space;
+	int i;
+	short *buffer = (short *) buff;
+	jack_default_audio_sample_t sample;
+	size_t samples = size/4;
+
+	/*DEBUG("jack_playAudio: (pid=%d)!\n", getpid());*/
+
+	if ( jd->shutdown ) {
+		ERROR("Refusing to play, because there is no client thread.\n");
+		freeJackData(audioOutput);
+		audioOutput->open = 0;
+		return 0;
+	}
+
+	while ( samples && !jd->shutdown ) {
+
+ 		if ( (space = jack_ringbuffer_write_space(jd->ringbuffer[0]))
+ 		     >= samples*sample_size ) {
+
+			/*space = MIN(space, samples*sample_size);*/
+			/*space = samples*sample_size;*/
+
+			/*for(i=0; i<space/sample_size; i++) {*/
+			for(i=0; i<samples; i++) {
+				sample = (jack_default_audio_sample_t) *(buffer++)/32768.0;
+
+				jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
+						      sample_size);
+
+				sample = (jack_default_audio_sample_t) *(buffer++)/32768.0;
+
+				jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
+						      sample_size);
+
+				/*samples--;*/
+			}
+			samples=0;
+
+ 		} else {
+			pthread_mutex_lock(&play_audio_lock);
+			pthread_cond_wait(&play_audio, &play_audio_lock);
+			pthread_mutex_unlock(&play_audio_lock);
+		}
+
+	}
+	return 0;
+}
+
+AudioOutputPlugin jackPlugin = {
+	"jack",
+	jack_testDefault,
+	jack_initDriver,
+	jack_finishDriver,
+	jack_openDevice,
+	jack_playAudio,
+	jack_dropBufferedAudio,
+	jack_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else /* HAVE JACK */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
+
+#endif /* HAVE_JACK */
diff --git a/trunk/src/audioOutputs/audioOutput_mvp.c b/trunk/src/audioOutputs/audioOutput_mvp.c
new file mode 100644
index 000000000..ea365c657
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_mvp.c
@@ -0,0 +1,284 @@
+/* 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
+ *
+ * Media MVP audio output based on code from MVPMC project:
+ * http://mvpmc.sourceforge.net/
+ *
+ * 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 "../audioOutput.h"
+
+#include <stdlib.h>
+
+#ifdef HAVE_MVP
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+
+typedef struct {
+	unsigned long dsp_status;
+	unsigned long stream_decode_type;
+	unsigned long sample_rate;
+	unsigned long bit_rate;
+	unsigned long raw[64 / sizeof(unsigned long)];
+} aud_status_t;
+
+#define MVP_SET_AUD_STOP		_IOW('a',1,int)
+#define MVP_SET_AUD_PLAY		_IOW('a',2,int)
+#define MVP_SET_AUD_PAUSE	_IOW('a',3,int)
+#define MVP_SET_AUD_UNPAUSE	_IOW('a',4,int)
+#define MVP_SET_AUD_SRC		_IOW('a',5,int)
+#define MVP_SET_AUD_MUTE		_IOW('a',6,int)
+#define MVP_SET_AUD_BYPASS	_IOW('a',8,int)
+#define MVP_SET_AUD_CHANNEL	_IOW('a',9,int)
+#define MVP_GET_AUD_STATUS	_IOR('a',10,aud_status_t)
+#define MVP_SET_AUD_VOLUME	_IOW('a',13,int)
+#define MVP_GET_AUD_VOLUME	_IOR('a',14,int)
+#define MVP_SET_AUD_STREAMTYPE	_IOW('a',15,int)
+#define MVP_SET_AUD_FORMAT	_IOW('a',16,int)
+#define MVP_GET_AUD_SYNC		_IOR('a',21,pts_sync_data_t*)
+#define MVP_SET_AUD_STC		_IOW('a',22,long long int *)
+#define MVP_SET_AUD_SYNC		_IOW('a',23,int)
+#define MVP_SET_AUD_END_STREAM	_IOW('a',25,int)
+#define MVP_SET_AUD_RESET	_IOW('a',26,int)
+#define MVP_SET_AUD_DAC_CLK	_IOW('a',27,int)
+#define MVP_GET_AUD_REGS		_IOW('a',28,aud_ctl_regs_t*)
+
+typedef struct _MvpData {
+	int fd;
+} MvpData;
+
+static int pcmfrequencies[][3] = {
+	{9, 8000, 32000},
+	{10, 11025, 44100},
+	{11, 12000, 48000},
+	{1, 16000, 32000},
+	{2, 22050, 44100},
+	{3, 24000, 48000},
+	{5, 32000, 32000},
+	{0, 44100, 44100},
+	{7, 48000, 48000},
+	{13, 64000, 32000},
+	{14, 88200, 44100},
+	{15, 96000, 48000}
+};
+
+static int numfrequencies = sizeof(pcmfrequencies) / 12;
+
+static int mvp_testDefault(void)
+{
+	int fd;
+
+	fd = open("/dev/adec_pcm", O_WRONLY);
+
+	if (fd) {
+		close(fd);
+		return 0;
+	}
+
+	WARNING("Error opening PCM device \"/dev/adec_pcm\": %s\n",
+		strerror(errno));
+
+	return -1;
+}
+
+static int mvp_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+	MvpData *md = xmalloc(sizeof(MvpData));
+	md->fd = -1;
+	audioOutput->data = md;
+
+	return 0;
+}
+
+static void mvp_finishDriver(AudioOutput * audioOutput)
+{
+	MvpData *md = audioOutput->data;
+	free(md);
+}
+
+static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
+			    int big_endian, int bits)
+{
+	int iloop;
+	int mix[5];
+
+	if (channels == 1)
+		mix[0] = 1;
+	else if (channels == 2)
+		mix[0] = 0;
+	else
+		return -1;
+
+	/* 0,1=24bit(24) , 2,3=16bit */
+	if (bits == 16)
+		mix[1] = 2;
+	else if (bits == 24)
+		mix[1] = 0;
+	else
+		return -1;
+
+	mix[3] = 0;	/* stream type? */
+
+	if (big_endian == 1)
+		mix[4] = 1;
+	else if (big_endian == 0)
+		mix[4] = 0;
+	else
+		return -1;
+
+	/*
+	 * if there is an exact match for the frequency, use it.
+	 */
+	for (iloop = 0; iloop < numfrequencies; iloop++) {
+		if (rate == pcmfrequencies[iloop][1]) {
+			mix[2] = pcmfrequencies[iloop][0];
+			break;
+		}
+	}
+
+	if (iloop >= numfrequencies) {
+		ERROR("Can not find suitable output frequency for %ld\n", rate);
+		return -1;
+	}
+
+	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
+		ERROR("Can not set audio format\n");
+		return -1;
+	}
+
+	if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
+		ERROR("Can not set audio sync\n");
+		return -1;
+	}
+
+	if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
+		ERROR("Can not set audio play mode\n");
+		return -1;
+	}
+
+	return 0;
+}
+
+static int mvp_openDevice(AudioOutput * audioOutput)
+{
+	long long int stc = 0;
+	MvpData *md = audioOutput->data;
+	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+	int mix[5] = { 0, 2, 7, 1, 0 };
+
+	if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
+		ERROR("Error opening /dev/adec_pcm: %s\n", strerror(errno));
+		return -1;
+	}
+	if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
+		ERROR("Error setting audio source: %s\n", strerror(errno));
+		return -1;
+	}
+	if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
+		ERROR("Error setting audio streamtype: %s\n", strerror(errno));
+		return -1;
+	}
+	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
+		ERROR("Error setting audio format: %s\n", strerror(errno));
+		return -1;
+	}
+	ioctl(md->fd, MVP_SET_AUD_STC, &stc);
+	if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
+		ERROR("Error setting audio streamtype: %s\n", strerror(errno));
+		return -1;
+	}
+#ifdef WORDS_BIGENDIAN
+	mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 0,
+			 audioFormat->bits);
+#else
+	mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 1,
+			 audioFormat->bits);
+#endif
+	audioOutput->open = 1;
+	return 0;
+}
+
+static void mvp_closeDevice(AudioOutput * audioOutput)
+{
+	MvpData *md = audioOutput->data;
+	if (md->fd >= 0)
+		close(md->fd);
+	md->fd = -1;
+	audioOutput->open = 0;
+}
+
+static void mvp_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	MvpData *md = audioOutput->data;
+	if (md->fd >= 0) {
+		ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
+		close(md->fd);
+		md->fd = -1;
+		audioOutput->open = 0;
+	}
+}
+
+static int mvp_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+	MvpData *md = audioOutput->data;
+	int ret;
+
+	/* reopen the device since it was closed by dropBufferedAudio */
+	if (md->fd < 0)
+		mvp_openDevice(audioOutput);
+
+	while (size > 0) {
+		ret = write(md->fd, playChunk, size);
+		if (ret < 0) {
+			if (errno == EINTR)
+				continue;
+			ERROR("closing mvp PCM device due to write error: "
+			      "%s\n", strerror(errno));
+			mvp_closeDevice(audioOutput);
+			return -1;
+		}
+		playChunk += ret;
+		size -= ret;
+	}
+	return 0;
+}
+
+AudioOutputPlugin mvpPlugin = {
+	"mvp",
+	mvp_testDefault,
+	mvp_initDriver,
+	mvp_finishDriver,
+	mvp_openDevice,
+	mvp_playAudio,
+	mvp_dropBufferedAudio,
+	mvp_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else /* HAVE_MVP */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
+#endif /* HAVE_MVP */
diff --git a/trunk/src/audioOutputs/audioOutput_oss.c b/trunk/src/audioOutputs/audioOutput_oss.c
new file mode 100644
index 000000000..01293cbd1
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_oss.c
@@ -0,0 +1,575 @@
+/* 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
+ *
+ * OSS audio output (c) 2004, 2005, 2006, 2007 by Eric Wong <eric@petta-tech.com>
+ *                   and Warren Dukes <warren.dukes@gmail.com>
+ *
+ * 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 "../audioOutput.h"
+
+#include <stdlib.h>
+
+#ifdef HAVE_OSS
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+
+#if defined(__OpenBSD__) || defined(__NetBSD__)
+# include <soundcard.h>
+#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+# include <sys/soundcard.h>
+#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+
+#ifdef WORDS_BIGENDIAN
+# define	AFMT_S16_MPD	 AFMT_S16_BE
+#else
+# define	AFMT_S16_MPD	 AFMT_S16_LE
+#endif /* WORDS_BIGENDIAN */
+
+typedef struct _OssData {
+	int fd;
+	const char *device;
+	int channels;
+	int sampleRate;
+	int bitFormat;
+	int bits;
+	int *supported[3];
+	int numSupported[3];
+	int *unsupported[3];
+	int numUnsupported[3];
+} OssData;
+
+#define OSS_SUPPORTED		1
+#define OSS_UNSUPPORTED		0
+#define OSS_UNKNOWN		-1
+
+#define OSS_RATE		0
+#define OSS_CHANNELS		1
+#define OSS_BITS		2
+
+static int getIndexForParam(int param)
+{
+	int index = 0;
+
+	switch (param) {
+	case SNDCTL_DSP_SPEED:
+		index = OSS_RATE;
+		break;
+	case SNDCTL_DSP_CHANNELS:
+		index = OSS_CHANNELS;
+		break;
+	case SNDCTL_DSP_SAMPLESIZE:
+		index = OSS_BITS;
+		break;
+	}
+
+	return index;
+}
+
+static int findSupportedParam(OssData * od, int param, int val)
+{
+	int i;
+	int index = getIndexForParam(param);
+
+	for (i = 0; i < od->numSupported[index]; i++) {
+		if (od->supported[index][i] == val)
+			return 1;
+	}
+
+	return 0;
+}
+
+static int canConvert(int index, int val)
+{
+	switch (index) {
+	case OSS_BITS:
+		if (val != 16)
+			return 0;
+		break;
+	case OSS_CHANNELS:
+		if (val != 2)
+			return 0;
+		break;
+	}
+
+	return 1;
+}
+
+static int getSupportedParam(OssData * od, int param, int val)
+{
+	int i;
+	int index = getIndexForParam(param);
+	int ret = -1;
+	int least = val;
+	int diff;
+
+	for (i = 0; i < od->numSupported[index]; i++) {
+		diff = od->supported[index][i] - val;
+		if (diff < 0)
+			diff = -diff;
+		if (diff < least) {
+			if (!canConvert(index, od->supported[index][i])) {
+				continue;
+			}
+			least = diff;
+			ret = od->supported[index][i];
+		}
+	}
+
+	return ret;
+}
+
+static int findUnsupportedParam(OssData * od, int param, int val)
+{
+	int i;
+	int index = getIndexForParam(param);
+
+	for (i = 0; i < od->numUnsupported[index]; i++) {
+		if (od->unsupported[index][i] == val)
+			return 1;
+	}
+
+	return 0;
+}
+
+static void addSupportedParam(OssData * od, int param, int val)
+{
+	int index = getIndexForParam(param);
+
+	od->numSupported[index]++;
+	od->supported[index] = xrealloc(od->supported[index],
+				       od->numSupported[index] * sizeof(int));
+	od->supported[index][od->numSupported[index] - 1] = val;
+}
+
+static void addUnsupportedParam(OssData * od, int param, int val)
+{
+	int index = getIndexForParam(param);
+
+	od->numUnsupported[index]++;
+	od->unsupported[index] = xrealloc(od->unsupported[index],
+					 od->numUnsupported[index] *
+					 sizeof(int));
+	od->unsupported[index][od->numUnsupported[index] - 1] = val;
+}
+
+static void removeSupportedParam(OssData * od, int param, int val)
+{
+	int i = 0;
+	int j = 0;
+	int index = getIndexForParam(param);
+
+	for (i = 0; i < od->numSupported[index] - 1; i++) {
+		if (od->supported[index][i] == val)
+			j = 1;
+		od->supported[index][i] = od->supported[index][i + j];
+	}
+
+	od->numSupported[index]--;
+	od->supported[index] = xrealloc(od->supported[index],
+				       od->numSupported[index] * sizeof(int));
+}
+
+static void removeUnsupportedParam(OssData * od, int param, int val)
+{
+	int i = 0;
+	int j = 0;
+	int index = getIndexForParam(param);
+
+	for (i = 0; i < od->numUnsupported[index] - 1; i++) {
+		if (od->unsupported[index][i] == val)
+			j = 1;
+		od->unsupported[index][i] = od->unsupported[index][i + j];
+	}
+
+	od->numUnsupported[index]--;
+	od->unsupported[index] = xrealloc(od->unsupported[index],
+					 od->numUnsupported[index] *
+					 sizeof(int));
+}
+
+static int isSupportedParam(OssData * od, int param, int val)
+{
+	if (findSupportedParam(od, param, val))
+		return OSS_SUPPORTED;
+	if (findUnsupportedParam(od, param, val))
+		return OSS_UNSUPPORTED;
+	return OSS_UNKNOWN;
+}
+
+static void supportParam(OssData * od, int param, int val)
+{
+	int supported = isSupportedParam(od, param, val);
+
+	if (supported == OSS_SUPPORTED)
+		return;
+
+	if (supported == OSS_UNSUPPORTED) {
+		removeUnsupportedParam(od, param, val);
+	}
+
+	addSupportedParam(od, param, val);
+}
+
+static void unsupportParam(OssData * od, int param, int val)
+{
+	int supported = isSupportedParam(od, param, val);
+
+	if (supported == OSS_UNSUPPORTED)
+		return;
+
+	if (supported == OSS_SUPPORTED) {
+		removeSupportedParam(od, param, val);
+	}
+
+	addUnsupportedParam(od, param, val);
+}
+
+static OssData *newOssData(void)
+{
+	OssData *ret = xmalloc(sizeof(OssData));
+
+	ret->device = NULL;
+	ret->fd = -1;
+
+	ret->supported[OSS_RATE] = NULL;
+	ret->supported[OSS_CHANNELS] = NULL;
+	ret->supported[OSS_BITS] = NULL;
+	ret->unsupported[OSS_RATE] = NULL;
+	ret->unsupported[OSS_CHANNELS] = NULL;
+	ret->unsupported[OSS_BITS] = NULL;
+
+	ret->numSupported[OSS_RATE] = 0;
+	ret->numSupported[OSS_CHANNELS] = 0;
+	ret->numSupported[OSS_BITS] = 0;
+	ret->numUnsupported[OSS_RATE] = 0;
+	ret->numUnsupported[OSS_CHANNELS] = 0;
+	ret->numUnsupported[OSS_BITS] = 0;
+
+	supportParam(ret, SNDCTL_DSP_SPEED, 48000);
+	supportParam(ret, SNDCTL_DSP_SPEED, 44100);
+	supportParam(ret, SNDCTL_DSP_CHANNELS, 2);
+	supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16);
+
+	return ret;
+}
+
+static void freeOssData(OssData * od)
+{
+	if (od->supported[OSS_RATE])
+		free(od->supported[OSS_RATE]);
+	if (od->supported[OSS_CHANNELS])
+		free(od->supported[OSS_CHANNELS]);
+	if (od->supported[OSS_BITS])
+		free(od->supported[OSS_BITS]);
+	if (od->unsupported[OSS_RATE])
+		free(od->unsupported[OSS_RATE]);
+	if (od->unsupported[OSS_CHANNELS])
+		free(od->unsupported[OSS_CHANNELS]);
+	if (od->unsupported[OSS_BITS])
+		free(od->unsupported[OSS_BITS]);
+
+	free(od);
+}
+
+#define OSS_STAT_NO_ERROR 	0
+#define OSS_STAT_NOT_CHAR_DEV	-1
+#define OSS_STAT_NO_PERMS	-2
+#define OSS_STAT_DOESN_T_EXIST	-3
+#define OSS_STAT_OTHER		-4
+
+static int oss_statDevice(const char *device, int *stErrno)
+{
+	struct stat st;
+
+	if (0 == stat(device, &st)) {
+		if (!S_ISCHR(st.st_mode)) {
+			return OSS_STAT_NOT_CHAR_DEV;
+		}
+	} else {
+		*stErrno = errno;
+
+		switch (errno) {
+		case ENOENT:
+		case ENOTDIR:
+			return OSS_STAT_DOESN_T_EXIST;
+		case EACCES:
+			return OSS_STAT_NO_PERMS;
+		default:
+			return OSS_STAT_OTHER;
+		}
+	}
+
+	return 0;
+}
+
+static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" };
+
+static int oss_testDefault(void)
+{
+	int fd, i;
+
+	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+		if ((fd = open(default_devices[i], O_WRONLY)) >= 0) {
+			xclose(fd);
+			return 0;
+		}
+		WARNING("Error opening OSS device \"%s\": %s\n",
+		        default_devices[i], strerror(errno));
+	}
+
+	return -1;
+}
+
+static int oss_open_default(AudioOutput *ao, ConfigParam *param, OssData *od)
+{
+	int i;
+	int err[ARRAY_SIZE(default_devices)];
+	int ret[ARRAY_SIZE(default_devices)];
+
+	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+		ret[i] = oss_statDevice(default_devices[i], &err[i]);
+		if (ret[i] == 0) {
+			od->device = default_devices[i];
+			return 0;
+		}
+	}
+
+	if (param)
+		ERROR("error trying to open specified OSS device"
+	              " at line %i\n", param->line);
+	else
+		ERROR("error trying to open default OSS device\n");
+
+	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+		const char *dev = default_devices[i];
+		switch(ret[i]) {
+		case OSS_STAT_DOESN_T_EXIST:
+			ERROR("%s not found\n", dev);
+			break;
+		case OSS_STAT_NOT_CHAR_DEV:
+			ERROR("%s is not a character device\n", dev);
+			break;
+		case OSS_STAT_NO_PERMS:
+			ERROR("%s: permission denied\n", dev);
+			break;
+		default:
+			ERROR("Error accessing %s: %s", dev, strerror(err[i]));
+		}
+	}
+	exit(EXIT_FAILURE);
+	return 0; /* some compilers can be dumb... */
+}
+
+static int oss_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+	OssData *od = newOssData();
+	audioOutput->data = od;
+	if (param) {
+		BlockParam *bp = getBlockParam(param, "device");
+		if (bp) {
+			od->device = bp->value;
+			return 0;
+		}
+	}
+	return oss_open_default(audioOutput, param, od);
+}
+
+static void oss_finishDriver(AudioOutput * audioOutput)
+{
+	OssData *od = audioOutput->data;
+
+	freeOssData(od);
+}
+
+static int setParam(OssData * od, int param, int *value)
+{
+	int val = *value;
+	int copy;
+	int supported = isSupportedParam(od, param, val);
+
+	do {
+		if (supported == OSS_UNSUPPORTED) {
+			val = getSupportedParam(od, param, val);
+			if (copy < 0)
+				return -1;
+		}
+		copy = val;
+		if (ioctl(od->fd, param, &copy)) {
+			unsupportParam(od, param, val);
+			supported = OSS_UNSUPPORTED;
+		} else {
+			if (supported == OSS_UNKNOWN) {
+				supportParam(od, param, val);
+				supported = OSS_SUPPORTED;
+			}
+			val = copy;
+		}
+	} while (supported == OSS_UNSUPPORTED);
+
+	*value = val;
+
+	return 0;
+}
+
+static void oss_close(OssData * od)
+{
+	if (od->fd >= 0)
+		while (close(od->fd) && errno == EINTR) ;
+	od->fd = -1;
+}
+
+static int oss_open(AudioOutput * audioOutput)
+{
+	int tmp;
+	OssData *od = audioOutput->data;
+
+	if ((od->fd = open(od->device, O_WRONLY)) < 0) {
+		ERROR("Error opening OSS device \"%s\": %s\n", od->device,
+		      strerror(errno));
+		goto fail;
+	}
+
+	if (setParam(od, SNDCTL_DSP_CHANNELS, &od->channels)) {
+		ERROR("OSS device \"%s\" does not support %i channels: %s\n",
+		      od->device, od->channels, strerror(errno));
+		goto fail;
+	}
+
+	if (setParam(od, SNDCTL_DSP_SPEED, &od->sampleRate)) {
+		ERROR("OSS device \"%s\" does not support %i Hz audio: %s\n",
+		      od->device, od->sampleRate, strerror(errno));
+		goto fail;
+	}
+
+	switch (od->bits) {
+	case 8:
+		tmp = AFMT_S8;
+		break;
+	case 16:
+		tmp = AFMT_S16_MPD;
+	}
+
+	if (setParam(od, SNDCTL_DSP_SAMPLESIZE, &tmp)) {
+		ERROR("OSS device \"%s\" does not support %i bit audio: %s\n",
+		      od->device, tmp, strerror(errno));
+		goto fail;
+	}
+
+	audioOutput->open = 1;
+
+	return 0;
+
+fail:
+	oss_close(od);
+	audioOutput->open = 0;
+	return -1;
+}
+
+static int oss_openDevice(AudioOutput * audioOutput)
+{
+	int ret = -1;
+	OssData *od = audioOutput->data;
+	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+
+	od->channels = audioFormat->channels;
+	od->sampleRate = audioFormat->sampleRate;
+	od->bits = audioFormat->bits;
+
+	if ((ret = oss_open(audioOutput)) < 0)
+		return ret;
+
+	audioFormat->channels = od->channels;
+	audioFormat->sampleRate = od->sampleRate;
+	audioFormat->bits = od->bits;
+
+	DEBUG("oss device \"%s\" will be playing %i bit %i channel audio at "
+	      "%i Hz\n", od->device, od->bits, od->channels, od->sampleRate);
+
+	return ret;
+}
+
+static void oss_closeDevice(AudioOutput * audioOutput)
+{
+	OssData *od = audioOutput->data;
+
+	oss_close(od);
+
+	audioOutput->open = 0;
+}
+
+static void oss_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	OssData *od = audioOutput->data;
+
+	if (od->fd >= 0) {
+		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+		oss_close(od);
+	}
+}
+
+static int oss_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+	OssData *od = audioOutput->data;
+	int ret;
+
+	/* reopen the device since it was closed by dropBufferedAudio */
+	if (od->fd < 0 && oss_open(audioOutput) < 0)
+		return -1;
+
+	while (size > 0) {
+		ret = write(od->fd, playChunk, size);
+		if (ret < 0) {
+			if (errno == EINTR)
+				continue;
+			ERROR("closing oss device \"%s\" due to write error: "
+			      "%s\n", od->device, strerror(errno));
+			oss_closeDevice(audioOutput);
+			return -1;
+		}
+		playChunk += ret;
+		size -= ret;
+	}
+
+	return 0;
+}
+
+AudioOutputPlugin ossPlugin = {
+	"oss",
+	oss_testDefault,
+	oss_initDriver,
+	oss_finishDriver,
+	oss_openDevice,
+	oss_playAudio,
+	oss_dropBufferedAudio,
+	oss_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else /* HAVE OSS */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
+#endif /* HAVE_OSS */
diff --git a/trunk/src/audioOutputs/audioOutput_osx.c b/trunk/src/audioOutputs/audioOutput_osx.c
new file mode 100644
index 000000000..1caebade5
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_osx.c
@@ -0,0 +1,374 @@
+/* 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 "../audioOutput.h"
+
+#ifdef HAVE_OSX
+
+#include <AudioUnit/AudioUnit.h>
+#include <stdlib.h>
+#include <pthread.h>
+
+#include "../log.h"
+
+typedef struct _OsxData {
+	AudioUnit au;
+	pthread_mutex_t mutex;
+	pthread_cond_t condition;
+	char *buffer;
+	int bufferSize;
+	int pos;
+	int len;
+	int started;
+} OsxData;
+
+static OsxData *newOsxData()
+{
+	OsxData *ret = xmalloc(sizeof(OsxData));
+
+	pthread_mutex_init(&ret->mutex, NULL);
+	pthread_cond_init(&ret->condition, NULL);
+
+	ret->pos = 0;
+	ret->len = 0;
+	ret->started = 0;
+	ret->buffer = NULL;
+	ret->bufferSize = 0;
+
+	return ret;
+}
+
+static int osx_testDefault()
+{
+	/*AudioUnit au;
+	   ComponentDescription desc;
+	   Component comp;
+
+	   desc.componentType = kAudioUnitType_Output;
+	   desc.componentSubType = kAudioUnitSubType_Output;
+	   desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+	   desc.componentFlags = 0;
+	   desc.componentFlagsMask = 0;
+
+	   comp = FindNextComponent(NULL, &desc);
+	   if(!comp) {
+	   ERROR("Unable to open default OS X defice\n");
+	   return -1;
+	   }
+
+	   if(OpenAComponent(comp, &au) != noErr) {
+	   ERROR("Unable to open default OS X defice\n");
+	   return -1;
+	   }
+
+	   CloseComponent(au); */
+
+	return 0;
+}
+
+static int osx_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+	OsxData *od = newOsxData();
+
+	audioOutput->data = od;
+
+	return 0;
+}
+
+static void freeOsxData(OsxData * od)
+{
+	if (od->buffer)
+		free(od->buffer);
+	pthread_mutex_destroy(&od->mutex);
+	pthread_cond_destroy(&od->condition);
+	free(od);
+}
+
+static void osx_finishDriver(AudioOutput * audioOutput)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+	freeOsxData(od);
+}
+
+static void osx_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+
+	pthread_mutex_lock(&od->mutex);
+	od->len = 0;
+	pthread_mutex_unlock(&od->mutex);
+}
+
+static void osx_closeDevice(AudioOutput * audioOutput)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+
+	pthread_mutex_lock(&od->mutex);
+	while (od->len) {
+		pthread_cond_wait(&od->condition, &od->mutex);
+	}
+	pthread_mutex_unlock(&od->mutex);
+
+	if (od->started) {
+		AudioOutputUnitStop(od->au);
+		od->started = 0;
+	}
+
+	CloseComponent(od->au);
+	AudioUnitUninitialize(od->au);
+
+	audioOutput->open = 0;
+}
+
+static OSStatus osx_render(void *vdata,
+			   AudioUnitRenderActionFlags * ioActionFlags,
+			   const AudioTimeStamp * inTimeStamp,
+			   UInt32 inBusNumber, UInt32 inNumberFrames,
+			   AudioBufferList * bufferList)
+{
+	OsxData *od = (OsxData *) vdata;
+	AudioBuffer *buffer = &bufferList->mBuffers[0];
+	int bufferSize = buffer->mDataByteSize;
+	int bytesToCopy;
+	int curpos = 0;
+
+	/*DEBUG("osx_render: enter : %i\n", (int)bufferList->mNumberBuffers);
+	   DEBUG("osx_render: ioActionFlags: %p\n", ioActionFlags);
+	   if(ioActionFlags) {
+	   if(*ioActionFlags & kAudioUnitRenderAction_PreRender) {
+	   DEBUG("prerender\n");
+	   }
+	   if(*ioActionFlags & kAudioUnitRenderAction_PostRender) {
+	   DEBUG("post render\n");
+	   }
+	   if(*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence) {
+	   DEBUG("post render\n");
+	   }
+	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight) {
+	   DEBUG("prefilight\n");
+	   }
+	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Render) {
+	   DEBUG("render\n");
+	   }
+	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Complete) {
+	   DEBUG("complete\n");
+	   }
+	   } */
+
+	/* while(bufferSize) {
+	   DEBUG("osx_render: lock\n"); */
+	pthread_mutex_lock(&od->mutex);
+	/*
+	   DEBUG("%i:%i\n", bufferSize, od->len);
+	   while(od->go && od->len < bufferSize && 
+	   od->len < od->bufferSize)
+	   {
+	   DEBUG("osx_render: wait\n");
+	   pthread_cond_wait(&od->condition, &od->mutex);
+	   }
+	 */
+
+	bytesToCopy = od->len < bufferSize ? od->len : bufferSize;
+	bufferSize = bytesToCopy;
+	od->len -= bytesToCopy;
+
+	if (od->pos + bytesToCopy > od->bufferSize) {
+		int bytes = od->bufferSize - od->pos;
+		memcpy(buffer->mData + curpos, od->buffer + od->pos, bytes);
+		od->pos = 0;
+		curpos += bytes;
+		bytesToCopy -= bytes;
+	}
+
+	memcpy(buffer->mData + curpos, od->buffer + od->pos, bytesToCopy);
+	od->pos += bytesToCopy;
+	curpos += bytesToCopy;
+
+	if (od->pos >= od->bufferSize)
+		od->pos = 0;
+	/* DEBUG("osx_render: unlock\n"); */
+	pthread_mutex_unlock(&od->mutex);
+	pthread_cond_signal(&od->condition);
+	/* } */
+
+	buffer->mDataByteSize = bufferSize;
+
+	if (!bufferSize) {
+		my_usleep(1000);
+	}
+
+	/* DEBUG("osx_render: leave\n"); */
+	return 0;
+}
+
+static int osx_openDevice(AudioOutput * audioOutput)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+	ComponentDescription desc;
+	Component comp;
+	AURenderCallbackStruct callback;
+	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+	AudioStreamBasicDescription streamDesc;
+
+	desc.componentType = kAudioUnitType_Output;
+	desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+	desc.componentFlags = 0;
+	desc.componentFlagsMask = 0;
+
+	comp = FindNextComponent(NULL, &desc);
+	if (comp == 0) {
+		ERROR("Error finding OS X component\n");
+		return -1;
+	}
+
+	if (OpenAComponent(comp, &od->au) != noErr) {
+		ERROR("Unable to open OS X component\n");
+		return -1;
+	}
+
+	if (AudioUnitInitialize(od->au) != 0) {
+		CloseComponent(od->au);
+		ERROR("Unable to initialize OS X audio unit\n");
+		return -1;
+	}
+
+	callback.inputProc = osx_render;
+	callback.inputProcRefCon = od;
+
+	if (AudioUnitSetProperty(od->au, kAudioUnitProperty_SetRenderCallback,
+				 kAudioUnitScope_Input, 0,
+				 &callback, sizeof(callback)) != 0) {
+		AudioUnitUninitialize(od->au);
+		CloseComponent(od->au);
+		ERROR("unable to set callback for OS X audio unit\n");
+		return -1;
+	}
+
+	streamDesc.mSampleRate = audioFormat->sampleRate;
+	streamDesc.mFormatID = kAudioFormatLinearPCM;
+	streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+#ifdef WORDS_BIGENDIAN
+	streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+
+	streamDesc.mBytesPerPacket =
+	    audioFormat->channels * audioFormat->bits / 8;
+	streamDesc.mFramesPerPacket = 1;
+	streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket;
+	streamDesc.mChannelsPerFrame = audioFormat->channels;
+	streamDesc.mBitsPerChannel = audioFormat->bits;
+
+	if (AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
+				 kAudioUnitScope_Input, 0,
+				 &streamDesc, sizeof(streamDesc)) != 0) {
+		AudioUnitUninitialize(od->au);
+		CloseComponent(od->au);
+		ERROR("Unable to set format on OS X device\n");
+		return -1;
+	}
+
+	/* create a buffer of 1s */
+	od->bufferSize = (audioFormat->sampleRate) *
+	    (audioFormat->bits >> 3) * (audioFormat->channels);
+	od->buffer = xrealloc(od->buffer, od->bufferSize);
+
+	od->pos = 0;
+	od->len = 0;
+
+	audioOutput->open = 1;
+
+	return 0;
+}
+
+static int osx_play(AudioOutput * audioOutput, char *playChunk, int size)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+	int bytesToCopy;
+	int curpos;
+
+	/* DEBUG("osx_play: enter\n"); */
+
+	if (!od->started) {
+		int err;
+		od->started = 1;
+		err = AudioOutputUnitStart(od->au);
+		if (err) {
+			ERROR("unable to start audio output: %i\n", err);
+			return -1;
+		}
+	}
+
+	pthread_mutex_lock(&od->mutex);
+
+	while (size) {
+		/* DEBUG("osx_play: lock\n"); */
+		curpos = od->pos + od->len;
+		if (curpos >= od->bufferSize)
+			curpos -= od->bufferSize;
+
+		bytesToCopy = od->bufferSize < size ? od->bufferSize : size;
+
+		while (od->len > od->bufferSize - bytesToCopy) {
+			/* DEBUG("osx_play: wait\n"); */
+			pthread_cond_wait(&od->condition, &od->mutex);
+		}
+
+		bytesToCopy = od->bufferSize - od->len;
+		bytesToCopy = bytesToCopy < size ? bytesToCopy : size;
+		size -= bytesToCopy;
+		od->len += bytesToCopy;
+
+		if (curpos + bytesToCopy > od->bufferSize) {
+			int bytes = od->bufferSize - curpos;
+			memcpy(od->buffer + curpos, playChunk, bytes);
+			curpos = 0;
+			playChunk += bytes;
+			bytesToCopy -= bytes;
+		}
+
+		memcpy(od->buffer + curpos, playChunk, bytesToCopy);
+		curpos += bytesToCopy;
+		playChunk += bytesToCopy;
+
+	}
+	/* DEBUG("osx_play: unlock\n"); */
+	pthread_mutex_unlock(&od->mutex);
+
+	/* DEBUG("osx_play: leave\n"); */
+	return 0;
+}
+
+AudioOutputPlugin osxPlugin = {
+	"osx",
+	osx_testDefault,
+	osx_initDriver,
+	osx_finishDriver,
+	osx_openDevice,
+	osx_play,
+	osx_dropBufferedAudio,
+	osx_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else
+
+#include <stdio.h>
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
+#endif
diff --git a/trunk/src/audioOutputs/audioOutput_pulse.c b/trunk/src/audioOutputs/audioOutput_pulse.c
new file mode 100644
index 000000000..8948e0263
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_pulse.c
@@ -0,0 +1,221 @@
+/* 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 "../audioOutput.h"
+
+#include <stdlib.h>
+
+#ifdef HAVE_PULSE
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+#include <time.h>
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+
+#define MPD_PULSE_NAME "mpd"
+#define CONN_ATTEMPT_INTERVAL 60
+
+typedef struct _PulseData {
+	pa_simple *s;
+	char *server;
+	char *sink;
+	int connAttempts;
+	time_t lastAttempt;
+} PulseData;
+
+static PulseData *newPulseData(void)
+{
+	PulseData *ret;
+
+	ret = xmalloc(sizeof(PulseData));
+
+	ret->s = NULL;
+	ret->server = NULL;
+	ret->sink = NULL;
+	ret->connAttempts = 0;
+	ret->lastAttempt = 0;
+
+	return ret;
+}
+
+static void freePulseData(PulseData * pd)
+{
+	if (pd->server)
+		free(pd->server);
+	if (pd->sink)
+		free(pd->sink);
+	free(pd);
+}
+
+static int pulse_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+	BlockParam *server = NULL;
+	BlockParam *sink = NULL;
+	PulseData *pd;
+
+	if (param) {
+		server = getBlockParam(param, "server");
+		sink = getBlockParam(param, "sink");
+	}
+
+	pd = newPulseData();
+	pd->server = server ? xstrdup(server->value) : NULL;
+	pd->sink = sink ? xstrdup(sink->value) : NULL;
+	audioOutput->data = pd;
+
+	return 0;
+}
+
+static void pulse_finishDriver(AudioOutput * audioOutput)
+{
+	freePulseData((PulseData *) audioOutput->data);
+}
+
+static int pulse_testDefault(void)
+{
+	pa_simple *s;
+	pa_sample_spec ss;
+	int error;
+
+	ss.format = PA_SAMPLE_S16NE;
+	ss.rate = 44100;
+	ss.channels = 2;
+
+	s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
+			  MPD_PULSE_NAME, &ss, NULL, NULL, &error);
+	if (!s) {
+		WARNING("Cannot connect to default PulseAudio server: %s\n",
+			pa_strerror(error));
+		return -1;
+	}
+
+	pa_simple_free(s);
+
+	return 0;
+}
+
+static int pulse_openDevice(AudioOutput * audioOutput)
+{
+	PulseData *pd;
+	AudioFormat *audioFormat;
+	pa_sample_spec ss;
+	time_t t;
+	int error;
+
+	t = time(NULL);
+	pd = audioOutput->data;
+	audioFormat = &audioOutput->outAudioFormat;
+
+	if (pd->connAttempts != 0 &&
+	    (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
+		return -1;
+
+	pd->connAttempts++;
+	pd->lastAttempt = t;
+
+	if (audioFormat->bits != 16) {
+		ERROR("PulseAudio doesn't support %i bit audio\n",
+		      audioFormat->bits);
+		return -1;
+	}
+
+	ss.format = PA_SAMPLE_S16NE;
+	ss.rate = audioFormat->sampleRate;
+	ss.channels = audioFormat->channels;
+
+	pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
+			      pd->sink, audioOutput->name, &ss, NULL, NULL,
+			      &error);
+	if (!pd->s) {
+		ERROR("Cannot connect to server in PulseAudio output "
+		      "\"%s\" (attempt %i): %s\n", audioOutput->name,
+		      pd->connAttempts, pa_strerror(error));
+		return -1;
+	}
+
+	pd->connAttempts = 0;
+	audioOutput->open = 1;
+
+	DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
+	      "channel audio at %i Hz\n", audioOutput->name, audioFormat->bits,
+	      audioFormat->channels, audioFormat->sampleRate);
+
+	return 0;
+}
+
+static void pulse_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	PulseData *pd;
+	int error;
+
+	pd = audioOutput->data;
+	if (pa_simple_flush(pd->s, &error) < 0)
+		WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
+			audioOutput->name, pa_strerror(error));
+}
+
+static void pulse_closeDevice(AudioOutput * audioOutput)
+{
+	PulseData *pd;
+
+	pd = audioOutput->data;
+	if (pd->s) {
+		pa_simple_drain(pd->s, NULL);
+		pa_simple_free(pd->s);
+	}
+
+	audioOutput->open = 0;
+}
+
+static int pulse_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+	PulseData *pd;
+	int error;
+
+	pd = audioOutput->data;
+
+	if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
+		ERROR("PulseAudio output \"%s\" disconnecting due to write "
+		      "error: %s\n", audioOutput->name, pa_strerror(error));
+		pulse_closeDevice(audioOutput);
+		return -1;
+	}
+
+	return 0;
+}
+
+AudioOutputPlugin pulsePlugin = {
+	"pulse",
+	pulse_testDefault,
+	pulse_initDriver,
+	pulse_finishDriver,
+	pulse_openDevice,
+	pulse_playAudio,
+	pulse_dropBufferedAudio,
+	pulse_closeDevice,
+	NULL,	/* sendMetadataFunc */
+};
+
+#else /* HAVE_PULSE */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
+#endif /* HAVE_PULSE */
diff --git a/trunk/src/audioOutputs/audioOutput_shout.c b/trunk/src/audioOutputs/audioOutput_shout.c
new file mode 100644
index 000000000..7d93f8f85
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_shout.c
@@ -0,0 +1,636 @@
+/* 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 "../audioOutput.h"
+
+#include <stdlib.h>
+
+#ifdef HAVE_SHOUT
+
+#include "../conf.h"
+#include "../log.h"
+#include "../pcm_utils.h"
+
+#include <string.h>
+#include <time.h>
+
+#include <shout/shout.h>
+#include <vorbis/vorbisenc.h>
+
+#define CONN_ATTEMPT_INTERVAL	60
+
+static int shoutInitCount;
+
+/* lots of this code blatantly stolent from bossogg/bossao2 */
+
+typedef struct _ShoutData {
+	shout_t *shoutConn;
+	int shoutError;
+
+	ogg_stream_state os;
+	ogg_page og;
+	ogg_packet op;
+	ogg_packet header_main;
+	ogg_packet header_comments;
+	ogg_packet header_codebooks;
+
+	vorbis_dsp_state vd;
+	vorbis_block vb;
+	vorbis_info vi;
+	vorbis_comment vc;
+
+	float quality;
+	int bitrate;
+
+	int opened;
+
+	MpdTag *tag;
+	int tagToSend;
+
+	int connAttempts;
+	time_t lastAttempt;
+	int last_err;
+
+	/* just a pointer to audioOutput->outAudioFormat */
+	AudioFormat *audioFormat;
+} ShoutData;
+
+static ShoutData *newShoutData(void)
+{
+	ShoutData *ret = xmalloc(sizeof(ShoutData));
+
+	ret->shoutConn = shout_new();
+	ret->opened = 0;
+	ret->tag = NULL;
+	ret->tagToSend = 0;
+	ret->bitrate = -1;
+	ret->quality = -2.0;
+	ret->connAttempts = 0;
+	ret->lastAttempt = 0;
+	ret->audioFormat = NULL;
+	ret->last_err = SHOUTERR_UNCONNECTED;
+
+	return ret;
+}
+
+static void freeShoutData(ShoutData * sd)
+{
+	if (sd->shoutConn)
+		shout_free(sd->shoutConn);
+	if (sd->tag)
+		freeMpdTag(sd->tag);
+
+	free(sd);
+}
+
+#define checkBlockParam(name) { \
+	blockParam = getBlockParam(param, name); \
+	if (!blockParam) { \
+		FATAL("no \"%s\" defined for shout device defined at line " \
+				"%i\n", name, param->line); \
+	} \
+}
+
+static int myShout_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+	ShoutData *sd;
+	char *test;
+	int port;
+	char *host;
+	char *mount;
+	char *passwd;
+	char *user;
+	char *name;
+	BlockParam *blockParam;
+	unsigned int public = 0;
+
+	sd = newShoutData();
+
+	if (shoutInitCount == 0)
+		shout_init();
+
+	shoutInitCount++;
+
+	checkBlockParam("host");
+	host = blockParam->value;
+
+	checkBlockParam("mount");
+	mount = blockParam->value;
+
+	checkBlockParam("port");
+
+	port = strtol(blockParam->value, &test, 10);
+
+	if (*test != '\0' || port <= 0) {
+		FATAL("shout port \"%s\" is not a positive integer, line %i\n",
+		      blockParam->value, blockParam->line);
+	}
+
+	checkBlockParam("password");
+	passwd = blockParam->value;
+
+	checkBlockParam("name");
+	name = blockParam->value;
+
+	blockParam = getBlockParam(param, "public");
+	if (blockParam) {
+		if (0 == strcmp(blockParam->value, "yes")) {
+			public = 1;
+		} else if (0 == strcmp(blockParam->value, "no")) {
+			public = 0;
+		} else {
+			FATAL("public \"%s\" is not \"yes\" or \"no\" at line "
+			      "%i\n", param->value, param->line);
+		}
+	}
+
+	blockParam = getBlockParam(param, "user");
+	if (blockParam)
+		user = blockParam->value;
+	else
+		user = "source";
+
+	blockParam = getBlockParam(param, "quality");
+
+	if (blockParam) {
+		int line = blockParam->line;
+
+		sd->quality = strtod(blockParam->value, &test);
+
+		if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
+			FATAL("shout quality \"%s\" is not a number in the "
+			      "range -1 to 10, line %i\n", blockParam->value,
+			      blockParam->line);
+		}
+
+		blockParam = getBlockParam(param, "bitrate");
+
+		if (blockParam) {
+			FATAL("quality (line %i) and bitrate (line %i) are "
+			      "both defined for shout output\n", line,
+			      blockParam->line);
+		}
+	} else {
+		blockParam = getBlockParam(param, "bitrate");
+
+		if (!blockParam) {
+			FATAL("neither bitrate nor quality defined for shout "
+			      "output at line %i\n", param->line);
+		}
+
+		sd->bitrate = strtol(blockParam->value, &test, 10);
+
+		if (*test != '\0' || sd->bitrate <= 0) {
+			FATAL("bitrate at line %i should be a positive integer "
+			      "\n", blockParam->line);
+		}
+	}
+
+	checkBlockParam("format");
+	sd->audioFormat = &audioOutput->outAudioFormat;
+
+	if (shout_set_host(sd->shoutConn, host) != SHOUTERR_SUCCESS ||
+	    shout_set_port(sd->shoutConn, port) != SHOUTERR_SUCCESS ||
+	    shout_set_password(sd->shoutConn, passwd) != SHOUTERR_SUCCESS ||
+	    shout_set_mount(sd->shoutConn, mount) != SHOUTERR_SUCCESS ||
+	    shout_set_name(sd->shoutConn, name) != SHOUTERR_SUCCESS ||
+	    shout_set_user(sd->shoutConn, user) != SHOUTERR_SUCCESS ||
+	    shout_set_public(sd->shoutConn, public) != SHOUTERR_SUCCESS ||
+	    shout_set_nonblocking(sd->shoutConn, 1) != SHOUTERR_SUCCESS ||
+	    shout_set_format(sd->shoutConn, SHOUT_FORMAT_VORBIS)
+	    != SHOUTERR_SUCCESS ||
+	    shout_set_protocol(sd->shoutConn, SHOUT_PROTOCOL_HTTP)
+	    != SHOUTERR_SUCCESS ||
+	    shout_set_agent(sd->shoutConn, "MPD") != SHOUTERR_SUCCESS) {
+		FATAL("error configuring shout defined at line %i: %s\n",
+		      param->line, shout_get_error(sd->shoutConn));
+	}
+
+	/* optional paramters */
+	blockParam = getBlockParam(param, "genre");
+	if (blockParam && shout_set_genre(sd->shoutConn, blockParam->value)) {
+		FATAL("error configuring shout defined at line %i: %s\n",
+		      param->line, shout_get_error(sd->shoutConn));
+	}
+
+	blockParam = getBlockParam(param, "description");
+	if (blockParam && shout_set_description(sd->shoutConn,
+						blockParam->value)) {
+		FATAL("error configuring shout defined at line %i: %s\n",
+		      param->line, shout_get_error(sd->shoutConn));
+	}
+
+	{
+		char temp[11];
+		memset(temp, 0, sizeof(temp));
+
+		snprintf(temp, sizeof(temp), "%d", sd->audioFormat->channels);
+		shout_set_audio_info(sd->shoutConn, SHOUT_AI_CHANNELS, temp);
+
+		snprintf(temp, sizeof(temp), "%d", sd->audioFormat->sampleRate);
+
+		shout_set_audio_info(sd->shoutConn, SHOUT_AI_SAMPLERATE, temp);
+
+		if (sd->quality >= -1.0) {
+			snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
+			shout_set_audio_info(sd->shoutConn, SHOUT_AI_QUALITY,
+					     temp);
+		} else {
+			snprintf(temp, sizeof(temp), "%d", sd->bitrate);
+			shout_set_audio_info(sd->shoutConn, SHOUT_AI_BITRATE,
+					     temp);
+		}
+	}
+
+	audioOutput->data = sd;
+
+	return 0;
+}
+
+static int myShout_handleError(ShoutData * sd, int err)
+{
+	switch (err) {
+	case SHOUTERR_SUCCESS:
+		break;
+	case SHOUTERR_UNCONNECTED:
+	case SHOUTERR_SOCKET:
+		ERROR("Lost shout connection to %s:%i : %s\n",
+		      shout_get_host(sd->shoutConn),
+		      shout_get_port(sd->shoutConn),
+		      shout_get_error(sd->shoutConn));
+		sd->shoutError = 1;
+		return -1;
+	default:
+		ERROR("shout: connection to %s:%i error : %s\n",
+		      shout_get_host(sd->shoutConn),
+		      shout_get_port(sd->shoutConn),
+		      shout_get_error(sd->shoutConn));
+		sd->shoutError = 1;
+		return -1;
+	}
+
+	return 0;
+}
+
+static int write_page(ShoutData * sd)
+{
+	int err = 0;
+
+	/*DEBUG("shout_delay: %i\n", shout_delay(sd->shoutConn)); */
+	shout_sync(sd->shoutConn);
+	err = shout_send(sd->shoutConn, sd->og.header, sd->og.header_len);
+	if (myShout_handleError(sd, err) < 0)
+		return -1;
+	err = shout_send(sd->shoutConn, sd->og.body, sd->og.body_len);
+	if (myShout_handleError(sd, err) < 0)
+		return -1;
+
+	return 0;
+}
+
+static void finishEncoder(ShoutData * sd)
+{
+	vorbis_analysis_wrote(&sd->vd, 0);
+
+	while (vorbis_analysis_blockout(&sd->vd, &sd->vb) == 1) {
+		vorbis_analysis(&sd->vb, NULL);
+		vorbis_bitrate_addblock(&sd->vb);
+		while (vorbis_bitrate_flushpacket(&sd->vd, &sd->op)) {
+			ogg_stream_packetin(&sd->os, &sd->op);
+		}
+	}
+}
+
+static int flushEncoder(ShoutData * sd)
+{
+	return (ogg_stream_pageout(&sd->os, &sd->og) > 0);
+}
+
+static void clearEncoder(ShoutData * sd)
+{
+	finishEncoder(sd);
+	while (1 == flushEncoder(sd)) {
+		if (!sd->shoutError)
+			write_page(sd);
+	}
+
+	vorbis_comment_clear(&sd->vc);
+	ogg_stream_clear(&sd->os);
+	vorbis_block_clear(&sd->vb);
+	vorbis_dsp_clear(&sd->vd);
+	vorbis_info_clear(&sd->vi);
+}
+
+static void myShout_closeShoutConn(ShoutData * sd)
+{
+	if (sd->opened) {
+		clearEncoder(sd);
+
+		if (shout_close(sd->shoutConn) != SHOUTERR_SUCCESS) {
+			ERROR("problem closing connection to shout server: "
+			      "%s\n", shout_get_error(sd->shoutConn));
+		}
+	}
+
+	sd->last_err = SHOUTERR_UNCONNECTED;
+	sd->opened = 0;
+}
+
+static void myShout_finishDriver(AudioOutput * audioOutput)
+{
+	ShoutData *sd = (ShoutData *) audioOutput->data;
+
+	myShout_closeShoutConn(sd);
+
+	freeShoutData(sd);
+
+	shoutInitCount--;
+
+	if (shoutInitCount == 0)
+		shout_shutdown();
+}
+
+static void myShout_dropBufferedAudio(AudioOutput * audioOutput)
+{
+	/* needs to be implemented */
+}
+
+static void myShout_closeDevice(AudioOutput * audioOutput)
+{
+	ShoutData *sd = (ShoutData *) audioOutput->data;
+
+	myShout_closeShoutConn(sd);
+
+	audioOutput->open = 0;
+}
+
+#define addTag(name, value) { \
+	if(value) vorbis_comment_add_tag(&(sd->vc), name, value); \
+}
+
+static void copyTagToVorbisComment(ShoutData * sd)
+{
+	if (sd->tag) {
+		int i;
+
+		for (i = 0; i < sd->tag->numOfItems; i++) {
+			switch (sd->tag->items[i].type) {
+			case TAG_ITEM_ARTIST:
+				addTag("ARTIST", sd->tag->items[i].value);
+				break;
+			case TAG_ITEM_ALBUM:
+				addTag("ALBUM", sd->tag->items[i].value);
+				break;
+			case TAG_ITEM_TITLE:
+				addTag("TITLE", sd->tag->items[i].value);
+				break;
+			}
+		}
+	}
+}
+
+static int initEncoder(ShoutData * sd)
+{
+	vorbis_info_init(&(sd->vi));
+
+	if (sd->quality >= -1.0) {
+		if (0 != vorbis_encode_init_vbr(&(sd->vi),
+						sd->audioFormat->channels,
+						sd->audioFormat->sampleRate,
+						sd->quality * 0.1)) {
+			ERROR("problem setting up vorbis encoder for shout\n");
+			vorbis_info_clear(&(sd->vi));
+			return -1;
+		}
+	} else {
+		if (0 != vorbis_encode_init(&(sd->vi),
+					    sd->audioFormat->channels,
+					    sd->audioFormat->sampleRate, -1.0,
+					    sd->bitrate * 1000, -1.0)) {
+			ERROR("problem setting up vorbis encoder for shout\n");
+			vorbis_info_clear(&(sd->vi));
+			return -1;
+		}
+	}
+
+	vorbis_analysis_init(&(sd->vd), &(sd->vi));
+	vorbis_block_init(&(sd->vd), &(sd->vb));
+
+	ogg_stream_init(&(sd->os), rand());
+
+	vorbis_comment_init(&(sd->vc));
+
+	return 0;
+}
+
+static int myShout_openShoutConn(AudioOutput * audioOutput)
+{
+	ShoutData *sd = (ShoutData *) audioOutput->data;
+	time_t t = time(NULL);
+
+	if (sd->connAttempts != 0 &&
+	    (t - sd->lastAttempt) < CONN_ATTEMPT_INTERVAL) {
+		return -1;
+	}
+
+	sd->connAttempts++;
+
+	if (sd->last_err == SHOUTERR_UNCONNECTED)
+		sd->last_err = shout_open(sd->shoutConn);
+	switch (sd->last_err) {
+	case SHOUTERR_SUCCESS:
+	case SHOUTERR_CONNECTED:
+		break;
+	case SHOUTERR_BUSY:
+		sd->last_err = shout_get_connected(sd->shoutConn);
+		if (sd->last_err == SHOUTERR_CONNECTED)
+			break;
+		return -1;
+	default:
+		sd->lastAttempt = t;
+		ERROR("problem opening connection to shout server %s:%i "
+		      "(attempt %i): %s\n",
+		      shout_get_host(sd->shoutConn),
+		      shout_get_port(sd->shoutConn),
+		      sd->connAttempts, shout_get_error(sd->shoutConn));
+		return -1;
+	}
+
+	if (initEncoder(sd) < 0) {
+		shout_close(sd->shoutConn);
+		return -1;
+	}
+
+	sd->shoutError = 0;
+
+	copyTagToVorbisComment(sd);
+
+	vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
+				  &(sd->header_comments),
+				  &(sd->header_codebooks));
+
+	ogg_stream_packetin(&(sd->os), &(sd->header_main));
+	ogg_stream_packetin(&(sd->os), &(sd->header_comments));
+	ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
+
+	sd->opened = 1;
+	sd->tagToSend = 0;
+
+	while (ogg_stream_flush(&(sd->os), &(sd->og))) {
+		if (write_page(sd) < 0) {
+			myShout_closeShoutConn(sd);
+			return -1;
+		}
+	}
+
+	sd->connAttempts = 0;
+
+	return 0;
+}
+
+static int myShout_openDevice(AudioOutput * audioOutput)
+{
+	ShoutData *sd = (ShoutData *) audioOutput->data;
+
+	audioOutput->open = 1;
+
+	if (sd->opened)
+		return 0;
+
+	if (myShout_openShoutConn(audioOutput) < 0) {
+		audioOutput->open = 0;
+		return -1;
+	}
+
+	return 0;
+}
+
+static void myShout_sendMetadata(ShoutData * sd)
+{
+	if (!sd->opened || !sd->tag)
+		return;
+
+	clearEncoder(sd);
+	if (initEncoder(sd) < 0)
+		return;
+
+	copyTagToVorbisComment(sd);
+
+	vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
+				  &(sd->header_comments),
+				  &(sd->header_codebooks));
+
+	ogg_stream_packetin(&(sd->os), &(sd->header_main));
+	ogg_stream_packetin(&(sd->os), &(sd->header_comments));
+	ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
+
+	while (ogg_stream_flush(&(sd->os), &(sd->og))) {
+		if (write_page(sd) < 0) {
+			myShout_closeShoutConn(sd);
+			return;
+		}
+	}
+
+	/*if(sd->tag) freeMpdTag(sd->tag);
+	   sd->tag = NULL; */
+	sd->tagToSend = 0;
+}
+
+static int myShout_play(AudioOutput * audioOutput, char *playChunk, int size)
+{
+	int i, j;
+	ShoutData *sd = (ShoutData *) audioOutput->data;
+	float **vorbbuf;
+	int samples;
+	int bytes = sd->audioFormat->bits / 8;
+
+	if (sd->opened && sd->tagToSend)
+		myShout_sendMetadata(sd);
+
+	if (!sd->opened) {
+		if (myShout_openShoutConn(audioOutput) < 0) {
+			return -1;
+		}
+	}
+
+	samples = size / (bytes * sd->audioFormat->channels);
+
+	/* this is for only 16-bit audio */
+
+	vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples);
+
+	for (i = 0; i < samples; i++) {
+		for (j = 0; j < sd->audioFormat->channels; j++) {
+			vorbbuf[j][i] = (*((mpd_sint16 *) playChunk)) / 32768.0;
+			playChunk += bytes;
+		}
+	}
+
+	vorbis_analysis_wrote(&(sd->vd), samples);
+
+	while (1 == vorbis_analysis_blockout(&(sd->vd), &(sd->vb))) {
+		vorbis_analysis(&(sd->vb), NULL);
+		vorbis_bitrate_addblock(&(sd->vb));
+
+		while (vorbis_bitrate_flushpacket(&(sd->vd), &(sd->op))) {
+			ogg_stream_packetin(&(sd->os), &(sd->op));
+		}
+	}
+
+	while (ogg_stream_pageout(&(sd->os), &(sd->og)) != 0) {
+		if (write_page(sd) < 0) {
+			myShout_closeShoutConn(sd);
+			return -1;
+		}
+	}
+
+	return 0;
+}
+
+static void myShout_setTag(AudioOutput * audioOutput, MpdTag * tag)
+{
+	ShoutData *sd = (ShoutData *) audioOutput->data;
+
+	if (sd->tag)
+		freeMpdTag(sd->tag);
+	sd->tag = NULL;
+	sd->tagToSend = 0;
+
+	if (!tag)
+		return;
+
+	sd->tag = mpdTagDup(tag);
+	sd->tagToSend = 1;
+}
+
+AudioOutputPlugin shoutPlugin = {
+	"shout",
+	NULL,
+	myShout_initDriver,
+	myShout_finishDriver,
+	myShout_openDevice,
+	myShout_play,
+	myShout_dropBufferedAudio,
+	myShout_closeDevice,
+	myShout_setTag,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
+#endif