renamed src/audioOutputs/ to src/output/

Again, no CamelCase in the directory name.

13 files changed, 4178 insertions, 0 deletions

diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
new file mode 100644
index 00000000..1845f1b7
--- /dev/null
+++ b/src/output/alsa_plugin.c
@@ -0,0 +1,444 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_ALSA
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+static const char default_device[] = "default";
+
+#define MPD_ALSA_RETRY_NR 5
+
+#include "../utils.h"
+#include "../log.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 {
+	const char *device;
+
+	/** the mode flags passed to snd_pcm_open */
+	int mode;
+
+	snd_pcm_t *pcmHandle;
+	alsa_writei_t *writei;
+	unsigned int buffer_time;
+	unsigned int period_time;
+	int sampleSize;
+	int useMmap;
+} AlsaData;
+
+static AlsaData *newAlsaData(void)
+{
+	AlsaData *ret = xmalloc(sizeof(AlsaData));
+
+	ret->device = default_device;
+	ret->mode = 0;
+	ret->pcmHandle = NULL;
+	ret->writei = snd_pcm_writei;
+	ret->useMmap = 0;
+	ret->buffer_time = 0;
+	ret->period_time = 0;
+
+	return ret;
+}
+
+static void freeAlsaData(AlsaData * ad)
+{
+	if (ad->device && ad->device != default_device)
+		xfree(ad->device);
+	free(ad);
+}
+
+static void
+alsa_configure(AlsaData *ad, ConfigParam *param)
+{
+	BlockParam *bp;
+
+	if ((bp = getBlockParam(param, "device")))
+		ad->device = xstrdup(bp->value);
+	ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
+	if (ad->useMmap == CONF_BOOL_UNSET)
+		ad->useMmap = 0;
+	if ((bp = getBlockParam(param, "buffer_time")))
+		ad->buffer_time = atoi(bp->value);
+	if ((bp = getBlockParam(param, "period_time")))
+		ad->period_time = atoi(bp->value);
+
+#ifdef SND_PCM_NO_AUTO_RESAMPLE
+	if (!getBoolBlockParam(param, "auto_resample", true))
+		ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
+#endif
+
+#ifdef SND_PCM_NO_AUTO_CHANNELS
+	if (!getBoolBlockParam(param, "auto_channels", true))
+		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
+#endif
+
+#ifdef SND_PCM_NO_AUTO_FORMAT
+	if (!getBoolBlockParam(param, "auto_format", true))
+		ad->mode |= SND_PCM_NO_AUTO_FORMAT;
+#endif
+}
+
+static void *alsa_initDriver(mpd_unused struct audio_output *ao,
+			     mpd_unused const struct audio_format *audio_format,
+			     ConfigParam * param)
+{
+	/* no need for pthread_once thread-safety when reading config */
+	static int free_global_registered;
+	AlsaData *ad = newAlsaData();
+
+	if (!free_global_registered) {
+		atexit((void(*)(void))snd_config_update_free_global);
+		free_global_registered = 1;
+	}
+
+	if (param)
+		alsa_configure(ad, param);
+
+	return ad;
+}
+
+static void alsa_finishDriver(void *data)
+{
+	AlsaData *ad = data;
+
+	freeAlsaData(ad);
+}
+
+static int alsa_testDefault(void)
+{
+	snd_pcm_t *handle;
+
+	int ret = snd_pcm_open(&handle, default_device,
+	                       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+	if (ret) {
+		WARNING("Error opening default ALSA device: %s\n",
+			snd_strerror(-ret));
+		return -1;
+	} else
+		snd_pcm_close(handle);
+
+	return 0;
+}
+
+static snd_pcm_format_t get_bitformat(const struct audio_format *af)
+{
+	switch (af->bits) {
+	case 8: return SND_PCM_FORMAT_S8;
+	case 16: return SND_PCM_FORMAT_S16;
+	case 24: return SND_PCM_FORMAT_S24;
+	case 32: return SND_PCM_FORMAT_S32;
+	}
+	return SND_PCM_FORMAT_UNKNOWN;
+}
+
+static int alsa_openDevice(void *data, struct audio_format *audioFormat)
+{
+	AlsaData *ad = data;
+	snd_pcm_format_t bitformat;
+	snd_pcm_hw_params_t *hwparams;
+	snd_pcm_sw_params_t *swparams;
+	unsigned int sample_rate = audioFormat->sample_rate;
+	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;
+
+	if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
+		ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
+		      ad->device, audioFormat->bits);
+
+	err = snd_pcm_open(&ad->pcmHandle, ad->device,
+			   SND_PCM_STREAM_PLAYBACK, ad->mode);
+	if (err < 0) {
+		ad->pcmHandle = NULL;
+		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 == -EINVAL && audioFormat->bits != 16) {
+		/* fall back to 16 bit, let pcm_utils.c do the conversion */
+		err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams,
+						   SND_PCM_FORMAT_S16);
+		if (err == 0) {
+			DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n",
+			      ad->device, audioFormat->bits);
+			audioFormat->bits = 16;
+		}
+	}
+
+	if (err < 0) {
+		ERROR("ALSA device \"%s\" does not support %u 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 = (int8_t)channels;
+
+	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
+					      &sample_rate, NULL);
+	if (err < 0 || sample_rate == 0) {
+		ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
+		      ad->device, audioFormat->sample_rate);
+		goto fail;
+	}
+	audioFormat->sample_rate = sample_rate;
+
+	if (ad->buffer_time > 0) {
+		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;
+	}
+
+	if (period_time_ro > 0) {
+		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 > 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;
+
+	/* 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";
+	err = snd_pcm_sw_params(ad->pcmHandle, swparams);
+	if (err < 0)
+		goto error;
+
+	ad->sampleSize = audio_format_frame_size(audioFormat);
+
+	DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
+	      "%u Hz\n", ad->device, audioFormat->bits,
+	      channels, sample_rate);
+
+	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;
+	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 = snd_pcm_resume(ad->pcmHandle);
+		if (err == -EAGAIN)
+			return 0;
+		/* fall-through to snd_pcm_prepare: */
+	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;
+	/* this is no error, so just keep running */
+	case SND_PCM_STATE_RUNNING:
+		err = 0;
+		break;
+	default:
+		/* unknown state, do nothing */
+		break;
+	}
+
+	return err;
+}
+
+static void alsa_dropBufferedAudio(void *data)
+{
+	AlsaData *ad = data;
+
+	alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
+}
+
+static void alsa_closeDevice(void *data)
+{
+	AlsaData *ad = data;
+
+	if (ad->pcmHandle) {
+		if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
+			snd_pcm_drain(ad->pcmHandle);
+		}
+		snd_pcm_close(ad->pcmHandle);
+		ad->pcmHandle = NULL;
+	}
+}
+
+static int alsa_playAudio(void *data, const char *playChunk, size_t size)
+{
+	AlsaData *ad = 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(ad);
+				return -1;
+			}
+			continue;
+		}
+
+		playChunk += ret * ad->sampleSize;
+		size -= ret;
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin alsaPlugin = {
+	.name = "alsa",
+	.test_default_device = alsa_testDefault,
+	.init = alsa_initDriver,
+	.finish = alsa_finishDriver,
+	.open = alsa_openDevice,
+	.play = alsa_playAudio,
+	.cancel = alsa_dropBufferedAudio,
+	.close = alsa_closeDevice,
+};
+
+#else /* HAVE ALSA */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
+#endif /* HAVE_ALSA */
diff --git a/src/output/ao_plugin.c b/src/output/ao_plugin.c
new file mode 100644
index 00000000..e731f972
--- /dev/null
+++ b/src/output/ao_plugin.c
@@ -0,0 +1,253 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_AO
+
+#include "../utils.h"
+#include "../log.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 void *audioOutputAo_initDriver(struct audio_output *ao,
+				      mpd_unused const struct audio_format *audio_format,
+				      ConfigParam * param)
+{
+	ao_info *ai;
+	char *duplicated;
+	char *stk1;
+	char *stk2;
+	char *n1;
+	char *key;
+	char *value;
+	char *test;
+	AoData *ad = newAoData();
+	BlockParam *blockParam;
+
+	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,
+	      audio_output_get_name(ao));
+
+	blockParam = getBlockParam(param, "options");
+
+	if (blockParam) {
+		duplicated = xstrdup(blockParam->value);
+	} else
+		duplicated = xstrdup("");
+
+	if (strlen(duplicated)) {
+		stk1 = NULL;
+		n1 = strtok_r(duplicated, ";", &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(duplicated);
+
+	return ad;
+}
+
+static void freeAoData(AoData * ad)
+{
+	ao_free_options(ad->options);
+	free(ad);
+}
+
+static void audioOutputAo_finishDriver(void *data)
+{
+	AoData *ad = (AoData *)data;
+	freeAoData(ad);
+
+	driverInitCount--;
+
+	if (driverInitCount == 0)
+		ao_shutdown();
+}
+
+static void audioOutputAo_dropBufferedAudio(mpd_unused void *data)
+{
+	/* not supported by libao */
+}
+
+static void audioOutputAo_closeDevice(void *data)
+{
+	AoData *ad = (AoData *)data;
+
+	if (ad->device) {
+		ao_close(ad->device);
+		ad->device = NULL;
+	}
+}
+
+static int audioOutputAo_openDevice(void *data,
+				    struct audio_format *audio_format)
+{
+	ao_sample_format format;
+	AoData *ad = (AoData *)data;
+
+	if (ad->device) {
+		audioOutputAo_closeDevice(ad);
+	}
+
+	format.bits = audio_format->bits;
+	format.rate = audio_format->sample_rate;
+	format.byte_format = AO_FMT_NATIVE;
+	format.channels = audio_format->channels;
+
+	ad->device = ao_open_live(ad->driverId, &format, ad->options);
+
+	if (ad->device == NULL)
+		return -1;
+
+	return 0;
+}
+
+/**
+ * For whatever reason, libao wants a non-const pointer.  Let's hope
+ * it does not write to the buffer, and use the union deconst hack to
+ * work around this API misdesign.
+ */
+static int ao_play_deconst(ao_device *device, const void *output_samples,
+			   uint_32 num_bytes)
+{
+	union {
+		const void *in;
+		void *out;
+	} u;
+
+	u.in = output_samples;
+	return ao_play(device, u.out, num_bytes);
+}
+
+static int audioOutputAo_play(void *data, const char *playChunk, size_t size)
+{
+	AoData *ad = (AoData *)data;
+	size_t chunk_size;
+
+	if (ad->device == NULL)
+		return -1;
+
+	while (size > 0) {
+		chunk_size = (size_t)ad->writeSize > size
+			? size : (size_t)ad->writeSize;
+
+		if (ao_play_deconst(ad->device, playChunk, chunk_size) == 0) {
+			audioOutputAo_error();
+			ERROR("closing audio device due to write error\n");
+			audioOutputAo_closeDevice(ad);
+			return -1;
+		}
+
+		playChunk += chunk_size;
+		size -= chunk_size;
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin aoPlugin = {
+	.name = "ao",
+	.init = audioOutputAo_initDriver,
+	.finish = audioOutputAo_finishDriver,
+	.open = audioOutputAo_openDevice,
+	.play = audioOutputAo_play,
+	.cancel = audioOutputAo_dropBufferedAudio,
+	.close = audioOutputAo_closeDevice,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
+#endif
diff --git a/src/output/fifo_plugin.c b/src/output/fifo_plugin.c
new file mode 100644
index 00000000..d7eb91ff
--- /dev/null
+++ b/src/output/fifo_plugin.c
@@ -0,0 +1,290 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_FIFO
+
+#include "../log.h"
+#include "../utils.h"
+#include "../timer.h"
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
+#define FIFO_BUFFER_SIZE 65536 /* pipe capacity on Linux >= 2.6.11 */
+
+typedef struct _FifoData {
+	char *path;
+	int input;
+	int output;
+	int created;
+	Timer *timer;
+} FifoData;
+
+static FifoData *newFifoData(void)
+{
+	FifoData *ret;
+
+	ret = xmalloc(sizeof(FifoData));
+
+	ret->path = NULL;
+	ret->input = -1;
+	ret->output = -1;
+	ret->created = 0;
+	ret->timer = NULL;
+	
+	return ret;
+}
+
+static void freeFifoData(FifoData *fd)
+{
+	if (fd->path)
+		free(fd->path);
+
+	if (fd->timer)
+		timer_free(fd->timer);
+
+	free(fd);
+}
+
+static void removeFifo(FifoData *fd)
+{
+	DEBUG("Removing FIFO \"%s\"\n", fd->path);
+
+	if (unlink(fd->path) < 0) {
+		ERROR("Could not remove FIFO \"%s\": %s\n",
+		      fd->path, strerror(errno));
+		return;
+	}
+
+	fd->created = 0;
+}
+
+static void closeFifo(FifoData *fd)
+{
+	struct stat st;
+
+	if (fd->input >= 0) {
+		close(fd->input);
+		fd->input = -1;
+	}
+
+	if (fd->output >= 0) {
+		close(fd->output);
+		fd->output = -1;
+	}
+
+	if (fd->created && (stat(fd->path, &st) == 0))
+		removeFifo(fd);
+}
+
+static int makeFifo(FifoData *fd)
+{
+	if (mkfifo(fd->path, 0666) < 0) {
+		ERROR("Couldn't create FIFO \"%s\": %s\n",
+		      fd->path, strerror(errno));
+		return -1;
+	}
+
+	fd->created = 1;
+
+	return 0;
+}
+
+static int checkFifo(FifoData *fd)
+{
+	struct stat st;
+
+	if (stat(fd->path, &st) < 0) {
+		if (errno == ENOENT) {
+			/* Path doesn't exist */
+			return makeFifo(fd);
+		}
+
+		ERROR("Failed to stat FIFO \"%s\": %s\n",
+		      fd->path, strerror(errno));
+		return -1;
+	}
+
+	if (!S_ISFIFO(st.st_mode)) {
+		ERROR("\"%s\" already exists, but is not a FIFO\n", fd->path);
+		return -1;
+	}
+
+	return 0;
+}
+
+static int openFifo(FifoData *fd)
+{
+	if (checkFifo(fd) < 0)
+		return -1;
+
+	fd->input = open(fd->path, O_RDONLY|O_NONBLOCK);
+	if (fd->input < 0) {
+		ERROR("Could not open FIFO \"%s\" for reading: %s\n",
+		      fd->path, strerror(errno));
+		closeFifo(fd);
+		return -1;
+	}
+
+	fd->output = open(fd->path, O_WRONLY|O_NONBLOCK);
+	if (fd->output < 0) {
+		ERROR("Could not open FIFO \"%s\" for writing: %s\n",
+		      fd->path, strerror(errno));
+		closeFifo(fd);
+		return -1;
+	}
+
+	return 0;
+}
+
+static void *fifo_initDriver(mpd_unused struct audio_output *ao,
+			     mpd_unused const struct audio_format *audio_format,
+			     ConfigParam *param)
+{
+	FifoData *fd;
+	BlockParam *blockParam;
+	char *path;
+
+	blockParam = getBlockParam(param, "path");
+	if (!blockParam) {
+		FATAL("No \"path\" parameter specified for fifo output "
+		      "defined at line %i\n", param->line);
+	}
+
+	path = parsePath(blockParam->value);
+	if (!path) {
+		FATAL("Could not parse \"path\" parameter for fifo output "
+		      "at line %i\n", blockParam->line);
+	}
+
+	fd = newFifoData();
+	fd->path = path;
+
+	if (openFifo(fd) < 0) {
+		freeFifoData(fd);
+		return NULL;
+	}
+
+	return fd;
+}
+
+static void fifo_finishDriver(void *data)
+{
+	FifoData *fd = (FifoData *)data;
+
+	closeFifo(fd);
+	freeFifoData(fd);
+}
+
+static int fifo_openDevice(void *data,
+			   struct audio_format *audio_format)
+{
+	FifoData *fd = (FifoData *)data;
+
+	if (fd->timer)
+		timer_free(fd->timer);
+
+	fd->timer = timer_new(audio_format);
+
+	return 0;
+}
+
+static void fifo_closeDevice(void *data)
+{
+	FifoData *fd = (FifoData *)data;
+
+	if (fd->timer) {
+		timer_free(fd->timer);
+		fd->timer = NULL;
+	}
+}
+
+static void fifo_dropBufferedAudio(void *data)
+{
+	FifoData *fd = (FifoData *)data;
+	char buf[FIFO_BUFFER_SIZE];
+	int bytes = 1;
+
+	timer_reset(fd->timer);
+
+	while (bytes > 0 && errno != EINTR)
+		bytes = read(fd->input, buf, FIFO_BUFFER_SIZE);
+
+	if (bytes < 0 && errno != EAGAIN) {
+		WARNING("Flush of FIFO \"%s\" failed: %s\n",
+		        fd->path, strerror(errno));
+	}
+}
+
+static int fifo_playAudio(void *data,
+			  const char *playChunk, size_t size)
+{
+	FifoData *fd = (FifoData *)data;
+	size_t offset = 0;
+	ssize_t bytes;
+
+	if (!fd->timer->started)
+		timer_start(fd->timer);
+	else
+		timer_sync(fd->timer);
+
+	timer_add(fd->timer, size);
+
+	while (size) {
+		bytes = write(fd->output, playChunk + offset, size);
+		if (bytes < 0) {
+			switch (errno) {
+			case EAGAIN:
+				/* The pipe is full, so empty it */
+				fifo_dropBufferedAudio(fd);
+				continue;
+			case EINTR:
+				continue;
+			}
+
+			ERROR("Closing FIFO output \"%s\" due to write error: "
+			      "%s\n", fd->path, strerror(errno));
+			fifo_closeDevice(fd);
+			return -1;
+		}
+
+		size -= bytes;
+		offset += bytes;
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin fifoPlugin = {
+	.name = "fifo",
+	.init = fifo_initDriver,
+	.finish = fifo_finishDriver,
+	.open = fifo_openDevice,
+	.play = fifo_playAudio,
+	.cancel = fifo_dropBufferedAudio,
+	.close = fifo_closeDevice,
+};
+
+#else /* HAVE_FIFO */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(fifoPlugin)
+
+#endif /* !HAVE_FIFO */
diff --git a/src/output/jack_plugin.c b/src/output/jack_plugin.c
new file mode 100644
index 00000000..6f2bcd3a
--- /dev/null
+++ b/src/output/jack_plugin.c
@@ -0,0 +1,486 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_JACK
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+
+#include <jack/jack.h>
+#include <jack/types.h>
+#include <jack/ringbuffer.h>
+
+static const size_t sample_size = sizeof(jack_default_audio_sample_t);
+
+struct jack_data {
+	struct audio_output *ao;
+
+	/* configuration */
+	const char *name;
+	const char *output_ports[2];
+	int ringbuffer_size;
+
+	/* for srate() only */
+	struct audio_format *audio_format;
+
+	/* jack library stuff */
+	jack_port_t *ports[2];
+	jack_client_t *client;
+	jack_ringbuffer_t *ringbuffer[2];
+	int bps;
+	int shutdown;
+};
+
+static struct jack_data *
+mpd_jack_new(void)
+{
+	struct jack_data *ret;
+
+	ret = xcalloc(sizeof(*ret), 1);
+
+	ret->name = "mpd";
+	ret->ringbuffer_size = 32768;
+
+	return ret;
+}
+
+static void
+mpd_jack_client_free(struct jack_data *jd)
+{
+	assert(jd != NULL);
+
+	if (jd->client != NULL) {
+		jack_deactivate(jd->client);
+		jack_client_close(jd->client);
+		jd->client = NULL;
+	}
+
+	if (jd->ringbuffer[0] != NULL) {
+		jack_ringbuffer_free(jd->ringbuffer[0]);
+		jd->ringbuffer[0] = NULL;
+	}
+
+	if (jd->ringbuffer[1] != NULL) {
+		jack_ringbuffer_free(jd->ringbuffer[1]);
+		jd->ringbuffer[1] = NULL;
+	}
+}
+
+static void
+mpd_jack_free(struct jack_data *jd)
+{
+	int i;
+
+	assert(jd != NULL);
+
+	mpd_jack_client_free(jd);
+
+	if (strcmp(jd->name, "mpd") != 0)
+		xfree(jd->name);
+
+	for ( i = ARRAY_SIZE(jd->output_ports); --i >= 0; ) {
+		if (!jd->output_ports[i])
+			continue;
+		xfree(jd->output_ports[i]);
+	}
+
+	free(jd);
+}
+
+static void
+mpd_jack_finish(void *data)
+{
+	struct jack_data *jd = data;
+	mpd_jack_free(jd);
+}
+
+static int
+mpd_jack_srate(mpd_unused jack_nframes_t rate, void *data)
+{
+	struct jack_data *jd = (struct jack_data *)data;
+	struct audio_format *audioFormat = jd->audio_format;
+
+	audioFormat->sample_rate = (int)jack_get_sample_rate(jd->client);
+
+	return 0;
+}
+
+static int
+mpd_jack_process(jack_nframes_t nframes, void *arg)
+{
+	struct jack_data *jd = (struct jack_data *) arg;
+	jack_default_audio_sample_t *out;
+	size_t available;
+
+	if (nframes <= 0)
+		return 0;
+
+	for (unsigned i = 0; i < 2; ++i) {
+		available = jack_ringbuffer_read_space(jd->ringbuffer[i]);
+		assert(available % sample_size == 0);
+		available /= sample_size;
+		if (available > nframes)
+			available = nframes;
+
+		out = jack_port_get_buffer(jd->ports[i], nframes);
+		jack_ringbuffer_read(jd->ringbuffer[i],
+				     (char *)out, available * sample_size);
+
+		while (available < nframes)
+			/* ringbuffer underrun, fill with silence */
+			out[available++] = 0.0;
+	}
+
+	return 0;
+}
+
+static void
+mpd_jack_shutdown(void *arg)
+{
+	struct jack_data *jd = (struct jack_data *) arg;
+	jd->shutdown = 1;
+}
+
+static void
+set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
+{
+	audio_format->sample_rate = jack_get_sample_rate(jd->client);
+	DEBUG("samplerate = %u\n", audio_format->sample_rate);
+	audio_format->channels = 2;
+
+	if (audio_format->bits != 16 && audio_format->bits != 24)
+		audio_format->bits = 24;
+
+	jd->bps = audio_format->channels
+		* sizeof(jack_default_audio_sample_t)
+		* audio_format->sample_rate;
+}
+
+static void
+mpd_jack_error(const char *msg)
+{
+	ERROR("jack: %s\n", msg);
+}
+
+static void *
+mpd_jack_init(struct audio_output *ao,
+	      mpd_unused const struct audio_format *audio_format,
+	      ConfigParam *param)
+{
+	struct jack_data *jd;
+	BlockParam *bp;
+	char *endptr;
+	int val;
+	char *cp = NULL;
+
+	jd = mpd_jack_new();
+	jd->ao = ao;
+
+	DEBUG("mpd_jack_init (pid=%d)\n", getpid());
+	if (param == NULL)
+		return jd;
+
+	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';
+		jd->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);
+
+		jd->output_ports[1] = xstrdup(cp);
+
+		if (strchr(cp,','))
+			FATAL("Only %d values are supported for '%s' "
+			      "at line %d\n",
+			      (int)ARRAY_SIZE(jd->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) {
+			jd->ringbuffer_size = val < 32768 ? 32768 : val;
+			DEBUG("ringbuffer_size=%d\n", jd->ringbuffer_size);
+		} else {
+			FATAL("%s is not a number; ringbuf_size=%d\n",
+			      bp->value, jd->ringbuffer_size);
+		}
+	}
+
+	if ( (bp = getBlockParam(param, "name"))
+	     && (strcmp(bp->value, "mpd") != 0) ) {
+		jd->name = xstrdup(bp->value);
+		DEBUG("name=%s\n", jd->name);
+	}
+
+	return jd;
+}
+
+static int
+mpd_jack_test_default_device(void)
+{
+	return 0;
+}
+
+static int
+mpd_jack_connect(struct jack_data *jd, struct audio_format *audio_format)
+{
+	const char **jports;
+	char *port_name;
+
+	jd->audio_format = audio_format;
+
+	if ( (jd->client = jack_client_new(jd->name)) == NULL ) {
+		ERROR("jack server not running?\n");
+		return -1;
+	}
+
+	jack_set_error_function(mpd_jack_error);
+	jack_set_process_callback(jd->client, mpd_jack_process, jd);
+	jack_set_sample_rate_callback(jd->client, mpd_jack_srate, jd);
+	jack_on_shutdown(jd->client, mpd_jack_shutdown, jd);
+
+	if ( jack_activate(jd->client) ) {
+		ERROR("cannot activate client\n");
+		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");
+		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");
+		return -1;
+	}
+
+	/*  hay que buscar que hay  */
+	if (!jd->output_ports[1] &&
+	    (jports = jack_get_ports(jd->client, NULL, NULL,
+				     JackPortIsPhysical | JackPortIsInput))) {
+		jd->output_ports[0] = jports[0];
+		jd->output_ports[1] = jports[1] ? jports[1] : jports[0];
+		DEBUG("output_ports: %s %s\n",
+		      jd->output_ports[0], jd->output_ports[1]);
+		free(jports);
+	}
+
+	if ( jd->output_ports[1] ) {
+		jd->ringbuffer[0] = jack_ringbuffer_create(jd->ringbuffer_size);
+		jd->ringbuffer[1] = jack_ringbuffer_create(jd->ringbuffer_size);
+		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(jd->name)));
+
+		sprintf(port_name, "%s:left", jd->name);
+		if ( (jack_connect(jd->client, port_name,
+				   jd->output_ports[0])) != 0 ) {
+			ERROR("%s is not a valid Jack Client / Port\n",
+			      jd->output_ports[0]);
+			free(port_name);
+			return -1;
+		}
+		sprintf(port_name, "%s:right", jd->name);
+		if ( (jack_connect(jd->client, port_name,
+				   jd->output_ports[1])) != 0 ) {
+			ERROR("%s is not a valid Jack Client / Port\n",
+			      jd->output_ports[1]);
+			free(port_name);
+			return -1;
+		}
+		free(port_name);
+	}
+
+	return 1;
+}
+
+static int
+mpd_jack_open(void *data, struct audio_format *audio_format)
+{
+	struct jack_data *jd = data;
+
+	assert(jd != NULL);
+
+	if (jd->client == NULL && mpd_jack_connect(jd, audio_format) < 0) {
+		mpd_jack_client_free(jd);
+		return -1;
+	}
+
+	set_audioformat(jd, audio_format);
+
+	return 0;
+}
+
+static void
+mpd_jack_close(mpd_unused void *data)
+{
+	/*mpd_jack_finish(audioOutput);*/
+}
+
+static void
+mpd_jack_cancel (mpd_unused void *data)
+{
+}
+
+static inline jack_default_audio_sample_t
+sample_16_to_jack(int16_t sample)
+{
+	return sample / (jack_default_audio_sample_t)(1 << (16 - 1));
+}
+
+static void
+mpd_jack_write_samples_16(struct jack_data *jd, const int16_t *src,
+			  unsigned num_samples)
+{
+	jack_default_audio_sample_t sample;
+
+	while (num_samples-- > 0) {
+		sample = sample_16_to_jack(*src++);
+		jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
+				      sizeof(sample));
+
+		sample = sample_16_to_jack(*src++);
+		jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
+				      sizeof(sample));
+	}
+}
+
+static inline jack_default_audio_sample_t
+sample_24_to_jack(int32_t sample)
+{
+	return sample / (jack_default_audio_sample_t)(1 << (24 - 1));
+}
+
+static void
+mpd_jack_write_samples_24(struct jack_data *jd, const int32_t *src,
+			  unsigned num_samples)
+{
+	jack_default_audio_sample_t sample;
+
+	while (num_samples-- > 0) {
+		sample = sample_24_to_jack(*src++);
+		jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
+				      sizeof(sample));
+
+		sample = sample_24_to_jack(*src++);
+		jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
+				      sizeof(sample));
+	}
+}
+
+static void
+mpd_jack_write_samples(struct jack_data *jd, const void *src,
+		       unsigned num_samples)
+{
+	switch (jd->audio_format->bits) {
+	case 16:
+		mpd_jack_write_samples_16(jd, (const int16_t*)src,
+					  num_samples);
+		break;
+
+	case 24:
+		mpd_jack_write_samples_24(jd, (const int32_t*)src,
+					  num_samples);
+		break;
+
+	default:
+		assert(false);
+	}
+}
+
+static int
+mpd_jack_play(void *data, const char *buff, size_t size)
+{
+	struct jack_data *jd = data;
+	const size_t frame_size = audio_format_frame_size(jd->audio_format);
+	size_t space, space1;
+
+	if (jd->shutdown) {
+		ERROR("Refusing to play, because there is no client thread.\n");
+		mpd_jack_client_free(jd);
+		audio_output_closed(jd->ao);
+		return 0;
+	}
+
+	assert(size % frame_size == 0);
+	size /= frame_size;
+	while (size > 0 && !jd->shutdown) {
+		space = jack_ringbuffer_write_space(jd->ringbuffer[0]);
+		space1 = jack_ringbuffer_write_space(jd->ringbuffer[1]);
+		if (space > space1)
+			/* send data symmetrically */
+			space = space1;
+
+		space /= sample_size;
+		if (space > 0) {
+			if (space > size)
+				space = size;
+
+			mpd_jack_write_samples(jd, buff, space);
+
+			buff += space * frame_size;
+			size -= space;
+		} else {
+			/* XXX do something more intelligent to
+			   synchronize */
+			my_usleep(10000);
+		}
+
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin jackPlugin = {
+	.name = "jack",
+	.test_default_device = mpd_jack_test_default_device,
+	.init = mpd_jack_init,
+	.finish = mpd_jack_finish,
+	.open = mpd_jack_open,
+	.play = mpd_jack_play,
+	.cancel = mpd_jack_cancel,
+	.close = mpd_jack_close,
+};
+
+#else /* HAVE JACK */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
+
+#endif /* HAVE_JACK */
diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c
new file mode 100644
index 00000000..70dd25f9
--- /dev/null
+++ b/src/output/mvp_plugin.c
@@ -0,0 +1,280 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_MVP
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.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 {
+	struct audio_output *audio_output;
+	struct audio_format audio_format;
+	int fd;
+} MvpData;
+
+static unsigned 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 const unsigned numfrequencies =
+	sizeof(pcmfrequencies) / sizeof(pcmfrequencies[0]);
+
+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 void *mvp_initDriver(mpd_unused struct audio_output *audio_output,
+			    mpd_unused const struct audio_format *audio_format,
+			    mpd_unused ConfigParam *param)
+{
+	MvpData *md = xmalloc(sizeof(MvpData));
+	md->audio_output = audio_output;
+	md->fd = -1;
+
+	return md;
+}
+
+static void mvp_finishDriver(void *data)
+{
+	MvpData *md = data;
+	free(md);
+}
+
+static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
+			    int big_endian, unsigned bits)
+{
+	unsigned iloop;
+	unsigned 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(void *data, struct audio_format *audioFormat)
+{
+	MvpData *md = data;
+	long long int stc = 0;
+	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->sample_rate, audioFormat->channels,
+			 0, audioFormat->bits);
+#else
+	mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
+			 1, audioFormat->bits);
+#endif
+	md->audio_format = *audioFormat;
+	return 0;
+}
+
+static void mvp_closeDevice(void *data)
+{
+	MvpData *md = data;
+	if (md->fd >= 0)
+		close(md->fd);
+	md->fd = -1;
+}
+
+static void mvp_dropBufferedAudio(void *data)
+{
+	MvpData *md = data;
+	if (md->fd >= 0) {
+		ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
+		close(md->fd);
+		md->fd = -1;
+		audio_output_closed(md->audio_output);
+	}
+}
+
+static int mvp_playAudio(void *data, const char *playChunk, size_t size)
+{
+	MvpData *md = data;
+	ssize_t ret;
+
+	/* reopen the device since it was closed by dropBufferedAudio */
+	if (md->fd < 0)
+		mvp_openDevice(md, &md->audio_format);
+
+	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(md);
+			return -1;
+		}
+		playChunk += ret;
+		size -= ret;
+	}
+	return 0;
+}
+
+const struct audio_output_plugin mvpPlugin = {
+	.name = "mvp",
+	.test_default_device = mvp_testDefault,
+	.init = mvp_initDriver,
+	.finish = mvp_finishDriver,
+	.open = mvp_openDevice,
+	.play = mvp_playAudio,
+	.cancel = mvp_dropBufferedAudio,
+	.close = mvp_closeDevice,
+};
+
+#else /* HAVE_MVP */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
+#endif /* HAVE_MVP */
diff --git a/src/output/null_plugin.c b/src/output/null_plugin.c
new file mode 100644
index 00000000..ff3a9833
--- /dev/null
+++ b/src/output/null_plugin.c
@@ -0,0 +1,85 @@
+/* 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 "../output_api.h"
+#include "../timer.h"
+#include "../utils.h"
+
+struct null_data {
+	Timer *timer;
+};
+
+static void *null_initDriver(mpd_unused struct audio_output *audioOutput,
+			     mpd_unused const struct audio_format *audio_format,
+			     mpd_unused ConfigParam *param)
+{
+	struct null_data *nd = xmalloc(sizeof(*nd));
+	nd->timer = NULL;
+	return nd;
+}
+
+static int null_openDevice(void *data,
+			   struct audio_format *audio_format)
+{
+	struct null_data *nd = data;
+
+	nd->timer = timer_new(audio_format);
+	return 0;
+}
+
+static void null_closeDevice(void *data)
+{
+	struct null_data *nd = data;
+
+	if (nd->timer != NULL) {
+		timer_free(nd->timer);
+		nd->timer = NULL;
+	}
+}
+
+static int null_playAudio(void *data,
+			  mpd_unused const char *playChunk, size_t size)
+{
+	struct null_data *nd = data;
+	Timer *timer = nd->timer;
+
+	if (!timer->started)
+		timer_start(timer);
+	else
+		timer_sync(timer);
+
+	timer_add(timer, size);
+
+	return 0;
+}
+
+static void null_dropBufferedAudio(void *data)
+{
+	struct null_data *nd = data;
+
+	timer_reset(nd->timer);
+}
+
+const struct audio_output_plugin nullPlugin = {
+	.name = "null",
+	.init = null_initDriver,
+	.open = null_openDevice,
+	.play = null_playAudio,
+	.cancel = null_dropBufferedAudio,
+	.close = null_closeDevice,
+};
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
new file mode 100644
index 00000000..67ea11fe
--- /dev/null
+++ b/src/output/oss_plugin.c
@@ -0,0 +1,571 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_OSS
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.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;
+	struct audio_format audio_format;
+	int bitFormat;
+	int *supported[3];
+	int numSupported[3];
+	int *unsupported[3];
+	int numUnsupported[3];
+} OssData;
+
+enum oss_support {
+	OSS_SUPPORTED = 1,
+	OSS_UNSUPPORTED = 0,
+	OSS_UNKNOWN = -1,
+};
+
+enum oss_param {
+	OSS_RATE = 0,
+	OSS_CHANNELS = 1,
+	OSS_BITS = 2,
+};
+
+static enum oss_param
+getIndexForParam(unsigned param)
+{
+	enum oss_param idx = OSS_RATE;
+
+	switch (param) {
+	case SNDCTL_DSP_SPEED:
+		idx = OSS_RATE;
+		break;
+	case SNDCTL_DSP_CHANNELS:
+		idx = OSS_CHANNELS;
+		break;
+	case SNDCTL_DSP_SAMPLESIZE:
+		idx = OSS_BITS;
+		break;
+	}
+
+	return idx;
+}
+
+static int findSupportedParam(OssData * od, unsigned param, int val)
+{
+	int i;
+	enum oss_param idx = getIndexForParam(param);
+
+	for (i = 0; i < od->numSupported[idx]; i++) {
+		if (od->supported[idx][i] == val)
+			return 1;
+	}
+
+	return 0;
+}
+
+static int canConvert(int idx, int val)
+{
+	switch (idx) {
+	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, unsigned param, int val)
+{
+	int i;
+	enum oss_param idx = getIndexForParam(param);
+	int ret = -1;
+	int least = val;
+	int diff;
+
+	for (i = 0; i < od->numSupported[idx]; i++) {
+		diff = od->supported[idx][i] - val;
+		if (diff < 0)
+			diff = -diff;
+		if (diff < least) {
+			if (!canConvert(idx, od->supported[idx][i])) {
+				continue;
+			}
+			least = diff;
+			ret = od->supported[idx][i];
+		}
+	}
+
+	return ret;
+}
+
+static int findUnsupportedParam(OssData * od, unsigned param, int val)
+{
+	int i;
+	enum oss_param idx = getIndexForParam(param);
+
+	for (i = 0; i < od->numUnsupported[idx]; i++) {
+		if (od->unsupported[idx][i] == val)
+			return 1;
+	}
+
+	return 0;
+}
+
+static void addSupportedParam(OssData * od, unsigned param, int val)
+{
+	enum oss_param idx = getIndexForParam(param);
+
+	od->numSupported[idx]++;
+	od->supported[idx] = xrealloc(od->supported[idx],
+				      od->numSupported[idx] * sizeof(int));
+	od->supported[idx][od->numSupported[idx] - 1] = val;
+}
+
+static void addUnsupportedParam(OssData * od, unsigned param, int val)
+{
+	enum oss_param idx = getIndexForParam(param);
+
+	od->numUnsupported[idx]++;
+	od->unsupported[idx] = xrealloc(od->unsupported[idx],
+					od->numUnsupported[idx] *
+					sizeof(int));
+	od->unsupported[idx][od->numUnsupported[idx] - 1] = val;
+}
+
+static void removeSupportedParam(OssData * od, unsigned param, int val)
+{
+	int i;
+	int j = 0;
+	enum oss_param idx = getIndexForParam(param);
+
+	for (i = 0; i < od->numSupported[idx] - 1; i++) {
+		if (od->supported[idx][i] == val)
+			j = 1;
+		od->supported[idx][i] = od->supported[idx][i + j];
+	}
+
+	od->numSupported[idx]--;
+	od->supported[idx] = xrealloc(od->supported[idx],
+				      od->numSupported[idx] * sizeof(int));
+}
+
+static void removeUnsupportedParam(OssData * od, unsigned param, int val)
+{
+	int i;
+	int j = 0;
+	enum oss_param idx = getIndexForParam(param);
+
+	for (i = 0; i < od->numUnsupported[idx] - 1; i++) {
+		if (od->unsupported[idx][i] == val)
+			j = 1;
+		od->unsupported[idx][i] = od->unsupported[idx][i + j];
+	}
+
+	od->numUnsupported[idx]--;
+	od->unsupported[idx] = xrealloc(od->unsupported[idx],
+					od->numUnsupported[idx] *
+					sizeof(int));
+}
+
+static enum oss_support
+isSupportedParam(OssData * od, unsigned 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, unsigned param, int val)
+{
+	enum oss_support 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, unsigned param, int val)
+{
+	enum oss_support 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 void *oss_open_default(ConfigParam *param)
+{
+	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) {
+			OssData *od = newOssData();
+			od->device = default_devices[i];
+			return od;
+		}
+	}
+
+	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\n", dev, strerror(err[i]));
+		}
+	}
+	exit(EXIT_FAILURE);
+	return NULL; /* some compilers can be dumb... */
+}
+
+static void *oss_initDriver(mpd_unused struct audio_output *audioOutput,
+			    mpd_unused const struct audio_format *audio_format,
+			    ConfigParam * param)
+{
+	if (param) {
+		BlockParam *bp = getBlockParam(param, "device");
+		if (bp) {
+			OssData *od = newOssData();
+			od->device = bp->value;
+			return od;
+		}
+	}
+	return oss_open_default(param);
+}
+
+static void oss_finishDriver(void *data)
+{
+	OssData *od = data;
+
+	freeOssData(od);
+}
+
+static int setParam(OssData * od, unsigned param, int *value)
+{
+	int val = *value;
+	int copy;
+	enum oss_support 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(OssData *od)
+{
+	int tmp;
+
+	if ((od->fd = open(od->device, O_WRONLY)) < 0) {
+		ERROR("Error opening OSS device \"%s\": %s\n", od->device,
+		      strerror(errno));
+		goto fail;
+	}
+
+	tmp = od->audio_format.channels;
+	if (setParam(od, SNDCTL_DSP_CHANNELS, &tmp)) {
+		ERROR("OSS device \"%s\" does not support %u channels: %s\n",
+		      od->device, od->audio_format.channels, strerror(errno));
+		goto fail;
+	}
+	od->audio_format.channels = tmp;
+
+	tmp = od->audio_format.sample_rate;
+	if (setParam(od, SNDCTL_DSP_SPEED, &tmp)) {
+		ERROR("OSS device \"%s\" does not support %u Hz audio: %s\n",
+		      od->device, od->audio_format.sample_rate,
+		      strerror(errno));
+		goto fail;
+	}
+	od->audio_format.sample_rate = tmp;
+
+	switch (od->audio_format.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 %u bit audio: %s\n",
+		      od->device, tmp, strerror(errno));
+		goto fail;
+	}
+
+	return 0;
+
+fail:
+	oss_close(od);
+	return -1;
+}
+
+static int oss_openDevice(void *data,
+			  struct audio_format *audioFormat)
+{
+	int ret;
+	OssData *od = data;
+
+	od->audio_format = *audioFormat;
+
+	if ((ret = oss_open(od)) < 0)
+		return ret;
+
+	*audioFormat = od->audio_format;
+
+	DEBUG("oss device \"%s\" will be playing %u bit %u channel audio at "
+	      "%u Hz\n", od->device,
+	      od->audio_format.bits, od->audio_format.channels,
+	      od->audio_format.sample_rate);
+
+	return ret;
+}
+
+static void oss_closeDevice(void *data)
+{
+	OssData *od = data;
+
+	oss_close(od);
+}
+
+static void oss_dropBufferedAudio(void *data)
+{
+	OssData *od = data;
+
+	if (od->fd >= 0) {
+		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+		oss_close(od);
+	}
+}
+
+static int oss_playAudio(void *data,
+			 const char *playChunk, size_t size)
+{
+	OssData *od = data;
+	ssize_t ret;
+
+	/* reopen the device since it was closed by dropBufferedAudio */
+	if (od->fd < 0 && oss_open(od) < 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(od);
+			return -1;
+		}
+		playChunk += ret;
+		size -= ret;
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin ossPlugin = {
+	.name = "oss",
+	.test_default_device = oss_testDefault,
+	.init = oss_initDriver,
+	.finish = oss_finishDriver,
+	.open = oss_openDevice,
+	.play = oss_playAudio,
+	.cancel = oss_dropBufferedAudio,
+	.close = oss_closeDevice,
+};
+
+#else /* HAVE OSS */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
+#endif /* HAVE_OSS */
diff --git a/src/output/osx_plugin.c b/src/output/osx_plugin.c
new file mode 100644
index 00000000..65060cc8
--- /dev/null
+++ b/src/output/osx_plugin.c
@@ -0,0 +1,368 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_OSX
+
+#include <AudioUnit/AudioUnit.h>
+
+#include "../utils.h"
+#include "../log.h"
+
+typedef struct _OsxData {
+	AudioUnit au;
+	pthread_mutex_t mutex;
+	pthread_cond_t condition;
+	char *buffer;
+	size_t bufferSize;
+	size_t pos;
+	size_t 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(struct audio_output *audioOutput,
+			  mpd_unused const struct audio_format *audio_format,
+			  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(struct audio_output *audioOutput)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+	freeOsxData(od);
+}
+
+static void osx_dropBufferedAudio(struct audio_output *audioOutput)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+
+	pthread_mutex_lock(&od->mutex);
+	od->len = 0;
+	pthread_mutex_unlock(&od->mutex);
+}
+
+static void osx_closeDevice(struct audio_output *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);
+}
+
+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];
+	size_t bufferSize = buffer->mDataByteSize;
+	size_t 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) {
+		size_t 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(struct audio_output *audioOutput,
+			  struct audio_format *audioFormat)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+	ComponentDescription desc;
+	Component comp;
+	AURenderCallbackStruct callback;
+	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->sample_rate;
+	streamDesc.mFormatID = kAudioFormatLinearPCM;
+	streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+#ifdef WORDS_BIGENDIAN
+	streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+
+	streamDesc.mBytesPerPacket = audio_format_frame_size(audioFormat);
+	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->sample_rate) *
+		audio_format_frame_size(audioFormat);
+	od->buffer = xrealloc(od->buffer, od->bufferSize);
+
+	od->pos = 0;
+	od->len = 0;
+
+	return 0;
+}
+
+static int osx_play(struct audio_output *audioOutput,
+		    const char *playChunk, size_t size)
+{
+	OsxData *od = (OsxData *) audioOutput->data;
+	size_t bytesToCopy;
+	size_t 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) {
+			size_t 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;
+}
+
+const struct audio_output_plugin osxPlugin = {
+	.name = "osx",
+	.test_default_device = osx_testDefault,
+	.init = osx_initDriver,
+	.finish = osx_finishDriver,
+	.open = osx_openDevice,
+	.play = osx_play,
+	.cancel = osx_dropBufferedAudio,
+	.close = osx_closeDevice,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
+#endif
diff --git a/src/output/pulse_plugin.c b/src/output/pulse_plugin.c
new file mode 100644
index 00000000..b1ce3d04
--- /dev/null
+++ b/src/output/pulse_plugin.c
@@ -0,0 +1,218 @@
+/* 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 "../output_api.h"
+
+#ifdef HAVE_PULSE
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+
+#define MPD_PULSE_NAME "mpd"
+#define CONN_ATTEMPT_INTERVAL 60
+
+typedef struct _PulseData {
+	struct audio_output *ao;
+
+	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 void *pulse_initDriver(struct audio_output *ao,
+			      mpd_unused const struct audio_format *audio_format,
+			      ConfigParam * param)
+{
+	BlockParam *server = NULL;
+	BlockParam *sink = NULL;
+	PulseData *pd;
+
+	if (param) {
+		server = getBlockParam(param, "server");
+		sink = getBlockParam(param, "sink");
+	}
+
+	pd = newPulseData();
+	pd->ao = ao;
+	pd->server = server ? xstrdup(server->value) : NULL;
+	pd->sink = sink ? xstrdup(sink->value) : NULL;
+
+	return pd;
+}
+
+static void pulse_finishDriver(void *data)
+{
+	PulseData *pd = data;
+
+	freePulseData(pd);
+}
+
+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(void *data,
+			    struct audio_format *audioFormat)
+{
+	PulseData *pd = data;
+	pa_sample_spec ss;
+	time_t t;
+	int error;
+
+	t = time(NULL);
+
+	if (pd->connAttempts != 0 &&
+	    (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
+		return -1;
+
+	pd->connAttempts++;
+	pd->lastAttempt = t;
+
+	/* MPD doesn't support the other pulseaudio sample formats, so
+	   we just force MPD to send us everything as 16 bit */
+	audioFormat->bits = 16;
+
+	ss.format = PA_SAMPLE_S16NE;
+	ss.rate = audioFormat->sample_rate;
+	ss.channels = audioFormat->channels;
+
+	pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
+			      pd->sink, audio_output_get_name(pd->ao),
+			      &ss, NULL, NULL,
+			      &error);
+	if (!pd->s) {
+		ERROR("Cannot connect to server in PulseAudio output "
+		      "\"%s\" (attempt %i): %s\n",
+		      audio_output_get_name(pd->ao),
+		      pd->connAttempts, pa_strerror(error));
+		return -1;
+	}
+
+	pd->connAttempts = 0;
+
+	DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
+	      "channel audio at %i Hz\n",
+	      audio_output_get_name(pd->ao),
+	      audioFormat->bits,
+	      audioFormat->channels, audioFormat->sample_rate);
+
+	return 0;
+}
+
+static void pulse_dropBufferedAudio(void *data)
+{
+	PulseData *pd = data;
+	int error;
+
+	if (pa_simple_flush(pd->s, &error) < 0)
+		WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
+			audio_output_get_name(pd->ao),
+			pa_strerror(error));
+}
+
+static void pulse_closeDevice(void *data)
+{
+	PulseData *pd = data;
+
+	if (pd->s) {
+		pa_simple_drain(pd->s, NULL);
+		pa_simple_free(pd->s);
+	}
+}
+
+static int pulse_playAudio(void *data,
+			   const char *playChunk, size_t size)
+{
+	PulseData *pd = data;
+	int error;
+
+	if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
+		ERROR("PulseAudio output \"%s\" disconnecting due to write "
+		      "error: %s\n",
+		      audio_output_get_name(pd->ao),
+		      pa_strerror(error));
+		pulse_closeDevice(pd);
+		return -1;
+	}
+
+	return 0;
+}
+
+const struct audio_output_plugin pulsePlugin = {
+	.name = "pulse",
+	.test_default_device = pulse_testDefault,
+	.init = pulse_initDriver,
+	.finish = pulse_finishDriver,
+	.open = pulse_openDevice,
+	.play = pulse_playAudio,
+	.cancel = pulse_dropBufferedAudio,
+	.close = pulse_closeDevice,
+};
+
+#else /* HAVE_PULSE */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
+#endif /* HAVE_PULSE */
diff --git a/src/output/shout_mp3.c b/src/output/shout_mp3.c
new file mode 100644
index 00000000..e57b0c38
--- /dev/null
+++ b/src/output/shout_mp3.c
@@ -0,0 +1,188 @@
+/* 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 "shout_plugin.h"
+
+#ifdef HAVE_SHOUT_MP3
+
+#include "../utils.h"
+#include <lame/lame.h>
+
+struct lame_data {
+	lame_global_flags *gfp;
+};
+
+
+static int shout_mp3_encoder_init(struct shout_data *sd)
+{
+	struct lame_data *ld;
+
+	if (NULL == (ld = xmalloc(sizeof(*ld))))
+		FATAL("error initializing lame encoder data\n");
+	sd->encoder_data = ld;
+
+	return 0;
+}
+
+static int shout_mp3_encoder_clear_encoder(struct shout_data *sd)
+{
+	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+	struct shout_buffer *buf = &sd->buf;
+	int ret;
+
+	if ((ret = lame_encode_flush(ld->gfp, buf->data + buf->len,
+				     buf->len)) < 0)
+		ERROR("error flushing lame buffers\n");
+
+	return (ret > 0);
+}
+
+static void shout_mp3_encoder_finish(struct shout_data *sd)
+{
+	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+
+	lame_close(ld->gfp);
+	ld->gfp = NULL;
+}
+
+static int shout_mp3_encoder_init_encoder(struct shout_data *sd)
+{
+	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+
+	if (NULL == (ld->gfp = lame_init())) {
+		ERROR("error initializing lame encoder for shout\n");
+		return -1;
+	}
+
+	if (sd->quality >= -1.0) {
+		if (0 != lame_set_VBR(ld->gfp, vbr_rh)) {
+			ERROR("error setting lame VBR mode\n");
+			return -1;
+		}
+		if (0 != lame_set_VBR_q(ld->gfp, sd->quality)) {
+			ERROR("error setting lame VBR quality\n");
+			return -1;
+		}
+	} else {
+		if (0 != lame_set_brate(ld->gfp, sd->bitrate)) {
+			ERROR("error setting lame bitrate\n");
+			return -1;
+		}
+	}
+
+	if (0 != lame_set_num_channels(ld->gfp,
+				       sd->audio_format.channels)) {
+		ERROR("error setting lame num channels\n");
+		return -1;
+	}
+
+	if (0 != lame_set_in_samplerate(ld->gfp,
+					sd->audio_format.sample_rate)) {
+		ERROR("error setting lame sample rate\n");
+		return -1;
+	}
+
+	if (0 > lame_init_params(ld->gfp))
+		FATAL("error initializing lame params\n");
+
+	return 0;
+}
+
+static int shout_mp3_encoder_send_metadata(struct shout_data *sd,
+					   char * song, size_t size)
+{
+	char artist[size];
+	char title[size];
+	int i;
+	struct tag *tag = sd->tag;
+
+	strncpy(artist, "", size);
+	strncpy(title, "", size);
+
+	for (i = 0; i < tag->numOfItems; i++) {
+		switch (tag->items[i]->type) {
+		case TAG_ITEM_ARTIST:
+			strncpy(artist, tag->items[i]->value, size);
+			break;
+		case TAG_ITEM_TITLE:
+			strncpy(title, tag->items[i]->value, size);
+			break;
+
+		default:
+			break;
+		}
+	}
+	snprintf(song, size, "%s - %s", title, artist);
+
+	return 1;
+}
+
+static int shout_mp3_encoder_encode(struct shout_data *sd,
+				    const char * chunk, size_t len)
+{
+	unsigned int i;
+	int j;
+	float (*lamebuf)[2];
+	struct shout_buffer *buf = &(sd->buf);
+	unsigned int samples;
+	int bytes = audio_format_sample_size(&sd->audio_format);
+	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+	int bytes_out;
+
+	samples = len / (bytes * sd->audio_format.channels);
+	/* rough estimate, from lame.h */
+	lamebuf = xmalloc(sizeof(float) * (1.25 * samples + 7200));
+
+	/* this is for only 16-bit audio */
+
+	for (i = 0; i < samples; i++) {
+		for (j = 0; j < sd->audio_format.channels; j++) {
+			lamebuf[j][i] = *((const int16_t *) chunk);
+			chunk += bytes;
+		}
+	}
+
+	bytes_out = lame_encode_buffer_float(ld->gfp, lamebuf[0], lamebuf[1],
+					     samples, buf->data,
+					     sizeof(buf->data) - buf->len);
+	free(lamebuf);
+
+	if (0 > bytes_out) {
+		ERROR("error encoding lame buffer for shout\n");
+		lame_close(ld->gfp);
+		ld->gfp = NULL;
+		return -1;
+	} else
+		buf->len = bytes_out; /* signed to unsigned conversion */
+
+	return 0;
+}
+
+const struct shout_encoder_plugin shout_mp3_encoder = {
+	"mp3",
+	SHOUT_FORMAT_MP3,
+
+	shout_mp3_encoder_clear_encoder,
+	shout_mp3_encoder_encode,
+	shout_mp3_encoder_finish,
+	shout_mp3_encoder_init,
+	shout_mp3_encoder_init_encoder,
+	shout_mp3_encoder_send_metadata,
+};
+
+#endif
diff --git a/src/output/shout_ogg.c b/src/output/shout_ogg.c
new file mode 100644
index 00000000..248ffd1e
--- /dev/null
+++ b/src/output/shout_ogg.c
@@ -0,0 +1,306 @@
+/* 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 "shout_plugin.h"
+
+#ifdef HAVE_SHOUT_OGG
+
+#include "../utils.h"
+#include <vorbis/vorbisenc.h>
+
+struct ogg_vorbis_data {
+	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;
+};
+
+static void add_tag(struct ogg_vorbis_data *od, const char *name, char *value)
+{
+	if (value) {
+		union {
+			const char *in;
+			char *out;
+		} u = { .in = name };
+		vorbis_comment_add_tag(&od->vc, u.out, value);
+	}
+}
+
+static void copy_tag_to_vorbis_comment(struct shout_data *sd)
+{
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+	if (sd->tag) {
+		int i;
+
+		for (i = 0; i < sd->tag->numOfItems; i++) {
+			switch (sd->tag->items[i]->type) {
+			case TAG_ITEM_ARTIST:
+				add_tag(od, "ARTIST", sd->tag->items[i]->value);
+				break;
+			case TAG_ITEM_ALBUM:
+				add_tag(od, "ALBUM", sd->tag->items[i]->value);
+				break;
+			case TAG_ITEM_TITLE:
+				add_tag(od, "TITLE", sd->tag->items[i]->value);
+				break;
+
+			default:
+				break;
+			}
+		}
+	}
+}
+
+static int copy_ogg_buffer_to_shout_buffer(ogg_page *og,
+					   struct shout_buffer *buf)
+{
+	if (sizeof(buf->data) - buf->len >= (size_t)og->header_len) {
+		memcpy(buf->data + buf->len,
+		       og->header, og->header_len);
+		buf->len += og->header_len;
+	} else {
+		ERROR("%s: not enough buffer space!\n", __func__);
+		return -1;
+	}
+
+	if (sizeof(buf->data) - buf->len >= (size_t)og->body_len) {
+		memcpy(buf->data + buf->len,
+		       og->body, og->body_len);
+		buf->len += og->body_len;
+	} else {
+		ERROR("%s: not enough buffer space!\n", __func__);
+		return -1;
+	}
+
+	return 0;
+}
+
+static int flush_ogg_buffer(struct shout_data *sd)
+{
+	struct shout_buffer *buf = &sd->buf;
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+	int ret = 0;
+
+	if (ogg_stream_flush(&od->os, &od->og))
+		ret = copy_ogg_buffer_to_shout_buffer(&od->og, buf);
+
+	return ret;
+}
+
+static int send_ogg_vorbis_header(struct shout_data *sd)
+{
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+	vorbis_analysis_headerout(&od->vd, &od->vc,
+				  &od->header_main,
+				  &od->header_comments,
+				  &od->header_codebooks);
+
+	ogg_stream_packetin(&od->os, &od->header_main);
+	ogg_stream_packetin(&od->os, &od->header_comments);
+	ogg_stream_packetin(&od->os, &od->header_codebooks);
+
+	return flush_ogg_buffer(sd);
+}
+
+static void finish_encoder(struct ogg_vorbis_data *od)
+{
+	vorbis_analysis_wrote(&od->vd, 0);
+
+	while (vorbis_analysis_blockout(&od->vd, &od->vb) == 1) {
+		vorbis_analysis(&od->vb, NULL);
+		vorbis_bitrate_addblock(&od->vb);
+		while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
+			ogg_stream_packetin(&od->os, &od->op);
+		}
+	}
+}
+
+static int shout_ogg_encoder_clear_encoder(struct shout_data *sd)
+{
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+	int ret;
+
+	finish_encoder(od);
+	if ((ret = ogg_stream_pageout(&od->os, &od->og)))
+		copy_ogg_buffer_to_shout_buffer(&od->og, &sd->buf);
+
+	vorbis_comment_clear(&od->vc);
+	ogg_stream_clear(&od->os);
+	vorbis_block_clear(&od->vb);
+	vorbis_dsp_clear(&od->vd);
+	vorbis_info_clear(&od->vi);
+
+	return ret;
+}
+
+static void shout_ogg_encoder_finish(struct shout_data *sd)
+{
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+	if (od) {
+		free(od);
+		sd->encoder_data = NULL;
+	}
+}
+
+static int shout_ogg_encoder_init(struct shout_data *sd)
+{
+	struct ogg_vorbis_data *od;
+
+	if (NULL == (od = xmalloc(sizeof(*od))))
+		FATAL("error initializing ogg vorbis encoder data\n");
+	sd->encoder_data = od;
+
+	return 0;
+}
+
+static int reinit_encoder(struct shout_data *sd)
+{
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+	vorbis_info_init(&od->vi);
+
+	if (sd->quality >= -1.0) {
+		if (0 != vorbis_encode_init_vbr(&od->vi,
+						sd->audio_format.channels,
+						sd->audio_format.sample_rate,
+						sd->quality * 0.1)) {
+			ERROR("error initializing vorbis vbr\n");
+			vorbis_info_clear(&od->vi);
+			return -1;
+		}
+	} else {
+		if (0 != vorbis_encode_init(&od->vi,
+					    sd->audio_format.channels,
+					    sd->audio_format.sample_rate, -1.0,
+					    sd->bitrate * 1000, -1.0)) {
+			ERROR("error initializing vorbis encoder\n");
+			vorbis_info_clear(&od->vi);
+			return -1;
+		}
+	}
+
+	vorbis_analysis_init(&od->vd, &od->vi);
+	vorbis_block_init(&od->vd, &od->vb);
+	ogg_stream_init(&od->os, rand());
+	vorbis_comment_init(&od->vc);
+
+	return 0;
+}
+
+static int shout_ogg_encoder_init_encoder(struct shout_data *sd)
+{
+	if (reinit_encoder(sd))
+		return -1;
+
+	if (send_ogg_vorbis_header(sd)) {
+		ERROR("error sending ogg vorbis header for shout\n");
+		return -1;
+	}
+
+	return 0;
+}
+
+static int shout_ogg_encoder_send_metadata(struct shout_data *sd,
+					   mpd_unused char * song,
+					   mpd_unused size_t size)
+{
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+	shout_ogg_encoder_clear_encoder(sd);
+	if (reinit_encoder(sd))
+		return 0;
+
+	copy_tag_to_vorbis_comment(sd);
+
+	vorbis_analysis_headerout(&od->vd, &od->vc,
+				  &od->header_main,
+				  &od->header_comments,
+				  &od->header_codebooks);
+
+	ogg_stream_packetin(&od->os, &od->header_main);
+	ogg_stream_packetin(&od->os, &od->header_comments);
+	ogg_stream_packetin(&od->os, &od->header_codebooks);
+
+	flush_ogg_buffer(sd);
+
+	return 0;
+}
+
+static int shout_ogg_encoder_encode(struct shout_data *sd,
+				    const char *chunk, size_t size)
+{
+	struct shout_buffer *buf = &sd->buf;
+	unsigned int i;
+	int j;
+	float **vorbbuf;
+	unsigned int samples;
+	int bytes = audio_format_sample_size(&sd->audio_format);
+	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+	samples = size / (bytes * sd->audio_format.channels);
+	vorbbuf = vorbis_analysis_buffer(&od->vd, samples);
+
+	/* this is for only 16-bit audio */
+
+	for (i = 0; i < samples; i++) {
+		for (j = 0; j < sd->audio_format.channels; j++) {
+			vorbbuf[j][i] = (*((const int16_t *) chunk)) / 32768.0;
+			chunk += bytes;
+		}
+	}
+
+	vorbis_analysis_wrote(&od->vd, samples);
+
+	while (1 == vorbis_analysis_blockout(&od->vd, &od->vb)) {
+		vorbis_analysis(&od->vb, NULL);
+		vorbis_bitrate_addblock(&od->vb);
+
+		while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
+			ogg_stream_packetin(&od->os, &od->op);
+		}
+	}
+
+	if (ogg_stream_pageout(&od->os, &od->og))
+		copy_ogg_buffer_to_shout_buffer(&od->og, buf);
+
+	return 0;
+}
+
+const struct shout_encoder_plugin shout_ogg_encoder = {
+	"ogg",
+	SHOUT_FORMAT_VORBIS,
+
+	shout_ogg_encoder_clear_encoder,
+	shout_ogg_encoder_encode,
+	shout_ogg_encoder_finish,
+	shout_ogg_encoder_init,
+	shout_ogg_encoder_init_encoder,
+	shout_ogg_encoder_send_metadata,
+};
+
+#endif
diff --git a/src/output/shout_plugin.c b/src/output/shout_plugin.c
new file mode 100644
index 00000000..dd909bda
--- /dev/null
+++ b/src/output/shout_plugin.c
@@ -0,0 +1,596 @@
+/* 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 "shout_plugin.h"
+
+#ifdef HAVE_SHOUT
+
+#include "../utils.h"
+
+#include <assert.h>
+
+#define CONN_ATTEMPT_INTERVAL 60
+#define DEFAULT_CONN_TIMEOUT  2
+
+static int shout_init_count;
+
+static const struct shout_encoder_plugin *const shout_encoder_plugins[] = {
+#ifdef HAVE_SHOUT_MP3
+	&shout_mp3_encoder,
+#endif
+#ifdef HAVE_SHOUT_OGG
+	&shout_ogg_encoder,
+#endif
+	NULL
+};
+
+static const struct shout_encoder_plugin *
+shout_encoder_plugin_get(const char *name)
+{
+	unsigned i;
+
+	for (i = 0; shout_encoder_plugins[i] != NULL; ++i)
+		if (strcmp(shout_encoder_plugins[i]->name, name) == 0)
+			return shout_encoder_plugins[i];
+
+	return NULL;
+}
+
+static struct shout_data *new_shout_data(void)
+{
+	struct shout_data *ret = xmalloc(sizeof(*ret));
+
+	ret->shout_conn = shout_new();
+	ret->shout_meta = shout_metadata_new();
+	ret->opened = 0;
+	ret->tag = NULL;
+	ret->tag_to_send = 0;
+	ret->bitrate = -1;
+	ret->quality = -2.0;
+	ret->timeout = DEFAULT_CONN_TIMEOUT;
+	ret->conn_attempts = 0;
+	ret->last_attempt = 0;
+	ret->timer = NULL;
+	ret->buf.len = 0;
+
+	return ret;
+}
+
+static void free_shout_data(struct shout_data *sd)
+{
+	if (sd->shout_meta)
+		shout_metadata_free(sd->shout_meta);
+	if (sd->shout_conn)
+		shout_free(sd->shout_conn);
+	if (sd->tag)
+		tag_free(sd->tag);
+	if (sd->timer)
+		timer_free(sd->timer);
+
+	free(sd);
+}
+
+#define check_block_param(name) {		  \
+		block_param = getBlockParam(param, name);	\
+		if (!block_param) {					\
+			FATAL("no \"%s\" defined for shout device defined at line " \
+			      "%i\n", name, param->line);		\
+		}							\
+	}
+
+static void *my_shout_init_driver(struct audio_output *audio_output,
+				  const struct audio_format *audio_format,
+				  ConfigParam *param)
+{
+	struct shout_data *sd;
+	char *test;
+	int port;
+	char *host;
+	char *mount;
+	char *passwd;
+	const char *encoding;
+	unsigned protocol;
+	const char *user;
+	char *name;
+	BlockParam *block_param;
+	int public;
+
+	sd = new_shout_data();
+	sd->audio_output = audio_output;
+
+	if (shout_init_count == 0)
+		shout_init();
+
+	shout_init_count++;
+
+	check_block_param("host");
+	host = block_param->value;
+
+	check_block_param("mount");
+	mount = block_param->value;
+
+	check_block_param("port");
+
+	port = strtol(block_param->value, &test, 10);
+
+	if (*test != '\0' || port <= 0) {
+		FATAL("shout port \"%s\" is not a positive integer, line %i\n",
+		      block_param->value, block_param->line);
+	}
+
+	check_block_param("password");
+	passwd = block_param->value;
+
+	check_block_param("name");
+	name = block_param->value;
+
+	public = getBoolBlockParam(param, "public", 1);
+	if (public == CONF_BOOL_UNSET)
+		public = 0;
+
+	block_param = getBlockParam(param, "user");
+	if (block_param)
+		user = block_param->value;
+	else
+		user = "source";
+
+	block_param = getBlockParam(param, "quality");
+
+	if (block_param) {
+		int line = block_param->line;
+
+		sd->quality = strtod(block_param->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", block_param->value,
+			      block_param->line);
+		}
+
+		block_param = getBlockParam(param, "bitrate");
+
+		if (block_param) {
+			FATAL("quality (line %i) and bitrate (line %i) are "
+			      "both defined for shout output\n", line,
+			      block_param->line);
+		}
+	} else {
+		block_param = getBlockParam(param, "bitrate");
+
+		if (!block_param) {
+			FATAL("neither bitrate nor quality defined for shout "
+			      "output at line %i\n", param->line);
+		}
+
+		sd->bitrate = strtol(block_param->value, &test, 10);
+
+		if (*test != '\0' || sd->bitrate <= 0) {
+			FATAL("bitrate at line %i should be a positive integer "
+			      "\n", block_param->line);
+		}
+	}
+
+	check_block_param("format");
+
+	assert(audio_format != NULL);
+	sd->audio_format = *audio_format;
+
+	block_param = getBlockParam(param, "encoding");
+	if (block_param) {
+		if (0 == strcmp(block_param->value, "mp3"))
+			encoding = block_param->value;
+		else if (0 == strcmp(block_param->value, "ogg"))
+			encoding = block_param->value;
+		else
+			FATAL("shout encoding \"%s\" is not \"ogg\" or "
+			      "\"mp3\", line %i\n", block_param->value,
+			      block_param->line);
+	} else {
+		encoding = "ogg";
+	}
+
+	sd->encoder = shout_encoder_plugin_get(encoding);
+	if (sd->encoder == NULL)
+		FATAL("couldn't find shout encoder plugin for \"%s\" "
+		      "at line %i\n", encoding, block_param->line);
+
+	check_block_param("protocol");
+
+	block_param = getBlockParam(param, "protocol");
+	if (block_param) {
+		if (0 == strcmp(block_param->value, "shoutcast") &&
+		    0 != strcmp(encoding, "mp3"))
+			FATAL("you cannot stream \"%s\" to shoutcast, use mp3\n",
+			      encoding);
+		else if (0 == strcmp(block_param->value, "shoutcast"))
+			protocol = SHOUT_PROTOCOL_ICY;
+		else if (0 == strcmp(block_param->value, "icecast1"))
+			protocol = SHOUT_PROTOCOL_XAUDIOCAST;
+		else if (0 == strcmp(block_param->value, "icecast2"))
+			protocol = SHOUT_PROTOCOL_HTTP;
+		else
+			FATAL("shout protocol \"%s\" is not \"shoutcast\" or "
+			      "\"icecast1\"or "
+			      "\"icecast2\", line %i\n", block_param->value,
+			      block_param->line);
+	} else {
+		protocol = SHOUT_PROTOCOL_HTTP;
+	}
+
+	if (shout_set_host(sd->shout_conn, host) != SHOUTERR_SUCCESS ||
+	    shout_set_port(sd->shout_conn, port) != SHOUTERR_SUCCESS ||
+	    shout_set_password(sd->shout_conn, passwd) != SHOUTERR_SUCCESS ||
+	    shout_set_mount(sd->shout_conn, mount) != SHOUTERR_SUCCESS ||
+	    shout_set_name(sd->shout_conn, name) != SHOUTERR_SUCCESS ||
+	    shout_set_user(sd->shout_conn, user) != SHOUTERR_SUCCESS ||
+	    shout_set_public(sd->shout_conn, public) != SHOUTERR_SUCCESS ||
+	    shout_set_nonblocking(sd->shout_conn, 1) != SHOUTERR_SUCCESS ||
+	    shout_set_format(sd->shout_conn, sd->encoder->shout_format)
+	    != SHOUTERR_SUCCESS ||
+	    shout_set_protocol(sd->shout_conn, protocol) != SHOUTERR_SUCCESS ||
+	    shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
+		FATAL("error configuring shout defined at line %i: %s\n",
+		      param->line, shout_get_error(sd->shout_conn));
+	}
+
+	/* optional paramters */
+	block_param = getBlockParam(param, "timeout");
+	if (block_param) {
+		sd->timeout = (int)strtol(block_param->value, &test, 10);
+		if (*test != '\0' || sd->timeout <= 0) {
+			FATAL("shout timeout is not a positive integer, "
+			      "line %i\n", block_param->line);
+		}
+	}
+
+	block_param = getBlockParam(param, "genre");
+	if (block_param && shout_set_genre(sd->shout_conn, block_param->value)) {
+		FATAL("error configuring shout defined at line %i: %s\n",
+		      param->line, shout_get_error(sd->shout_conn));
+	}
+
+	block_param = getBlockParam(param, "description");
+	if (block_param && shout_set_description(sd->shout_conn,
+						 block_param->value)) {
+		FATAL("error configuring shout defined at line %i: %s\n",
+		      param->line, shout_get_error(sd->shout_conn));
+	}
+
+	{
+		char temp[11];
+		memset(temp, 0, sizeof(temp));
+
+		snprintf(temp, sizeof(temp), "%u", sd->audio_format.channels);
+		shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
+
+		snprintf(temp, sizeof(temp), "%u", sd->audio_format.sample_rate);
+
+		shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
+
+		if (sd->quality >= -1.0) {
+			snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
+			shout_set_audio_info(sd->shout_conn, SHOUT_AI_QUALITY,
+					     temp);
+		} else {
+			snprintf(temp, sizeof(temp), "%d", sd->bitrate);
+			shout_set_audio_info(sd->shout_conn, SHOUT_AI_BITRATE,
+					     temp);
+		}
+	}
+
+	if (sd->encoder->init_func(sd) != 0)
+		FATAL("shout: encoder plugin '%s' failed to initialize\n",
+		      sd->encoder->name);
+
+	return sd;
+}
+
+static int handle_shout_error(struct shout_data *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->shout_conn),
+		      shout_get_port(sd->shout_conn),
+		      shout_get_error(sd->shout_conn));
+		sd->shout_error = 1;
+		return -1;
+	default:
+		ERROR("shout: connection to %s:%i error: %s\n",
+		      shout_get_host(sd->shout_conn),
+		      shout_get_port(sd->shout_conn),
+		      shout_get_error(sd->shout_conn));
+		sd->shout_error = 1;
+		return -1;
+	}
+
+	return 0;
+}
+
+static int write_page(struct shout_data *sd)
+{
+	int err;
+
+	if (sd->buf.len == 0)
+		return 0;
+
+	shout_sync(sd->shout_conn);
+	err = shout_send(sd->shout_conn, sd->buf.data, sd->buf.len);
+	if (handle_shout_error(sd, err) < 0)
+		return -1;
+	sd->buf.len = 0;
+
+	return 0;
+}
+
+static void close_shout_conn(struct shout_data * sd)
+{
+	if (sd->opened) {
+		if (sd->encoder->clear_encoder_func(sd))
+			write_page(sd);
+	}
+
+	if (shout_get_connected(sd->shout_conn) != SHOUTERR_UNCONNECTED &&
+	    shout_close(sd->shout_conn) != SHOUTERR_SUCCESS) {
+		ERROR("problem closing connection to shout server: %s\n",
+		      shout_get_error(sd->shout_conn));
+	}
+
+	sd->opened = 0;
+}
+
+static void my_shout_finish_driver(void *data)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+
+	close_shout_conn(sd);
+
+	sd->encoder->finish_func(sd);
+	free_shout_data(sd);
+
+	shout_init_count--;
+
+	if (shout_init_count == 0)
+		shout_shutdown();
+}
+
+static void my_shout_drop_buffered_audio(void *data)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+	timer_reset(sd->timer);
+
+	/* needs to be implemented for shout */
+}
+
+static void my_shout_close_device(void *data)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+
+	close_shout_conn(sd);
+
+	if (sd->timer) {
+		timer_free(sd->timer);
+		sd->timer = NULL;
+	}
+}
+
+static int shout_connect(struct shout_data *sd)
+{
+	time_t t = time(NULL);
+	int state = shout_get_connected(sd->shout_conn);
+
+	/* already connected */
+	if (state == SHOUTERR_CONNECTED)
+		return 0;
+
+	/* waiting to connect */
+	if (state == SHOUTERR_BUSY && sd->conn_attempts != 0) {
+		/* timeout waiting to connect */
+		if ((t - sd->last_attempt) > sd->timeout) {
+			ERROR("timeout connecting to shout server %s:%i "
+			      "(attempt %i)\n",
+			      shout_get_host(sd->shout_conn),
+			      shout_get_port(sd->shout_conn),
+			      sd->conn_attempts);
+			return -1;
+		}
+
+		return 1;
+	}
+
+	/* we're in some funky state, so just reset it to unconnected */
+	if (state != SHOUTERR_UNCONNECTED)
+		shout_close(sd->shout_conn);
+
+	/* throttle new connection attempts */
+	if (sd->conn_attempts != 0 &&
+	    (t - sd->last_attempt) <= CONN_ATTEMPT_INTERVAL) {
+		return -1;
+	}
+
+	/* initiate a new connection */
+
+	sd->conn_attempts++;
+	sd->last_attempt = t;
+
+	state = shout_open(sd->shout_conn);
+	switch (state) {
+	case SHOUTERR_SUCCESS:
+	case SHOUTERR_CONNECTED:
+		return 0;
+	case SHOUTERR_BUSY:
+		return 1;
+	default:
+		ERROR("problem opening connection to shout server %s:%i "
+		      "(attempt %i): %s\n",
+		      shout_get_host(sd->shout_conn),
+		      shout_get_port(sd->shout_conn),
+		      sd->conn_attempts, shout_get_error(sd->shout_conn));
+		return -1;
+	}
+}
+
+static int open_shout_conn(void *data)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+	int status;
+
+	status = shout_connect(sd);
+	if (status != 0)
+		return status;
+
+	if (sd->encoder->init_encoder_func(sd) < 0) {
+		shout_close(sd->shout_conn);
+		return -1;
+	}
+
+	write_page(sd);
+
+	sd->shout_error = 0;
+	sd->opened = 1;
+	sd->tag_to_send = 1;
+	sd->conn_attempts = 0;
+
+	return 0;
+}
+
+static int my_shout_open_device(void *data,
+				struct audio_format *audio_format)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+
+	if (!sd->opened && open_shout_conn(sd) < 0)
+		return -1;
+
+	if (sd->timer)
+		timer_free(sd->timer);
+
+	sd->timer = timer_new(audio_format);
+
+	return 0;
+}
+
+static void send_metadata(struct shout_data * sd)
+{
+	static const int size = 1024;
+	char song[size];
+
+	if (!sd->opened || !sd->tag)
+		return;
+
+	if (sd->encoder->send_metadata_func(sd, song, size)) {
+		shout_metadata_add(sd->shout_meta, "song", song);
+		if (SHOUTERR_SUCCESS != shout_set_metadata(sd->shout_conn,
+							   sd->shout_meta)) {
+			ERROR("error setting shout metadata\n");
+			return;
+		}
+	}
+
+	sd->tag_to_send = 0;
+}
+
+static int my_shout_play(void *data,
+			 const char *chunk, size_t size)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+	int status;
+
+	if (!sd->timer->started)
+		timer_start(sd->timer);
+
+	timer_add(sd->timer, size);
+
+	if (sd->opened && sd->tag_to_send)
+		send_metadata(sd);
+
+	if (!sd->opened) {
+		status = open_shout_conn(sd);
+		if (status < 0) {
+			my_shout_close_device(sd);
+			return -1;
+		} else if (status > 0) {
+			timer_sync(sd->timer);
+			return 0;
+		}
+	}
+
+	if (sd->encoder->encode_func(sd, chunk, size)) {
+		my_shout_close_device(sd);
+		return -1;
+	}
+
+	if (write_page(sd) < 0) {
+		my_shout_close_device(sd);
+		return -1;
+	}
+
+	return 0;
+}
+
+static void my_shout_pause(void *data)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+	static const char silence[1020];
+	int ret;
+
+	/* play silence until the player thread sends us a command */
+
+	while (sd->opened && !audio_output_is_pending(sd->audio_output)) {
+		ret = my_shout_play(data, silence, sizeof(silence));
+		if (ret != 0)
+			break;
+	}
+}
+
+static void my_shout_set_tag(void *data,
+			     const struct tag *tag)
+{
+	struct shout_data *sd = (struct shout_data *)data;
+
+	if (sd->tag)
+		tag_free(sd->tag);
+	sd->tag = NULL;
+	sd->tag_to_send = 0;
+
+	if (!tag)
+		return;
+
+	sd->tag = tag_dup(tag);
+	sd->tag_to_send = 1;
+}
+
+const struct audio_output_plugin shoutPlugin = {
+	.name = "shout",
+	.init = my_shout_init_driver,
+	.finish = my_shout_finish_driver,
+	.open = my_shout_open_device,
+	.play = my_shout_play,
+	.pause = my_shout_pause,
+	.cancel = my_shout_drop_buffered_audio,
+	.close = my_shout_close_device,
+	.send_tag = my_shout_set_tag,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
+#endif
diff --git a/src/output/shout_plugin.h b/src/output/shout_plugin.h
new file mode 100644
index 00000000..2cfe68f2
--- /dev/null
+++ b/src/output/shout_plugin.h
@@ -0,0 +1,93 @@
+/* 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
+ */
+
+#ifndef AUDIO_OUTPUT_SHOUT_H
+#define AUDIO_OUTPUT_SHOUT_H
+
+#include "../output_api.h"
+
+#ifdef HAVE_SHOUT
+
+#include "../conf.h"
+#include "../timer.h"
+
+#include <shout/shout.h>
+
+struct shout_data;
+
+struct shout_encoder_plugin {
+	const char *name;
+	unsigned int shout_format;
+
+	int (*clear_encoder_func)(struct shout_data *sd);
+	int (*encode_func)(struct shout_data *sd,
+			   const char *chunk, size_t len);
+	void (*finish_func)(struct shout_data *sd);
+	int (*init_func)(struct shout_data *sd);
+	int (*init_encoder_func) (struct shout_data *sd);
+	/* Called when there is a new MpdTag to encode into the
+	   stream.  If this function returns non-zero, then the
+	   resulting song will be passed to the shout server as
+	   metadata.  This allows the Ogg encoder to send metadata via
+	   Vorbis comments in the stream, while an MP3 encoder can use
+	   the Shout Server's metadata API. */
+	int (*send_metadata_func)(struct shout_data *sd,
+				  char *song, size_t size);
+};
+
+struct shout_buffer {
+	unsigned char data[8192];
+	size_t len;
+};
+
+struct shout_data {
+	struct audio_output *audio_output;
+
+	shout_t *shout_conn;
+	shout_metadata_t *shout_meta;
+	int shout_error;
+
+	const struct shout_encoder_plugin *encoder;
+	void *encoder_data;
+
+	float quality;
+	int bitrate;
+
+	int opened;
+
+	struct tag *tag;
+	int tag_to_send;
+
+	int timeout;
+	int conn_attempts;
+	time_t last_attempt;
+
+	Timer *timer;
+
+	/* the configured audio format */
+	struct audio_format audio_format;
+
+	struct shout_buffer buf;
+};
+
+extern const struct shout_encoder_plugin shout_mp3_encoder;
+extern const struct shout_encoder_plugin shout_ogg_encoder;
+
+#endif
+
+#endif