Making branch for 0.13.0 fixes.

git-svn-id: https://svn.musicpd.org/mpd/branches/branch-0.13.0-fixes@6330 09075e82-0dd4-0310-85a5-a0d7c8717e4f

8 files changed, 0 insertions, 3203 deletions

diff --git a/trunk/src/audioOutputs/audioOutput_alsa.c b/trunk/src/audioOutputs/audioOutput_alsa.c
deleted file mode 100644
index 3ade3df46..000000000
--- a/trunk/src/audioOutputs/audioOutput_alsa.c
+++ /dev/null
@@ -1,427 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#include <stdlib.h>
-
-#ifdef HAVE_ALSA
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-#define MPD_ALSA_BUFFER_TIME_US 500000
-/* the default period time of xmms is 50 ms, so let's use that as well.
- * a user can tweak this parameter via the "period_time" config parameter.
- */
-#define MPD_ALSA_PERIOD_TIME_US 50000
-#define MPD_ALSA_RETRY_NR 5
-
-#include "../conf.h"
-#include "../log.h"
-
-#include <string.h>
-
-#include <alsa/asoundlib.h>
-
-typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
-					snd_pcm_uframes_t size);
-
-typedef struct _AlsaData {
-	char *device;
-	snd_pcm_t *pcmHandle;
-	alsa_writei_t *writei;
-	unsigned int buffer_time;
-	unsigned int period_time;
-	int sampleSize;
-	int useMmap;
-	int canPause;
-	int canResume;
-} AlsaData;
-
-static AlsaData *newAlsaData(void)
-{
-	AlsaData *ret = xmalloc(sizeof(AlsaData));
-
-	ret->device = NULL;
-	ret->pcmHandle = NULL;
-	ret->writei = snd_pcm_writei;
-	ret->useMmap = 0;
-	ret->buffer_time = MPD_ALSA_BUFFER_TIME_US;
-	ret->period_time = MPD_ALSA_PERIOD_TIME_US;
-
-	return ret;
-}
-
-static void freeAlsaData(AlsaData * ad)
-{
-	if (ad->device)
-		free(ad->device);
-
-	free(ad);
-}
-
-static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param)
-{
-	AlsaData *ad = newAlsaData();
-
-	if (param) {
-		BlockParam *bp = getBlockParam(param, "device");
-		ad->device = bp ? xstrdup(bp->value) : xstrdup("default");
-
-		if ((bp = getBlockParam(param, "use_mmap")) &&
-		    !strcasecmp(bp->value, "yes"))
-			ad->useMmap = 1;
-		if ((bp = getBlockParam(param, "buffer_time")))
-			ad->buffer_time = atoi(bp->value);
-		if ((bp = getBlockParam(param, "period_time")))
-			ad->period_time = atoi(bp->value);
-	} else
-		ad->device = xstrdup("default");
-	audioOutput->data = ad;
-
-	return 0;
-}
-
-static void alsa_finishDriver(AudioOutput * audioOutput)
-{
-	AlsaData *ad = audioOutput->data;
-
-	freeAlsaData(ad);
-}
-
-static int alsa_testDefault(void)
-{
-	snd_pcm_t *handle;
-
-	int ret = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK,
-			       SND_PCM_NONBLOCK);
-	snd_config_update_free_global();
-
-	if (ret) {
-		WARNING("Error opening default alsa device: %s\n",
-			snd_strerror(-ret));
-		return -1;
-	} else
-		snd_pcm_close(handle);
-
-	return 0;
-}
-
-static int alsa_openDevice(AudioOutput * audioOutput)
-{
-	AlsaData *ad = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
-	snd_pcm_format_t bitformat;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_sw_params_t *swparams;
-	unsigned int sampleRate = audioFormat->sampleRate;
-	unsigned int channels = audioFormat->channels;
-	snd_pcm_uframes_t alsa_buffer_size;
-	snd_pcm_uframes_t alsa_period_size;
-	int err;
-	const char *cmd = NULL;
-	int retry = MPD_ALSA_RETRY_NR;
-	unsigned int period_time, period_time_ro;
-	unsigned int buffer_time;
-
-	switch (audioFormat->bits) {
-	case 8:
-		bitformat = SND_PCM_FORMAT_S8;
-		break;
-	case 16:
-		bitformat = SND_PCM_FORMAT_S16;
-		break;
-	case 24:
-		bitformat = SND_PCM_FORMAT_S24;
-		break;
-	case 32:
-		bitformat = SND_PCM_FORMAT_S32;
-		break;
-	default:
-		ERROR("ALSA device \"%s\" doesn't support %i bit audio\n",
-		      ad->device, audioFormat->bits);
-		return -1;
-	}
-
-	err = snd_pcm_open(&ad->pcmHandle, ad->device,
-			   SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-	snd_config_update_free_global();
-	if (err < 0) {
-		ad->pcmHandle = NULL;
-		goto error;
-	}
-
-	cmd = "snd_pcm_nonblock";
-	err = snd_pcm_nonblock(ad->pcmHandle, 0);
-	if (err < 0)
-		goto error;
-
-	period_time_ro = period_time = ad->period_time;
-configure_hw:
-	/* configure HW params */
-	snd_pcm_hw_params_alloca(&hwparams);
-
-	cmd = "snd_pcm_hw_params_any";
-	err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
-	if (err < 0)
-		goto error;
-
-	if (ad->useMmap) {
-		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
-						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
-		if (err < 0) {
-			ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
-			      " %s\n", ad->device, snd_strerror(-err));
-			ERROR("Falling back to direct write mode\n");
-			ad->useMmap = 0;
-		} else
-			ad->writei = snd_pcm_mmap_writei;
-	}
-
-	if (!ad->useMmap) {
-		cmd = "snd_pcm_hw_params_set_access";
-		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
-						   SND_PCM_ACCESS_RW_INTERLEAVED);
-		if (err < 0)
-			goto error;
-		ad->writei = snd_pcm_writei;
-	}
-
-	err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
-	if (err < 0) {
-		ERROR("ALSA device \"%s\" does not support %i bit audio: "
-		      "%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
-		goto fail;
-	}
-
-	err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
-						  &channels);
-	if (err < 0) {
-		ERROR("ALSA device \"%s\" does not support %i channels: "
-		      "%s\n", ad->device, (int)audioFormat->channels,
-		      snd_strerror(-err));
-		goto fail;
-	}
-	audioFormat->channels = channels;
-
-	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
-					      &sampleRate, NULL);
-	if (err < 0 || sampleRate == 0) {
-		ERROR("ALSA device \"%s\" does not support %i Hz audio\n",
-		      ad->device, (int)audioFormat->sampleRate);
-		goto fail;
-	}
-	audioFormat->sampleRate = sampleRate;
-
-	buffer_time = ad->buffer_time;
-	cmd = "snd_pcm_hw_params_set_buffer_time_near";
-	err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
-						     &buffer_time, NULL);
-	if (err < 0)
-		goto error;
-
-	period_time = period_time_ro;
-	cmd = "snd_pcm_hw_params_set_period_time_near";
-	err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
-						     &period_time, NULL);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_hw_params";
-	err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
-	if (err == -EPIPE && --retry > 0) {
-		period_time_ro = period_time_ro >> 1;
-		goto configure_hw;
-	} else if (err < 0)
-		goto error;
-	if (retry != MPD_ALSA_RETRY_NR)
-		DEBUG("ALSA period_time set to %d\n", period_time);
-
-	cmd = "snd_pcm_hw_params_get_buffer_size";
-	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_hw_params_get_period_size";
-	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
-						NULL);
-	if (err < 0)
-		goto error;
-
-	ad->canPause = snd_pcm_hw_params_can_pause(hwparams);
-	ad->canResume = snd_pcm_hw_params_can_resume(hwparams);
-
-	/* configure SW params */
-	snd_pcm_sw_params_alloca(&swparams);
-
-	cmd = "snd_pcm_sw_params_current";
-	err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_start_threshold";
-	err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
-						    alsa_buffer_size -
-						    alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_avail_min";
-	err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
-					      alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_xfer_align";
-	err = snd_pcm_sw_params_set_xfer_align(ad->pcmHandle, swparams, 1);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params";
-	err = snd_pcm_sw_params(ad->pcmHandle, swparams);
-	if (err < 0)
-		goto error;
-
-	ad->sampleSize = (audioFormat->bits / 8) * audioFormat->channels;
-
-	audioOutput->open = 1;
-
-	DEBUG("alsa device \"%s\" will be playing %i bit, %i channel audio at "
-	      "%i Hz\n", ad->device, (int)audioFormat->bits,
-	      channels, sampleRate);
-
-	return 0;
-
-error:
-	if (cmd) {
-		ERROR("Error opening alsa device \"%s\" (%s): %s\n",
-		      ad->device, cmd, snd_strerror(-err));
-	} else {
-		ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
-		      snd_strerror(-err));
-	}
-fail:
-	if (ad->pcmHandle)
-		snd_pcm_close(ad->pcmHandle);
-	ad->pcmHandle = NULL;
-	audioOutput->open = 0;
-	return -1;
-}
-
-static int alsa_errorRecovery(AlsaData * ad, int err)
-{
-	if (err == -EPIPE) {
-		DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
-	} else if (err == -ESTRPIPE) {
-		DEBUG("alsa device \"%s\" was suspended\n", ad->device);
-	}
-
-	switch (snd_pcm_state(ad->pcmHandle)) {
-	case SND_PCM_STATE_PAUSED:
-		err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
-		break;
-	case SND_PCM_STATE_SUSPENDED:
-		err = ad->canResume ?
-		    snd_pcm_resume(ad->pcmHandle) :
-		    snd_pcm_prepare(ad->pcmHandle);
-		break;
-	case SND_PCM_STATE_SETUP:
-	case SND_PCM_STATE_XRUN:
-		err = snd_pcm_prepare(ad->pcmHandle);
-		break;
-	case SND_PCM_STATE_DISCONNECTED:
-		/* so alsa_closeDevice won't try to drain: */
-		snd_pcm_close(ad->pcmHandle);
-		ad->pcmHandle = NULL;
-		break;
-	default:
-		/* unknown state, do nothing */
-		break;
-	}
-
-	return err;
-}
-
-static void alsa_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	AlsaData *ad = audioOutput->data;
-
-	alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
-}
-
-static void alsa_closeDevice(AudioOutput * audioOutput)
-{
-	AlsaData *ad = audioOutput->data;
-
-	if (ad->pcmHandle) {
-		snd_pcm_drain(ad->pcmHandle);
-		snd_pcm_close(ad->pcmHandle);
-		ad->pcmHandle = NULL;
-	}
-
-	audioOutput->open = 0;
-}
-
-static int alsa_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
-{
-	AlsaData *ad = audioOutput->data;
-	int ret;
-
-	size /= ad->sampleSize;
-
-	while (size > 0) {
-		ret = ad->writei(ad->pcmHandle, playChunk, size);
-
-		if (ret == -EAGAIN || ret == -EINTR)
-			continue;
-
-		if (ret < 0) {
-			if (alsa_errorRecovery(ad, ret) < 0) {
-				ERROR("closing alsa device \"%s\" due to write "
-				      "error: %s\n", ad->device,
-				      snd_strerror(-errno));
-				alsa_closeDevice(audioOutput);
-				return -1;
-			}
-			continue;
-		}
-
-		playChunk += ret * ad->sampleSize;
-		size -= ret;
-	}
-
-	return 0;
-}
-
-AudioOutputPlugin alsaPlugin = {
-	"alsa",
-	alsa_testDefault,
-	alsa_initDriver,
-	alsa_finishDriver,
-	alsa_openDevice,
-	alsa_playAudio,
-	alsa_dropBufferedAudio,
-	alsa_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else /* HAVE ALSA */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
-#endif /* HAVE_ALSA */
diff --git a/trunk/src/audioOutputs/audioOutput_ao.c b/trunk/src/audioOutputs/audioOutput_ao.c
deleted file mode 100644
index a7f437ef4..000000000
--- a/trunk/src/audioOutputs/audioOutput_ao.c
+++ /dev/null
@@ -1,246 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#ifdef HAVE_AO
-
-#include "../conf.h"
-#include "../log.h"
-
-#include <string.h>
-
-#include <ao/ao.h>
-
-static int driverInitCount;
-
-typedef struct _AoData {
-	int writeSize;
-	int driverId;
-	ao_option *options;
-	ao_device *device;
-} AoData;
-
-static AoData *newAoData(void)
-{
-	AoData *ret = xmalloc(sizeof(AoData));
-	ret->device = NULL;
-	ret->options = NULL;
-
-	return ret;
-}
-
-static void audioOutputAo_error(void)
-{
-	if (errno == AO_ENOTLIVE) {
-		ERROR("not a live ao device\n");
-	} else if (errno == AO_EOPENDEVICE) {
-		ERROR("not able to open audio device\n");
-	} else if (errno == AO_EBADOPTION) {
-		ERROR("bad driver option\n");
-	}
-}
-
-static int audioOutputAo_initDriver(AudioOutput * audioOutput,
-				    ConfigParam * param)
-{
-	ao_info *ai;
-	char *dup;
-	char *stk1;
-	char *stk2;
-	char *n1;
-	char *key;
-	char *value;
-	char *test;
-	AoData *ad = newAoData();
-	BlockParam *blockParam;
-
-	audioOutput->data = ad;
-
-	if ((blockParam = getBlockParam(param, "write_size"))) {
-		ad->writeSize = strtol(blockParam->value, &test, 10);
-		if (*test != '\0') {
-			FATAL("\"%s\" is not a valid write size at line %i\n",
-			      blockParam->value, blockParam->line);
-		}
-	} else
-		ad->writeSize = 1024;
-
-	if (driverInitCount == 0) {
-		ao_initialize();
-	}
-	driverInitCount++;
-
-	blockParam = getBlockParam(param, "driver");
-
-	if (!blockParam || 0 == strcmp(blockParam->value, "default")) {
-		ad->driverId = ao_default_driver_id();
-	} else if ((ad->driverId = ao_driver_id(blockParam->value)) < 0) {
-		FATAL("\"%s\" is not a valid ao driver at line %i\n",
-		      blockParam->value, blockParam->line);
-	}
-
-	if ((ai = ao_driver_info(ad->driverId)) == NULL) {
-		FATAL("problems getting driver info for device defined at line %i\n"
-		      "you may not have permission to the audio device\n", param->line);
-	}
-
-	DEBUG("using ao driver \"%s\" for \"%s\"\n", ai->short_name,
-	      audioOutput->name);
-
-	blockParam = getBlockParam(param, "options");
-
-	if (blockParam) {
-		dup = xstrdup(blockParam->value);
-	} else
-		dup = xstrdup("");
-
-	if (strlen(dup)) {
-		stk1 = NULL;
-		n1 = strtok_r(dup, ";", &stk1);
-		while (n1) {
-			stk2 = NULL;
-			key = strtok_r(n1, "=", &stk2);
-			if (!key)
-				FATAL("problems parsing options \"%s\"\n", n1);
-			/*found = 0;
-			   for(i=0;i<ai->option_count;i++) {
-			   if(strcmp(ai->options[i],key)==0) {
-			   found = 1;
-			   break;
-			   }
-			   }
-			   if(!found) {
-			   FATAL("\"%s\" is not an option for "
-			   "\"%s\" ao driver\n",key,
-			   ai->short_name);
-			   } */
-			value = strtok_r(NULL, "", &stk2);
-			if (!value)
-				FATAL("problems parsing options \"%s\"\n", n1);
-			ao_append_option(&ad->options, key, value);
-			n1 = strtok_r(NULL, ";", &stk1);
-		}
-	}
-	free(dup);
-
-	return 0;
-}
-
-static void freeAoData(AoData * ad)
-{
-	ao_free_options(ad->options);
-	free(ad);
-}
-
-static void audioOutputAo_finishDriver(AudioOutput * audioOutput)
-{
-	AoData *ad = (AoData *) audioOutput->data;
-	freeAoData(ad);
-
-	driverInitCount--;
-
-	if (driverInitCount == 0)
-		ao_shutdown();
-}
-
-static void audioOutputAo_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	/* not supported by libao */
-}
-
-static void audioOutputAo_closeDevice(AudioOutput * audioOutput)
-{
-	AoData *ad = (AoData *) audioOutput->data;
-
-	if (ad->device) {
-		ao_close(ad->device);
-		ad->device = NULL;
-	}
-
-	audioOutput->open = 0;
-}
-
-static int audioOutputAo_openDevice(AudioOutput * audioOutput)
-{
-	ao_sample_format format;
-	AoData *ad = (AoData *) audioOutput->data;
-
-	if (ad->device) {
-		audioOutputAo_closeDevice(audioOutput);
-	}
-
-	format.bits = audioOutput->outAudioFormat.bits;
-	format.rate = audioOutput->outAudioFormat.sampleRate;
-	format.byte_format = AO_FMT_NATIVE;
-	format.channels = audioOutput->outAudioFormat.channels;
-
-	ad->device = ao_open_live(ad->driverId, &format, ad->options);
-
-	if (ad->device == NULL)
-		return -1;
-
-	audioOutput->open = 1;
-
-	return 0;
-}
-
-static int audioOutputAo_play(AudioOutput * audioOutput, char *playChunk,
-			      int size)
-{
-	int send;
-	AoData *ad = (AoData *) audioOutput->data;
-
-	if (ad->device == NULL)
-		return -1;
-
-	while (size > 0) {
-		send = ad->writeSize > size ? size : ad->writeSize;
-
-		if (ao_play(ad->device, playChunk, send) == 0) {
-			audioOutputAo_error();
-			ERROR("closing audio device due to write error\n");
-			audioOutputAo_closeDevice(audioOutput);
-			return -1;
-		}
-
-		playChunk += send;
-		size -= send;
-	}
-
-	return 0;
-}
-
-AudioOutputPlugin aoPlugin = {
-	"ao",
-	NULL,
-	audioOutputAo_initDriver,
-	audioOutputAo_finishDriver,
-	audioOutputAo_openDevice,
-	audioOutputAo_play,
-	audioOutputAo_dropBufferedAudio,
-	audioOutputAo_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else
-
-#include <stdio.h>
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
-#endif
diff --git a/trunk/src/audioOutputs/audioOutput_jack.c b/trunk/src/audioOutputs/audioOutput_jack.c
deleted file mode 100644
index 1fdfaf4bb..000000000
--- a/trunk/src/audioOutputs/audioOutput_jack.c
+++ /dev/null
@@ -1,440 +0,0 @@
-/* jack plug in for the Music Player Daemon (MPD)
- * (c)2006 by anarch(anarchsss@gmail.com)
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#ifdef HAVE_JACK
-
-#include <stdlib.h>
-#include <errno.h>
-
-#include "../conf.h"
-#include "../log.h"
-
-#include <string.h>
-#include <pthread.h>
-
-#include <jack/jack.h>
-#include <jack/types.h>
-#include <jack/ringbuffer.h>
-
-pthread_mutex_t play_audio_lock = PTHREAD_MUTEX_INITIALIZER;
-pthread_cond_t  play_audio = PTHREAD_COND_INITIALIZER;
-
-/*#include "dmalloc.h"*/
-
-#define MIN(a, b) ((a) < (b) ? (a) : (b))
-/*#define SAMPLE_SIZE  sizeof(jack_default_audio_sample_t);*/
-
-
-static char *name = "mpd";
-static char *output_ports[2];
-static int ringbuf_sz = 32768;
-size_t sample_size = sizeof(jack_default_audio_sample_t);
-
-typedef struct _JackData {
-	jack_port_t *ports[2];
-	jack_client_t *client;
-	jack_ringbuffer_t *ringbuffer[2];
-	int bps;
-	int shutdown;
-} JackData;
-
-/*JackData *jd = NULL;*/
-
-static JackData *newJackData(void)
-{
-	JackData *ret;
-	ret = xcalloc(sizeof(JackData), 1);
-
-	return ret;
-}
-
-static void freeJackData(AudioOutput *audioOutput)
-{
-	JackData *jd = audioOutput->data;
-	if (jd) {
-		if (jd->ringbuffer[0])
-			jack_ringbuffer_free(jd->ringbuffer[0]);
-		if (jd->ringbuffer[1])
-			jack_ringbuffer_free(jd->ringbuffer[1]);
-		free(jd);
-		audioOutput->data = NULL;
-	}
-}
-
-static void jack_finishDriver(AudioOutput *audioOutput)
-{
-	JackData *jd = audioOutput->data;
-	int i;
-
-	if ( jd && jd->client ) {
-		jack_deactivate(jd->client);
-		jack_client_close(jd->client);
-	}
-	DEBUG("disconnect_jack (pid=%d)\n", getpid ());
-
- 	if ( strcmp(name, "mpd") ) {
- 		free(name);
- 		name = "mpd";
- 	}
-
-	for ( i = ARRAY_SIZE(output_ports); --i >= 0; ) {
-		if (!output_ports[i])
- 			continue;
- 		free(output_ports[i]);
- 		output_ports[i] = NULL;
- 	}
-
-	freeJackData(audioOutput);
-}
-
-static int srate(jack_nframes_t rate, void *data)
-{
-	JackData *jd = (JackData *) ((AudioOutput*) data)->data;
- 	AudioFormat *audioFormat = &(((AudioOutput*) data)->outAudioFormat);
-
- 	audioFormat->sampleRate = (int)jack_get_sample_rate(jd->client);
-
-	return 0;
-}
-
-static int process(jack_nframes_t nframes, void *arg)
-{
-	size_t i;
-	JackData *jd = (JackData *) arg;
-	jack_default_audio_sample_t *out[2];
-	size_t avail_data, avail_frames;
-
-	if ( nframes <= 0 )
-		return 0;
-
-	out[0] = jack_port_get_buffer(jd->ports[0], nframes);
-	out[1] = jack_port_get_buffer(jd->ports[1], nframes);
-
-	while ( nframes ) {
-		avail_data = jack_ringbuffer_read_space(jd->ringbuffer[1]);
-
-		if ( avail_data > 0 ) {
-		    avail_frames = avail_data / sample_size;
-
-		    if (avail_frames > nframes) {
-			avail_frames = nframes;
-			avail_data = nframes*sample_size;
-		    }
-
-		    jack_ringbuffer_read(jd->ringbuffer[0], (char *)out[0],
-					 avail_data);
-		    jack_ringbuffer_read(jd->ringbuffer[1], (char *)out[1],
-					 avail_data);
-
-		    nframes -= avail_frames;
-		    out[0] += avail_data;
-		    out[1] += avail_data;
-		} else {
-		    for (i = 0; i < nframes; i++)
-  			out[0][i] = out[1][i] = 0.0;
-		    nframes = 0;
-		}
-
-		if (pthread_mutex_trylock (&play_audio_lock) == 0) {
-			pthread_cond_signal (&play_audio);
-			pthread_mutex_unlock (&play_audio_lock);
-		}
-	}
-
-
-	/*DEBUG("process (pid=%d)\n", getpid());*/
-	return 0;
-}
-
-static void shutdown_callback(void *arg)
-{
-	JackData *jd = (JackData *) arg;
-	jd->shutdown = 1;
-}
-
-static void set_audioformat(AudioOutput *audioOutput)
-{
-	JackData *jd = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
-
-	audioFormat->sampleRate = (int) jack_get_sample_rate(jd->client);
-	DEBUG("samplerate = %d\n", audioFormat->sampleRate);
-	audioFormat->channels = 2;
-	audioFormat->bits = 16;
-	jd->bps = audioFormat->channels
-		* sizeof(jack_default_audio_sample_t)
-		* audioFormat->sampleRate;
-}
-
-static void error_callback(const char *msg)
-{
-	ERROR("jack: %s\n", msg);
-}
-
-static int jack_initDriver(AudioOutput *audioOutput, ConfigParam *param)
-{
-	BlockParam *bp;
-	char *endptr;
-	int val;
- 	char *cp = NULL;
-
-	DEBUG("jack_initDriver (pid=%d)\n", getpid());
-	if ( ! param ) return 0;
-
-	if ( (bp = getBlockParam(param, "ports")) ) {
-		DEBUG("output_ports=%s\n", bp->value);
-
-		if (!(cp = strchr(bp->value, ',')))
-			FATAL("expected comma and a second value for '%s' "
-			      "at line %d: %s\n",
-			      bp->name, bp->line, bp->value);
-
-		*cp = '\0';
-		output_ports[0] = xstrdup(bp->value);
-		*cp++ = ',';
-
-		if (!*cp)
-			FATAL("expected a second value for '%s' at line %d: "
-			      "%s\n", bp->name, bp->line, bp->value);
-
-		output_ports[1] = xstrdup(cp);
-
-		if (strchr(cp,','))
-			FATAL("Only %d values are supported for '%s' "
-			      "at line %d\n", (int)ARRAY_SIZE(output_ports),
-			      bp->name, bp->line);
-	}
-
-	if ( (bp = getBlockParam(param, "ringbuffer_size")) ) {
-		errno = 0;
-		val = strtol(bp->value, &endptr, 10);
-
-		if ( errno == 0 && endptr != bp->value) {
-			ringbuf_sz = val < 32768 ? 32768 : val;
-			DEBUG("ringbuffer_size=%d\n", ringbuf_sz);
-		} else {
-			FATAL("%s is not a number; ringbuf_size=%d\n",
-			      bp->value, ringbuf_sz);
-		}
-	}
-
-	if ( (bp = getBlockParam(param, "name"))
-	     && (strcmp(bp->value, "mpd") != 0) ) {
-		name = xstrdup(bp->value);
-		DEBUG("name=%s\n", name);
-	}
-
- 	return 0;
-}
-
-static int jack_testDefault(void)
-{
-	return 0;
-}
-
-static int connect_jack(AudioOutput *audioOutput)
-{
-	JackData *jd = audioOutput->data;
-	char **jports;
-	char *port_name;
-
-	if ( (jd->client = jack_client_new(name)) == NULL ) {
-		ERROR("jack server not running?\n");
-		freeJackData(audioOutput);
-		return -1;
-	}
-
-	jack_set_error_function(error_callback);
-	jack_set_process_callback(jd->client, process, (void *)jd);
-	jack_set_sample_rate_callback(jd->client, (JackProcessCallback)srate,
-				      (void *)audioOutput);
-	jack_on_shutdown(jd->client, shutdown_callback, (void *)jd);
-
-	if ( jack_activate(jd->client) ) {
-		ERROR("cannot activate client");
-		freeJackData(audioOutput);
-		return -1;
-	}
-
-	jd->ports[0] = jack_port_register(jd->client, "left",
-					  JACK_DEFAULT_AUDIO_TYPE,
-					  JackPortIsOutput, 0);
-	if ( !jd->ports[0] ) {
-		ERROR("Cannot register left output port.\n");
-		freeJackData(audioOutput);
-		return -1;
-	}
-
-	jd->ports[1] = jack_port_register(jd->client, "right",
-					  JACK_DEFAULT_AUDIO_TYPE,
-					  JackPortIsOutput, 0);
-	if ( !jd->ports[1] ) {
-		ERROR("Cannot register right output port.\n");
-		freeJackData(audioOutput);
-		return -1;
-	}
-
-	/*  hay que buscar que hay  */
-	if ( !output_ports[1]
-	     && (jports = (char **)jack_get_ports(jd->client, NULL, NULL,
-							JackPortIsPhysical|
-							JackPortIsInput)) ) {
-		output_ports[0] = jports[0];
-		output_ports[1] = jports[1] ? jports[1] : jports[0];
-		DEBUG("output_ports: %s %s\n", output_ports[0], output_ports[1]);
-		free(jports);
-	}
-
-	if ( output_ports[1] ) {
-		jd->ringbuffer[0] = jack_ringbuffer_create(ringbuf_sz);
-		jd->ringbuffer[1] = jack_ringbuffer_create(ringbuf_sz);
-		memset(jd->ringbuffer[0]->buf, 0, jd->ringbuffer[0]->size);
-		memset(jd->ringbuffer[1]->buf, 0, jd->ringbuffer[1]->size);
-
-		port_name = xmalloc(sizeof(char)*(7+strlen(name)));
-
-		sprintf(port_name, "%s:left", name);
-		if ( (jack_connect(jd->client, port_name,
-				   output_ports[0])) != 0 ) {
-			ERROR("%s is not a valid Jack Client / Port ",
-			      output_ports[0]);
-			freeJackData(audioOutput);
-			free(port_name);
-			return -1;
-		}
-		sprintf(port_name, "%s:right", name);
-		if ( (jack_connect(jd->client, port_name,
-				   output_ports[1])) != 0 ) {
-			ERROR("%s is not a valid Jack Client / Port ",
-			      output_ports[1]);
-			freeJackData(audioOutput);
-			free(port_name);
-			return -1;
-		}
-		free(port_name);
-	}
-
-	DEBUG("connect_jack (pid=%d)\n", getpid());
-	return 1;
-}
-
-static int jack_openDevice(AudioOutput *audioOutput)
-{
-	JackData *jd = audioOutput->data;
-
-	if ( !jd ) {
-		DEBUG("connect!\n");
-		jd = newJackData();
-		audioOutput->data = jd;
-
-		if (connect_jack(audioOutput) < 0) {
-			freeJackData(audioOutput);
-			audioOutput->open = 0;
-			return -1;
-		}
-	}
-
-	set_audioformat(audioOutput);
-	audioOutput->open = 1;
-
-	DEBUG("jack_openDevice (pid=%d)!\n", getpid ());
-	return 0;
-}
-
-
-static void jack_closeDevice(AudioOutput * audioOutput)
-{
-	/*jack_finishDriver(audioOutput);*/
-	audioOutput->open = 0;
-	DEBUG("jack_closeDevice (pid=%d)\n", getpid());
-}
-
-static void jack_dropBufferedAudio (AudioOutput * audioOutput)
-{
-}
-
-static int jack_playAudio(AudioOutput * audioOutput, char *buff, int size)
-{
-	JackData *jd = audioOutput->data;
-	size_t space;
-	int i;
-	short *buffer = (short *) buff;
-	jack_default_audio_sample_t sample;
-	size_t samples = size/4;
-
-	/*DEBUG("jack_playAudio: (pid=%d)!\n", getpid());*/
-
-	if ( jd->shutdown ) {
-		ERROR("Refusing to play, because there is no client thread.\n");
-		freeJackData(audioOutput);
-		audioOutput->open = 0;
-		return 0;
-	}
-
-	while ( samples && !jd->shutdown ) {
-
- 		if ( (space = jack_ringbuffer_write_space(jd->ringbuffer[0]))
- 		     >= samples*sample_size ) {
-
-			/*space = MIN(space, samples*sample_size);*/
-			/*space = samples*sample_size;*/
-
-			/*for(i=0; i<space/sample_size; i++) {*/
-			for(i=0; i<samples; i++) {
-				sample = (jack_default_audio_sample_t) *(buffer++)/32768.0;
-
-				jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
-						      sample_size);
-
-				sample = (jack_default_audio_sample_t) *(buffer++)/32768.0;
-
-				jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
-						      sample_size);
-
-				/*samples--;*/
-			}
-			samples=0;
-
- 		} else {
-			pthread_mutex_lock(&play_audio_lock);
-			pthread_cond_wait(&play_audio, &play_audio_lock);
-			pthread_mutex_unlock(&play_audio_lock);
-		}
-
-	}
-	return 0;
-}
-
-AudioOutputPlugin jackPlugin = {
-	"jack",
-	jack_testDefault,
-	jack_initDriver,
-	jack_finishDriver,
-	jack_openDevice,
-	jack_playAudio,
-	jack_dropBufferedAudio,
-	jack_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else /* HAVE JACK */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
-
-#endif /* HAVE_JACK */
diff --git a/trunk/src/audioOutputs/audioOutput_mvp.c b/trunk/src/audioOutputs/audioOutput_mvp.c
deleted file mode 100644
index ea365c657..000000000
--- a/trunk/src/audioOutputs/audioOutput_mvp.c
+++ /dev/null
@@ -1,284 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * Media MVP audio output based on code from MVPMC project:
- * http://mvpmc.sourceforge.net/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#include <stdlib.h>
-
-#ifdef HAVE_MVP
-
-#include "../conf.h"
-#include "../log.h"
-
-#include <string.h>
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <unistd.h>
-#include <errno.h>
-
-typedef struct {
-	unsigned long dsp_status;
-	unsigned long stream_decode_type;
-	unsigned long sample_rate;
-	unsigned long bit_rate;
-	unsigned long raw[64 / sizeof(unsigned long)];
-} aud_status_t;
-
-#define MVP_SET_AUD_STOP		_IOW('a',1,int)
-#define MVP_SET_AUD_PLAY		_IOW('a',2,int)
-#define MVP_SET_AUD_PAUSE	_IOW('a',3,int)
-#define MVP_SET_AUD_UNPAUSE	_IOW('a',4,int)
-#define MVP_SET_AUD_SRC		_IOW('a',5,int)
-#define MVP_SET_AUD_MUTE		_IOW('a',6,int)
-#define MVP_SET_AUD_BYPASS	_IOW('a',8,int)
-#define MVP_SET_AUD_CHANNEL	_IOW('a',9,int)
-#define MVP_GET_AUD_STATUS	_IOR('a',10,aud_status_t)
-#define MVP_SET_AUD_VOLUME	_IOW('a',13,int)
-#define MVP_GET_AUD_VOLUME	_IOR('a',14,int)
-#define MVP_SET_AUD_STREAMTYPE	_IOW('a',15,int)
-#define MVP_SET_AUD_FORMAT	_IOW('a',16,int)
-#define MVP_GET_AUD_SYNC		_IOR('a',21,pts_sync_data_t*)
-#define MVP_SET_AUD_STC		_IOW('a',22,long long int *)
-#define MVP_SET_AUD_SYNC		_IOW('a',23,int)
-#define MVP_SET_AUD_END_STREAM	_IOW('a',25,int)
-#define MVP_SET_AUD_RESET	_IOW('a',26,int)
-#define MVP_SET_AUD_DAC_CLK	_IOW('a',27,int)
-#define MVP_GET_AUD_REGS		_IOW('a',28,aud_ctl_regs_t*)
-
-typedef struct _MvpData {
-	int fd;
-} MvpData;
-
-static int pcmfrequencies[][3] = {
-	{9, 8000, 32000},
-	{10, 11025, 44100},
-	{11, 12000, 48000},
-	{1, 16000, 32000},
-	{2, 22050, 44100},
-	{3, 24000, 48000},
-	{5, 32000, 32000},
-	{0, 44100, 44100},
-	{7, 48000, 48000},
-	{13, 64000, 32000},
-	{14, 88200, 44100},
-	{15, 96000, 48000}
-};
-
-static int numfrequencies = sizeof(pcmfrequencies) / 12;
-
-static int mvp_testDefault(void)
-{
-	int fd;
-
-	fd = open("/dev/adec_pcm", O_WRONLY);
-
-	if (fd) {
-		close(fd);
-		return 0;
-	}
-
-	WARNING("Error opening PCM device \"/dev/adec_pcm\": %s\n",
-		strerror(errno));
-
-	return -1;
-}
-
-static int mvp_initDriver(AudioOutput * audioOutput, ConfigParam * param)
-{
-	MvpData *md = xmalloc(sizeof(MvpData));
-	md->fd = -1;
-	audioOutput->data = md;
-
-	return 0;
-}
-
-static void mvp_finishDriver(AudioOutput * audioOutput)
-{
-	MvpData *md = audioOutput->data;
-	free(md);
-}
-
-static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
-			    int big_endian, int bits)
-{
-	int iloop;
-	int mix[5];
-
-	if (channels == 1)
-		mix[0] = 1;
-	else if (channels == 2)
-		mix[0] = 0;
-	else
-		return -1;
-
-	/* 0,1=24bit(24) , 2,3=16bit */
-	if (bits == 16)
-		mix[1] = 2;
-	else if (bits == 24)
-		mix[1] = 0;
-	else
-		return -1;
-
-	mix[3] = 0;	/* stream type? */
-
-	if (big_endian == 1)
-		mix[4] = 1;
-	else if (big_endian == 0)
-		mix[4] = 0;
-	else
-		return -1;
-
-	/*
-	 * if there is an exact match for the frequency, use it.
-	 */
-	for (iloop = 0; iloop < numfrequencies; iloop++) {
-		if (rate == pcmfrequencies[iloop][1]) {
-			mix[2] = pcmfrequencies[iloop][0];
-			break;
-		}
-	}
-
-	if (iloop >= numfrequencies) {
-		ERROR("Can not find suitable output frequency for %ld\n", rate);
-		return -1;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
-		ERROR("Can not set audio format\n");
-		return -1;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
-		ERROR("Can not set audio sync\n");
-		return -1;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
-		ERROR("Can not set audio play mode\n");
-		return -1;
-	}
-
-	return 0;
-}
-
-static int mvp_openDevice(AudioOutput * audioOutput)
-{
-	long long int stc = 0;
-	MvpData *md = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
-	int mix[5] = { 0, 2, 7, 1, 0 };
-
-	if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
-		ERROR("Error opening /dev/adec_pcm: %s\n", strerror(errno));
-		return -1;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
-		ERROR("Error setting audio source: %s\n", strerror(errno));
-		return -1;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
-		ERROR("Error setting audio streamtype: %s\n", strerror(errno));
-		return -1;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
-		ERROR("Error setting audio format: %s\n", strerror(errno));
-		return -1;
-	}
-	ioctl(md->fd, MVP_SET_AUD_STC, &stc);
-	if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
-		ERROR("Error setting audio streamtype: %s\n", strerror(errno));
-		return -1;
-	}
-#ifdef WORDS_BIGENDIAN
-	mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 0,
-			 audioFormat->bits);
-#else
-	mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 1,
-			 audioFormat->bits);
-#endif
-	audioOutput->open = 1;
-	return 0;
-}
-
-static void mvp_closeDevice(AudioOutput * audioOutput)
-{
-	MvpData *md = audioOutput->data;
-	if (md->fd >= 0)
-		close(md->fd);
-	md->fd = -1;
-	audioOutput->open = 0;
-}
-
-static void mvp_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	MvpData *md = audioOutput->data;
-	if (md->fd >= 0) {
-		ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
-		close(md->fd);
-		md->fd = -1;
-		audioOutput->open = 0;
-	}
-}
-
-static int mvp_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
-{
-	MvpData *md = audioOutput->data;
-	int ret;
-
-	/* reopen the device since it was closed by dropBufferedAudio */
-	if (md->fd < 0)
-		mvp_openDevice(audioOutput);
-
-	while (size > 0) {
-		ret = write(md->fd, playChunk, size);
-		if (ret < 0) {
-			if (errno == EINTR)
-				continue;
-			ERROR("closing mvp PCM device due to write error: "
-			      "%s\n", strerror(errno));
-			mvp_closeDevice(audioOutput);
-			return -1;
-		}
-		playChunk += ret;
-		size -= ret;
-	}
-	return 0;
-}
-
-AudioOutputPlugin mvpPlugin = {
-	"mvp",
-	mvp_testDefault,
-	mvp_initDriver,
-	mvp_finishDriver,
-	mvp_openDevice,
-	mvp_playAudio,
-	mvp_dropBufferedAudio,
-	mvp_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else /* HAVE_MVP */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
-#endif /* HAVE_MVP */
diff --git a/trunk/src/audioOutputs/audioOutput_oss.c b/trunk/src/audioOutputs/audioOutput_oss.c
deleted file mode 100644
index 01293cbd1..000000000
--- a/trunk/src/audioOutputs/audioOutput_oss.c
+++ /dev/null
@@ -1,575 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * OSS audio output (c) 2004, 2005, 2006, 2007 by Eric Wong <eric@petta-tech.com>
- *                   and Warren Dukes <warren.dukes@gmail.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#include <stdlib.h>
-
-#ifdef HAVE_OSS
-
-#include "../conf.h"
-#include "../log.h"
-
-#include <string.h>
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <unistd.h>
-#include <errno.h>
-
-#if defined(__OpenBSD__) || defined(__NetBSD__)
-# include <soundcard.h>
-#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
-# include <sys/soundcard.h>
-#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
-
-#ifdef WORDS_BIGENDIAN
-# define	AFMT_S16_MPD	 AFMT_S16_BE
-#else
-# define	AFMT_S16_MPD	 AFMT_S16_LE
-#endif /* WORDS_BIGENDIAN */
-
-typedef struct _OssData {
-	int fd;
-	const char *device;
-	int channels;
-	int sampleRate;
-	int bitFormat;
-	int bits;
-	int *supported[3];
-	int numSupported[3];
-	int *unsupported[3];
-	int numUnsupported[3];
-} OssData;
-
-#define OSS_SUPPORTED		1
-#define OSS_UNSUPPORTED		0
-#define OSS_UNKNOWN		-1
-
-#define OSS_RATE		0
-#define OSS_CHANNELS		1
-#define OSS_BITS		2
-
-static int getIndexForParam(int param)
-{
-	int index = 0;
-
-	switch (param) {
-	case SNDCTL_DSP_SPEED:
-		index = OSS_RATE;
-		break;
-	case SNDCTL_DSP_CHANNELS:
-		index = OSS_CHANNELS;
-		break;
-	case SNDCTL_DSP_SAMPLESIZE:
-		index = OSS_BITS;
-		break;
-	}
-
-	return index;
-}
-
-static int findSupportedParam(OssData * od, int param, int val)
-{
-	int i;
-	int index = getIndexForParam(param);
-
-	for (i = 0; i < od->numSupported[index]; i++) {
-		if (od->supported[index][i] == val)
-			return 1;
-	}
-
-	return 0;
-}
-
-static int canConvert(int index, int val)
-{
-	switch (index) {
-	case OSS_BITS:
-		if (val != 16)
-			return 0;
-		break;
-	case OSS_CHANNELS:
-		if (val != 2)
-			return 0;
-		break;
-	}
-
-	return 1;
-}
-
-static int getSupportedParam(OssData * od, int param, int val)
-{
-	int i;
-	int index = getIndexForParam(param);
-	int ret = -1;
-	int least = val;
-	int diff;
-
-	for (i = 0; i < od->numSupported[index]; i++) {
-		diff = od->supported[index][i] - val;
-		if (diff < 0)
-			diff = -diff;
-		if (diff < least) {
-			if (!canConvert(index, od->supported[index][i])) {
-				continue;
-			}
-			least = diff;
-			ret = od->supported[index][i];
-		}
-	}
-
-	return ret;
-}
-
-static int findUnsupportedParam(OssData * od, int param, int val)
-{
-	int i;
-	int index = getIndexForParam(param);
-
-	for (i = 0; i < od->numUnsupported[index]; i++) {
-		if (od->unsupported[index][i] == val)
-			return 1;
-	}
-
-	return 0;
-}
-
-static void addSupportedParam(OssData * od, int param, int val)
-{
-	int index = getIndexForParam(param);
-
-	od->numSupported[index]++;
-	od->supported[index] = xrealloc(od->supported[index],
-				       od->numSupported[index] * sizeof(int));
-	od->supported[index][od->numSupported[index] - 1] = val;
-}
-
-static void addUnsupportedParam(OssData * od, int param, int val)
-{
-	int index = getIndexForParam(param);
-
-	od->numUnsupported[index]++;
-	od->unsupported[index] = xrealloc(od->unsupported[index],
-					 od->numUnsupported[index] *
-					 sizeof(int));
-	od->unsupported[index][od->numUnsupported[index] - 1] = val;
-}
-
-static void removeSupportedParam(OssData * od, int param, int val)
-{
-	int i = 0;
-	int j = 0;
-	int index = getIndexForParam(param);
-
-	for (i = 0; i < od->numSupported[index] - 1; i++) {
-		if (od->supported[index][i] == val)
-			j = 1;
-		od->supported[index][i] = od->supported[index][i + j];
-	}
-
-	od->numSupported[index]--;
-	od->supported[index] = xrealloc(od->supported[index],
-				       od->numSupported[index] * sizeof(int));
-}
-
-static void removeUnsupportedParam(OssData * od, int param, int val)
-{
-	int i = 0;
-	int j = 0;
-	int index = getIndexForParam(param);
-
-	for (i = 0; i < od->numUnsupported[index] - 1; i++) {
-		if (od->unsupported[index][i] == val)
-			j = 1;
-		od->unsupported[index][i] = od->unsupported[index][i + j];
-	}
-
-	od->numUnsupported[index]--;
-	od->unsupported[index] = xrealloc(od->unsupported[index],
-					 od->numUnsupported[index] *
-					 sizeof(int));
-}
-
-static int isSupportedParam(OssData * od, int param, int val)
-{
-	if (findSupportedParam(od, param, val))
-		return OSS_SUPPORTED;
-	if (findUnsupportedParam(od, param, val))
-		return OSS_UNSUPPORTED;
-	return OSS_UNKNOWN;
-}
-
-static void supportParam(OssData * od, int param, int val)
-{
-	int supported = isSupportedParam(od, param, val);
-
-	if (supported == OSS_SUPPORTED)
-		return;
-
-	if (supported == OSS_UNSUPPORTED) {
-		removeUnsupportedParam(od, param, val);
-	}
-
-	addSupportedParam(od, param, val);
-}
-
-static void unsupportParam(OssData * od, int param, int val)
-{
-	int supported = isSupportedParam(od, param, val);
-
-	if (supported == OSS_UNSUPPORTED)
-		return;
-
-	if (supported == OSS_SUPPORTED) {
-		removeSupportedParam(od, param, val);
-	}
-
-	addUnsupportedParam(od, param, val);
-}
-
-static OssData *newOssData(void)
-{
-	OssData *ret = xmalloc(sizeof(OssData));
-
-	ret->device = NULL;
-	ret->fd = -1;
-
-	ret->supported[OSS_RATE] = NULL;
-	ret->supported[OSS_CHANNELS] = NULL;
-	ret->supported[OSS_BITS] = NULL;
-	ret->unsupported[OSS_RATE] = NULL;
-	ret->unsupported[OSS_CHANNELS] = NULL;
-	ret->unsupported[OSS_BITS] = NULL;
-
-	ret->numSupported[OSS_RATE] = 0;
-	ret->numSupported[OSS_CHANNELS] = 0;
-	ret->numSupported[OSS_BITS] = 0;
-	ret->numUnsupported[OSS_RATE] = 0;
-	ret->numUnsupported[OSS_CHANNELS] = 0;
-	ret->numUnsupported[OSS_BITS] = 0;
-
-	supportParam(ret, SNDCTL_DSP_SPEED, 48000);
-	supportParam(ret, SNDCTL_DSP_SPEED, 44100);
-	supportParam(ret, SNDCTL_DSP_CHANNELS, 2);
-	supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16);
-
-	return ret;
-}
-
-static void freeOssData(OssData * od)
-{
-	if (od->supported[OSS_RATE])
-		free(od->supported[OSS_RATE]);
-	if (od->supported[OSS_CHANNELS])
-		free(od->supported[OSS_CHANNELS]);
-	if (od->supported[OSS_BITS])
-		free(od->supported[OSS_BITS]);
-	if (od->unsupported[OSS_RATE])
-		free(od->unsupported[OSS_RATE]);
-	if (od->unsupported[OSS_CHANNELS])
-		free(od->unsupported[OSS_CHANNELS]);
-	if (od->unsupported[OSS_BITS])
-		free(od->unsupported[OSS_BITS]);
-
-	free(od);
-}
-
-#define OSS_STAT_NO_ERROR 	0
-#define OSS_STAT_NOT_CHAR_DEV	-1
-#define OSS_STAT_NO_PERMS	-2
-#define OSS_STAT_DOESN_T_EXIST	-3
-#define OSS_STAT_OTHER		-4
-
-static int oss_statDevice(const char *device, int *stErrno)
-{
-	struct stat st;
-
-	if (0 == stat(device, &st)) {
-		if (!S_ISCHR(st.st_mode)) {
-			return OSS_STAT_NOT_CHAR_DEV;
-		}
-	} else {
-		*stErrno = errno;
-
-		switch (errno) {
-		case ENOENT:
-		case ENOTDIR:
-			return OSS_STAT_DOESN_T_EXIST;
-		case EACCES:
-			return OSS_STAT_NO_PERMS;
-		default:
-			return OSS_STAT_OTHER;
-		}
-	}
-
-	return 0;
-}
-
-static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" };
-
-static int oss_testDefault(void)
-{
-	int fd, i;
-
-	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
-		if ((fd = open(default_devices[i], O_WRONLY)) >= 0) {
-			xclose(fd);
-			return 0;
-		}
-		WARNING("Error opening OSS device \"%s\": %s\n",
-		        default_devices[i], strerror(errno));
-	}
-
-	return -1;
-}
-
-static int oss_open_default(AudioOutput *ao, ConfigParam *param, OssData *od)
-{
-	int i;
-	int err[ARRAY_SIZE(default_devices)];
-	int ret[ARRAY_SIZE(default_devices)];
-
-	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
-		ret[i] = oss_statDevice(default_devices[i], &err[i]);
-		if (ret[i] == 0) {
-			od->device = default_devices[i];
-			return 0;
-		}
-	}
-
-	if (param)
-		ERROR("error trying to open specified OSS device"
-	              " at line %i\n", param->line);
-	else
-		ERROR("error trying to open default OSS device\n");
-
-	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
-		const char *dev = default_devices[i];
-		switch(ret[i]) {
-		case OSS_STAT_DOESN_T_EXIST:
-			ERROR("%s not found\n", dev);
-			break;
-		case OSS_STAT_NOT_CHAR_DEV:
-			ERROR("%s is not a character device\n", dev);
-			break;
-		case OSS_STAT_NO_PERMS:
-			ERROR("%s: permission denied\n", dev);
-			break;
-		default:
-			ERROR("Error accessing %s: %s", dev, strerror(err[i]));
-		}
-	}
-	exit(EXIT_FAILURE);
-	return 0; /* some compilers can be dumb... */
-}
-
-static int oss_initDriver(AudioOutput * audioOutput, ConfigParam * param)
-{
-	OssData *od = newOssData();
-	audioOutput->data = od;
-	if (param) {
-		BlockParam *bp = getBlockParam(param, "device");
-		if (bp) {
-			od->device = bp->value;
-			return 0;
-		}
-	}
-	return oss_open_default(audioOutput, param, od);
-}
-
-static void oss_finishDriver(AudioOutput * audioOutput)
-{
-	OssData *od = audioOutput->data;
-
-	freeOssData(od);
-}
-
-static int setParam(OssData * od, int param, int *value)
-{
-	int val = *value;
-	int copy;
-	int supported = isSupportedParam(od, param, val);
-
-	do {
-		if (supported == OSS_UNSUPPORTED) {
-			val = getSupportedParam(od, param, val);
-			if (copy < 0)
-				return -1;
-		}
-		copy = val;
-		if (ioctl(od->fd, param, &copy)) {
-			unsupportParam(od, param, val);
-			supported = OSS_UNSUPPORTED;
-		} else {
-			if (supported == OSS_UNKNOWN) {
-				supportParam(od, param, val);
-				supported = OSS_SUPPORTED;
-			}
-			val = copy;
-		}
-	} while (supported == OSS_UNSUPPORTED);
-
-	*value = val;
-
-	return 0;
-}
-
-static void oss_close(OssData * od)
-{
-	if (od->fd >= 0)
-		while (close(od->fd) && errno == EINTR) ;
-	od->fd = -1;
-}
-
-static int oss_open(AudioOutput * audioOutput)
-{
-	int tmp;
-	OssData *od = audioOutput->data;
-
-	if ((od->fd = open(od->device, O_WRONLY)) < 0) {
-		ERROR("Error opening OSS device \"%s\": %s\n", od->device,
-		      strerror(errno));
-		goto fail;
-	}
-
-	if (setParam(od, SNDCTL_DSP_CHANNELS, &od->channels)) {
-		ERROR("OSS device \"%s\" does not support %i channels: %s\n",
-		      od->device, od->channels, strerror(errno));
-		goto fail;
-	}
-
-	if (setParam(od, SNDCTL_DSP_SPEED, &od->sampleRate)) {
-		ERROR("OSS device \"%s\" does not support %i Hz audio: %s\n",
-		      od->device, od->sampleRate, strerror(errno));
-		goto fail;
-	}
-
-	switch (od->bits) {
-	case 8:
-		tmp = AFMT_S8;
-		break;
-	case 16:
-		tmp = AFMT_S16_MPD;
-	}
-
-	if (setParam(od, SNDCTL_DSP_SAMPLESIZE, &tmp)) {
-		ERROR("OSS device \"%s\" does not support %i bit audio: %s\n",
-		      od->device, tmp, strerror(errno));
-		goto fail;
-	}
-
-	audioOutput->open = 1;
-
-	return 0;
-
-fail:
-	oss_close(od);
-	audioOutput->open = 0;
-	return -1;
-}
-
-static int oss_openDevice(AudioOutput * audioOutput)
-{
-	int ret = -1;
-	OssData *od = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
-
-	od->channels = audioFormat->channels;
-	od->sampleRate = audioFormat->sampleRate;
-	od->bits = audioFormat->bits;
-
-	if ((ret = oss_open(audioOutput)) < 0)
-		return ret;
-
-	audioFormat->channels = od->channels;
-	audioFormat->sampleRate = od->sampleRate;
-	audioFormat->bits = od->bits;
-
-	DEBUG("oss device \"%s\" will be playing %i bit %i channel audio at "
-	      "%i Hz\n", od->device, od->bits, od->channels, od->sampleRate);
-
-	return ret;
-}
-
-static void oss_closeDevice(AudioOutput * audioOutput)
-{
-	OssData *od = audioOutput->data;
-
-	oss_close(od);
-
-	audioOutput->open = 0;
-}
-
-static void oss_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	OssData *od = audioOutput->data;
-
-	if (od->fd >= 0) {
-		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
-		oss_close(od);
-	}
-}
-
-static int oss_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
-{
-	OssData *od = audioOutput->data;
-	int ret;
-
-	/* reopen the device since it was closed by dropBufferedAudio */
-	if (od->fd < 0 && oss_open(audioOutput) < 0)
-		return -1;
-
-	while (size > 0) {
-		ret = write(od->fd, playChunk, size);
-		if (ret < 0) {
-			if (errno == EINTR)
-				continue;
-			ERROR("closing oss device \"%s\" due to write error: "
-			      "%s\n", od->device, strerror(errno));
-			oss_closeDevice(audioOutput);
-			return -1;
-		}
-		playChunk += ret;
-		size -= ret;
-	}
-
-	return 0;
-}
-
-AudioOutputPlugin ossPlugin = {
-	"oss",
-	oss_testDefault,
-	oss_initDriver,
-	oss_finishDriver,
-	oss_openDevice,
-	oss_playAudio,
-	oss_dropBufferedAudio,
-	oss_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else /* HAVE OSS */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
-#endif /* HAVE_OSS */
diff --git a/trunk/src/audioOutputs/audioOutput_osx.c b/trunk/src/audioOutputs/audioOutput_osx.c
deleted file mode 100644
index 1caebade5..000000000
--- a/trunk/src/audioOutputs/audioOutput_osx.c
+++ /dev/null
@@ -1,374 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#ifdef HAVE_OSX
-
-#include <AudioUnit/AudioUnit.h>
-#include <stdlib.h>
-#include <pthread.h>
-
-#include "../log.h"
-
-typedef struct _OsxData {
-	AudioUnit au;
-	pthread_mutex_t mutex;
-	pthread_cond_t condition;
-	char *buffer;
-	int bufferSize;
-	int pos;
-	int len;
-	int started;
-} OsxData;
-
-static OsxData *newOsxData()
-{
-	OsxData *ret = xmalloc(sizeof(OsxData));
-
-	pthread_mutex_init(&ret->mutex, NULL);
-	pthread_cond_init(&ret->condition, NULL);
-
-	ret->pos = 0;
-	ret->len = 0;
-	ret->started = 0;
-	ret->buffer = NULL;
-	ret->bufferSize = 0;
-
-	return ret;
-}
-
-static int osx_testDefault()
-{
-	/*AudioUnit au;
-	   ComponentDescription desc;
-	   Component comp;
-
-	   desc.componentType = kAudioUnitType_Output;
-	   desc.componentSubType = kAudioUnitSubType_Output;
-	   desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-	   desc.componentFlags = 0;
-	   desc.componentFlagsMask = 0;
-
-	   comp = FindNextComponent(NULL, &desc);
-	   if(!comp) {
-	   ERROR("Unable to open default OS X defice\n");
-	   return -1;
-	   }
-
-	   if(OpenAComponent(comp, &au) != noErr) {
-	   ERROR("Unable to open default OS X defice\n");
-	   return -1;
-	   }
-
-	   CloseComponent(au); */
-
-	return 0;
-}
-
-static int osx_initDriver(AudioOutput * audioOutput, ConfigParam * param)
-{
-	OsxData *od = newOsxData();
-
-	audioOutput->data = od;
-
-	return 0;
-}
-
-static void freeOsxData(OsxData * od)
-{
-	if (od->buffer)
-		free(od->buffer);
-	pthread_mutex_destroy(&od->mutex);
-	pthread_cond_destroy(&od->condition);
-	free(od);
-}
-
-static void osx_finishDriver(AudioOutput * audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-	freeOsxData(od);
-}
-
-static void osx_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-
-	pthread_mutex_lock(&od->mutex);
-	od->len = 0;
-	pthread_mutex_unlock(&od->mutex);
-}
-
-static void osx_closeDevice(AudioOutput * audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-
-	pthread_mutex_lock(&od->mutex);
-	while (od->len) {
-		pthread_cond_wait(&od->condition, &od->mutex);
-	}
-	pthread_mutex_unlock(&od->mutex);
-
-	if (od->started) {
-		AudioOutputUnitStop(od->au);
-		od->started = 0;
-	}
-
-	CloseComponent(od->au);
-	AudioUnitUninitialize(od->au);
-
-	audioOutput->open = 0;
-}
-
-static OSStatus osx_render(void *vdata,
-			   AudioUnitRenderActionFlags * ioActionFlags,
-			   const AudioTimeStamp * inTimeStamp,
-			   UInt32 inBusNumber, UInt32 inNumberFrames,
-			   AudioBufferList * bufferList)
-{
-	OsxData *od = (OsxData *) vdata;
-	AudioBuffer *buffer = &bufferList->mBuffers[0];
-	int bufferSize = buffer->mDataByteSize;
-	int bytesToCopy;
-	int curpos = 0;
-
-	/*DEBUG("osx_render: enter : %i\n", (int)bufferList->mNumberBuffers);
-	   DEBUG("osx_render: ioActionFlags: %p\n", ioActionFlags);
-	   if(ioActionFlags) {
-	   if(*ioActionFlags & kAudioUnitRenderAction_PreRender) {
-	   DEBUG("prerender\n");
-	   }
-	   if(*ioActionFlags & kAudioUnitRenderAction_PostRender) {
-	   DEBUG("post render\n");
-	   }
-	   if(*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence) {
-	   DEBUG("post render\n");
-	   }
-	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight) {
-	   DEBUG("prefilight\n");
-	   }
-	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Render) {
-	   DEBUG("render\n");
-	   }
-	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Complete) {
-	   DEBUG("complete\n");
-	   }
-	   } */
-
-	/* while(bufferSize) {
-	   DEBUG("osx_render: lock\n"); */
-	pthread_mutex_lock(&od->mutex);
-	/*
-	   DEBUG("%i:%i\n", bufferSize, od->len);
-	   while(od->go && od->len < bufferSize && 
-	   od->len < od->bufferSize)
-	   {
-	   DEBUG("osx_render: wait\n");
-	   pthread_cond_wait(&od->condition, &od->mutex);
-	   }
-	 */
-
-	bytesToCopy = od->len < bufferSize ? od->len : bufferSize;
-	bufferSize = bytesToCopy;
-	od->len -= bytesToCopy;
-
-	if (od->pos + bytesToCopy > od->bufferSize) {
-		int bytes = od->bufferSize - od->pos;
-		memcpy(buffer->mData + curpos, od->buffer + od->pos, bytes);
-		od->pos = 0;
-		curpos += bytes;
-		bytesToCopy -= bytes;
-	}
-
-	memcpy(buffer->mData + curpos, od->buffer + od->pos, bytesToCopy);
-	od->pos += bytesToCopy;
-	curpos += bytesToCopy;
-
-	if (od->pos >= od->bufferSize)
-		od->pos = 0;
-	/* DEBUG("osx_render: unlock\n"); */
-	pthread_mutex_unlock(&od->mutex);
-	pthread_cond_signal(&od->condition);
-	/* } */
-
-	buffer->mDataByteSize = bufferSize;
-
-	if (!bufferSize) {
-		my_usleep(1000);
-	}
-
-	/* DEBUG("osx_render: leave\n"); */
-	return 0;
-}
-
-static int osx_openDevice(AudioOutput * audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-	ComponentDescription desc;
-	Component comp;
-	AURenderCallbackStruct callback;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
-	AudioStreamBasicDescription streamDesc;
-
-	desc.componentType = kAudioUnitType_Output;
-	desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-	desc.componentFlags = 0;
-	desc.componentFlagsMask = 0;
-
-	comp = FindNextComponent(NULL, &desc);
-	if (comp == 0) {
-		ERROR("Error finding OS X component\n");
-		return -1;
-	}
-
-	if (OpenAComponent(comp, &od->au) != noErr) {
-		ERROR("Unable to open OS X component\n");
-		return -1;
-	}
-
-	if (AudioUnitInitialize(od->au) != 0) {
-		CloseComponent(od->au);
-		ERROR("Unable to initialize OS X audio unit\n");
-		return -1;
-	}
-
-	callback.inputProc = osx_render;
-	callback.inputProcRefCon = od;
-
-	if (AudioUnitSetProperty(od->au, kAudioUnitProperty_SetRenderCallback,
-				 kAudioUnitScope_Input, 0,
-				 &callback, sizeof(callback)) != 0) {
-		AudioUnitUninitialize(od->au);
-		CloseComponent(od->au);
-		ERROR("unable to set callback for OS X audio unit\n");
-		return -1;
-	}
-
-	streamDesc.mSampleRate = audioFormat->sampleRate;
-	streamDesc.mFormatID = kAudioFormatLinearPCM;
-	streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-#ifdef WORDS_BIGENDIAN
-	streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
-
-	streamDesc.mBytesPerPacket =
-	    audioFormat->channels * audioFormat->bits / 8;
-	streamDesc.mFramesPerPacket = 1;
-	streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket;
-	streamDesc.mChannelsPerFrame = audioFormat->channels;
-	streamDesc.mBitsPerChannel = audioFormat->bits;
-
-	if (AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
-				 kAudioUnitScope_Input, 0,
-				 &streamDesc, sizeof(streamDesc)) != 0) {
-		AudioUnitUninitialize(od->au);
-		CloseComponent(od->au);
-		ERROR("Unable to set format on OS X device\n");
-		return -1;
-	}
-
-	/* create a buffer of 1s */
-	od->bufferSize = (audioFormat->sampleRate) *
-	    (audioFormat->bits >> 3) * (audioFormat->channels);
-	od->buffer = xrealloc(od->buffer, od->bufferSize);
-
-	od->pos = 0;
-	od->len = 0;
-
-	audioOutput->open = 1;
-
-	return 0;
-}
-
-static int osx_play(AudioOutput * audioOutput, char *playChunk, int size)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-	int bytesToCopy;
-	int curpos;
-
-	/* DEBUG("osx_play: enter\n"); */
-
-	if (!od->started) {
-		int err;
-		od->started = 1;
-		err = AudioOutputUnitStart(od->au);
-		if (err) {
-			ERROR("unable to start audio output: %i\n", err);
-			return -1;
-		}
-	}
-
-	pthread_mutex_lock(&od->mutex);
-
-	while (size) {
-		/* DEBUG("osx_play: lock\n"); */
-		curpos = od->pos + od->len;
-		if (curpos >= od->bufferSize)
-			curpos -= od->bufferSize;
-
-		bytesToCopy = od->bufferSize < size ? od->bufferSize : size;
-
-		while (od->len > od->bufferSize - bytesToCopy) {
-			/* DEBUG("osx_play: wait\n"); */
-			pthread_cond_wait(&od->condition, &od->mutex);
-		}
-
-		bytesToCopy = od->bufferSize - od->len;
-		bytesToCopy = bytesToCopy < size ? bytesToCopy : size;
-		size -= bytesToCopy;
-		od->len += bytesToCopy;
-
-		if (curpos + bytesToCopy > od->bufferSize) {
-			int bytes = od->bufferSize - curpos;
-			memcpy(od->buffer + curpos, playChunk, bytes);
-			curpos = 0;
-			playChunk += bytes;
-			bytesToCopy -= bytes;
-		}
-
-		memcpy(od->buffer + curpos, playChunk, bytesToCopy);
-		curpos += bytesToCopy;
-		playChunk += bytesToCopy;
-
-	}
-	/* DEBUG("osx_play: unlock\n"); */
-	pthread_mutex_unlock(&od->mutex);
-
-	/* DEBUG("osx_play: leave\n"); */
-	return 0;
-}
-
-AudioOutputPlugin osxPlugin = {
-	"osx",
-	osx_testDefault,
-	osx_initDriver,
-	osx_finishDriver,
-	osx_openDevice,
-	osx_play,
-	osx_dropBufferedAudio,
-	osx_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else
-
-#include <stdio.h>
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
-#endif
diff --git a/trunk/src/audioOutputs/audioOutput_pulse.c b/trunk/src/audioOutputs/audioOutput_pulse.c
deleted file mode 100644
index 8948e0263..000000000
--- a/trunk/src/audioOutputs/audioOutput_pulse.c
+++ /dev/null
@@ -1,221 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#include <stdlib.h>
-
-#ifdef HAVE_PULSE
-
-#include "../conf.h"
-#include "../log.h"
-
-#include <string.h>
-#include <time.h>
-
-#include <pulse/simple.h>
-#include <pulse/error.h>
-
-#define MPD_PULSE_NAME "mpd"
-#define CONN_ATTEMPT_INTERVAL 60
-
-typedef struct _PulseData {
-	pa_simple *s;
-	char *server;
-	char *sink;
-	int connAttempts;
-	time_t lastAttempt;
-} PulseData;
-
-static PulseData *newPulseData(void)
-{
-	PulseData *ret;
-
-	ret = xmalloc(sizeof(PulseData));
-
-	ret->s = NULL;
-	ret->server = NULL;
-	ret->sink = NULL;
-	ret->connAttempts = 0;
-	ret->lastAttempt = 0;
-
-	return ret;
-}
-
-static void freePulseData(PulseData * pd)
-{
-	if (pd->server)
-		free(pd->server);
-	if (pd->sink)
-		free(pd->sink);
-	free(pd);
-}
-
-static int pulse_initDriver(AudioOutput * audioOutput, ConfigParam * param)
-{
-	BlockParam *server = NULL;
-	BlockParam *sink = NULL;
-	PulseData *pd;
-
-	if (param) {
-		server = getBlockParam(param, "server");
-		sink = getBlockParam(param, "sink");
-	}
-
-	pd = newPulseData();
-	pd->server = server ? xstrdup(server->value) : NULL;
-	pd->sink = sink ? xstrdup(sink->value) : NULL;
-	audioOutput->data = pd;
-
-	return 0;
-}
-
-static void pulse_finishDriver(AudioOutput * audioOutput)
-{
-	freePulseData((PulseData *) audioOutput->data);
-}
-
-static int pulse_testDefault(void)
-{
-	pa_simple *s;
-	pa_sample_spec ss;
-	int error;
-
-	ss.format = PA_SAMPLE_S16NE;
-	ss.rate = 44100;
-	ss.channels = 2;
-
-	s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
-			  MPD_PULSE_NAME, &ss, NULL, NULL, &error);
-	if (!s) {
-		WARNING("Cannot connect to default PulseAudio server: %s\n",
-			pa_strerror(error));
-		return -1;
-	}
-
-	pa_simple_free(s);
-
-	return 0;
-}
-
-static int pulse_openDevice(AudioOutput * audioOutput)
-{
-	PulseData *pd;
-	AudioFormat *audioFormat;
-	pa_sample_spec ss;
-	time_t t;
-	int error;
-
-	t = time(NULL);
-	pd = audioOutput->data;
-	audioFormat = &audioOutput->outAudioFormat;
-
-	if (pd->connAttempts != 0 &&
-	    (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
-		return -1;
-
-	pd->connAttempts++;
-	pd->lastAttempt = t;
-
-	if (audioFormat->bits != 16) {
-		ERROR("PulseAudio doesn't support %i bit audio\n",
-		      audioFormat->bits);
-		return -1;
-	}
-
-	ss.format = PA_SAMPLE_S16NE;
-	ss.rate = audioFormat->sampleRate;
-	ss.channels = audioFormat->channels;
-
-	pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
-			      pd->sink, audioOutput->name, &ss, NULL, NULL,
-			      &error);
-	if (!pd->s) {
-		ERROR("Cannot connect to server in PulseAudio output "
-		      "\"%s\" (attempt %i): %s\n", audioOutput->name,
-		      pd->connAttempts, pa_strerror(error));
-		return -1;
-	}
-
-	pd->connAttempts = 0;
-	audioOutput->open = 1;
-
-	DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
-	      "channel audio at %i Hz\n", audioOutput->name, audioFormat->bits,
-	      audioFormat->channels, audioFormat->sampleRate);
-
-	return 0;
-}
-
-static void pulse_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	PulseData *pd;
-	int error;
-
-	pd = audioOutput->data;
-	if (pa_simple_flush(pd->s, &error) < 0)
-		WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
-			audioOutput->name, pa_strerror(error));
-}
-
-static void pulse_closeDevice(AudioOutput * audioOutput)
-{
-	PulseData *pd;
-
-	pd = audioOutput->data;
-	if (pd->s) {
-		pa_simple_drain(pd->s, NULL);
-		pa_simple_free(pd->s);
-	}
-
-	audioOutput->open = 0;
-}
-
-static int pulse_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
-{
-	PulseData *pd;
-	int error;
-
-	pd = audioOutput->data;
-
-	if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
-		ERROR("PulseAudio output \"%s\" disconnecting due to write "
-		      "error: %s\n", audioOutput->name, pa_strerror(error));
-		pulse_closeDevice(audioOutput);
-		return -1;
-	}
-
-	return 0;
-}
-
-AudioOutputPlugin pulsePlugin = {
-	"pulse",
-	pulse_testDefault,
-	pulse_initDriver,
-	pulse_finishDriver,
-	pulse_openDevice,
-	pulse_playAudio,
-	pulse_dropBufferedAudio,
-	pulse_closeDevice,
-	NULL,	/* sendMetadataFunc */
-};
-
-#else /* HAVE_PULSE */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
-#endif /* HAVE_PULSE */
diff --git a/trunk/src/audioOutputs/audioOutput_shout.c b/trunk/src/audioOutputs/audioOutput_shout.c
deleted file mode 100644
index 7d93f8f85..000000000
--- a/trunk/src/audioOutputs/audioOutput_shout.c
+++ /dev/null
@@ -1,636 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../audioOutput.h"
-
-#include <stdlib.h>
-
-#ifdef HAVE_SHOUT
-
-#include "../conf.h"
-#include "../log.h"
-#include "../pcm_utils.h"
-
-#include <string.h>
-#include <time.h>
-
-#include <shout/shout.h>
-#include <vorbis/vorbisenc.h>
-
-#define CONN_ATTEMPT_INTERVAL	60
-
-static int shoutInitCount;
-
-/* lots of this code blatantly stolent from bossogg/bossao2 */
-
-typedef struct _ShoutData {
-	shout_t *shoutConn;
-	int shoutError;
-
-	ogg_stream_state os;
-	ogg_page og;
-	ogg_packet op;
-	ogg_packet header_main;
-	ogg_packet header_comments;
-	ogg_packet header_codebooks;
-
-	vorbis_dsp_state vd;
-	vorbis_block vb;
-	vorbis_info vi;
-	vorbis_comment vc;
-
-	float quality;
-	int bitrate;
-
-	int opened;
-
-	MpdTag *tag;
-	int tagToSend;
-
-	int connAttempts;
-	time_t lastAttempt;
-	int last_err;
-
-	/* just a pointer to audioOutput->outAudioFormat */
-	AudioFormat *audioFormat;
-} ShoutData;
-
-static ShoutData *newShoutData(void)
-{
-	ShoutData *ret = xmalloc(sizeof(ShoutData));
-
-	ret->shoutConn = shout_new();
-	ret->opened = 0;
-	ret->tag = NULL;
-	ret->tagToSend = 0;
-	ret->bitrate = -1;
-	ret->quality = -2.0;
-	ret->connAttempts = 0;
-	ret->lastAttempt = 0;
-	ret->audioFormat = NULL;
-	ret->last_err = SHOUTERR_UNCONNECTED;
-
-	return ret;
-}
-
-static void freeShoutData(ShoutData * sd)
-{
-	if (sd->shoutConn)
-		shout_free(sd->shoutConn);
-	if (sd->tag)
-		freeMpdTag(sd->tag);
-
-	free(sd);
-}
-
-#define checkBlockParam(name) { \
-	blockParam = getBlockParam(param, name); \
-	if (!blockParam) { \
-		FATAL("no \"%s\" defined for shout device defined at line " \
-				"%i\n", name, param->line); \
-	} \
-}
-
-static int myShout_initDriver(AudioOutput * audioOutput, ConfigParam * param)
-{
-	ShoutData *sd;
-	char *test;
-	int port;
-	char *host;
-	char *mount;
-	char *passwd;
-	char *user;
-	char *name;
-	BlockParam *blockParam;
-	unsigned int public = 0;
-
-	sd = newShoutData();
-
-	if (shoutInitCount == 0)
-		shout_init();
-
-	shoutInitCount++;
-
-	checkBlockParam("host");
-	host = blockParam->value;
-
-	checkBlockParam("mount");
-	mount = blockParam->value;
-
-	checkBlockParam("port");
-
-	port = strtol(blockParam->value, &test, 10);
-
-	if (*test != '\0' || port <= 0) {
-		FATAL("shout port \"%s\" is not a positive integer, line %i\n",
-		      blockParam->value, blockParam->line);
-	}
-
-	checkBlockParam("password");
-	passwd = blockParam->value;
-
-	checkBlockParam("name");
-	name = blockParam->value;
-
-	blockParam = getBlockParam(param, "public");
-	if (blockParam) {
-		if (0 == strcmp(blockParam->value, "yes")) {
-			public = 1;
-		} else if (0 == strcmp(blockParam->value, "no")) {
-			public = 0;
-		} else {
-			FATAL("public \"%s\" is not \"yes\" or \"no\" at line "
-			      "%i\n", param->value, param->line);
-		}
-	}
-
-	blockParam = getBlockParam(param, "user");
-	if (blockParam)
-		user = blockParam->value;
-	else
-		user = "source";
-
-	blockParam = getBlockParam(param, "quality");
-
-	if (blockParam) {
-		int line = blockParam->line;
-
-		sd->quality = strtod(blockParam->value, &test);
-
-		if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
-			FATAL("shout quality \"%s\" is not a number in the "
-			      "range -1 to 10, line %i\n", blockParam->value,
-			      blockParam->line);
-		}
-
-		blockParam = getBlockParam(param, "bitrate");
-
-		if (blockParam) {
-			FATAL("quality (line %i) and bitrate (line %i) are "
-			      "both defined for shout output\n", line,
-			      blockParam->line);
-		}
-	} else {
-		blockParam = getBlockParam(param, "bitrate");
-
-		if (!blockParam) {
-			FATAL("neither bitrate nor quality defined for shout "
-			      "output at line %i\n", param->line);
-		}
-
-		sd->bitrate = strtol(blockParam->value, &test, 10);
-
-		if (*test != '\0' || sd->bitrate <= 0) {
-			FATAL("bitrate at line %i should be a positive integer "
-			      "\n", blockParam->line);
-		}
-	}
-
-	checkBlockParam("format");
-	sd->audioFormat = &audioOutput->outAudioFormat;
-
-	if (shout_set_host(sd->shoutConn, host) != SHOUTERR_SUCCESS ||
-	    shout_set_port(sd->shoutConn, port) != SHOUTERR_SUCCESS ||
-	    shout_set_password(sd->shoutConn, passwd) != SHOUTERR_SUCCESS ||
-	    shout_set_mount(sd->shoutConn, mount) != SHOUTERR_SUCCESS ||
-	    shout_set_name(sd->shoutConn, name) != SHOUTERR_SUCCESS ||
-	    shout_set_user(sd->shoutConn, user) != SHOUTERR_SUCCESS ||
-	    shout_set_public(sd->shoutConn, public) != SHOUTERR_SUCCESS ||
-	    shout_set_nonblocking(sd->shoutConn, 1) != SHOUTERR_SUCCESS ||
-	    shout_set_format(sd->shoutConn, SHOUT_FORMAT_VORBIS)
-	    != SHOUTERR_SUCCESS ||
-	    shout_set_protocol(sd->shoutConn, SHOUT_PROTOCOL_HTTP)
-	    != SHOUTERR_SUCCESS ||
-	    shout_set_agent(sd->shoutConn, "MPD") != SHOUTERR_SUCCESS) {
-		FATAL("error configuring shout defined at line %i: %s\n",
-		      param->line, shout_get_error(sd->shoutConn));
-	}
-
-	/* optional paramters */
-	blockParam = getBlockParam(param, "genre");
-	if (blockParam && shout_set_genre(sd->shoutConn, blockParam->value)) {
-		FATAL("error configuring shout defined at line %i: %s\n",
-		      param->line, shout_get_error(sd->shoutConn));
-	}
-
-	blockParam = getBlockParam(param, "description");
-	if (blockParam && shout_set_description(sd->shoutConn,
-						blockParam->value)) {
-		FATAL("error configuring shout defined at line %i: %s\n",
-		      param->line, shout_get_error(sd->shoutConn));
-	}
-
-	{
-		char temp[11];
-		memset(temp, 0, sizeof(temp));
-
-		snprintf(temp, sizeof(temp), "%d", sd->audioFormat->channels);
-		shout_set_audio_info(sd->shoutConn, SHOUT_AI_CHANNELS, temp);
-
-		snprintf(temp, sizeof(temp), "%d", sd->audioFormat->sampleRate);
-
-		shout_set_audio_info(sd->shoutConn, SHOUT_AI_SAMPLERATE, temp);
-
-		if (sd->quality >= -1.0) {
-			snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
-			shout_set_audio_info(sd->shoutConn, SHOUT_AI_QUALITY,
-					     temp);
-		} else {
-			snprintf(temp, sizeof(temp), "%d", sd->bitrate);
-			shout_set_audio_info(sd->shoutConn, SHOUT_AI_BITRATE,
-					     temp);
-		}
-	}
-
-	audioOutput->data = sd;
-
-	return 0;
-}
-
-static int myShout_handleError(ShoutData * sd, int err)
-{
-	switch (err) {
-	case SHOUTERR_SUCCESS:
-		break;
-	case SHOUTERR_UNCONNECTED:
-	case SHOUTERR_SOCKET:
-		ERROR("Lost shout connection to %s:%i : %s\n",
-		      shout_get_host(sd->shoutConn),
-		      shout_get_port(sd->shoutConn),
-		      shout_get_error(sd->shoutConn));
-		sd->shoutError = 1;
-		return -1;
-	default:
-		ERROR("shout: connection to %s:%i error : %s\n",
-		      shout_get_host(sd->shoutConn),
-		      shout_get_port(sd->shoutConn),
-		      shout_get_error(sd->shoutConn));
-		sd->shoutError = 1;
-		return -1;
-	}
-
-	return 0;
-}
-
-static int write_page(ShoutData * sd)
-{
-	int err = 0;
-
-	/*DEBUG("shout_delay: %i\n", shout_delay(sd->shoutConn)); */
-	shout_sync(sd->shoutConn);
-	err = shout_send(sd->shoutConn, sd->og.header, sd->og.header_len);
-	if (myShout_handleError(sd, err) < 0)
-		return -1;
-	err = shout_send(sd->shoutConn, sd->og.body, sd->og.body_len);
-	if (myShout_handleError(sd, err) < 0)
-		return -1;
-
-	return 0;
-}
-
-static void finishEncoder(ShoutData * sd)
-{
-	vorbis_analysis_wrote(&sd->vd, 0);
-
-	while (vorbis_analysis_blockout(&sd->vd, &sd->vb) == 1) {
-		vorbis_analysis(&sd->vb, NULL);
-		vorbis_bitrate_addblock(&sd->vb);
-		while (vorbis_bitrate_flushpacket(&sd->vd, &sd->op)) {
-			ogg_stream_packetin(&sd->os, &sd->op);
-		}
-	}
-}
-
-static int flushEncoder(ShoutData * sd)
-{
-	return (ogg_stream_pageout(&sd->os, &sd->og) > 0);
-}
-
-static void clearEncoder(ShoutData * sd)
-{
-	finishEncoder(sd);
-	while (1 == flushEncoder(sd)) {
-		if (!sd->shoutError)
-			write_page(sd);
-	}
-
-	vorbis_comment_clear(&sd->vc);
-	ogg_stream_clear(&sd->os);
-	vorbis_block_clear(&sd->vb);
-	vorbis_dsp_clear(&sd->vd);
-	vorbis_info_clear(&sd->vi);
-}
-
-static void myShout_closeShoutConn(ShoutData * sd)
-{
-	if (sd->opened) {
-		clearEncoder(sd);
-
-		if (shout_close(sd->shoutConn) != SHOUTERR_SUCCESS) {
-			ERROR("problem closing connection to shout server: "
-			      "%s\n", shout_get_error(sd->shoutConn));
-		}
-	}
-
-	sd->last_err = SHOUTERR_UNCONNECTED;
-	sd->opened = 0;
-}
-
-static void myShout_finishDriver(AudioOutput * audioOutput)
-{
-	ShoutData *sd = (ShoutData *) audioOutput->data;
-
-	myShout_closeShoutConn(sd);
-
-	freeShoutData(sd);
-
-	shoutInitCount--;
-
-	if (shoutInitCount == 0)
-		shout_shutdown();
-}
-
-static void myShout_dropBufferedAudio(AudioOutput * audioOutput)
-{
-	/* needs to be implemented */
-}
-
-static void myShout_closeDevice(AudioOutput * audioOutput)
-{
-	ShoutData *sd = (ShoutData *) audioOutput->data;
-
-	myShout_closeShoutConn(sd);
-
-	audioOutput->open = 0;
-}
-
-#define addTag(name, value) { \
-	if(value) vorbis_comment_add_tag(&(sd->vc), name, value); \
-}
-
-static void copyTagToVorbisComment(ShoutData * sd)
-{
-	if (sd->tag) {
-		int i;
-
-		for (i = 0; i < sd->tag->numOfItems; i++) {
-			switch (sd->tag->items[i].type) {
-			case TAG_ITEM_ARTIST:
-				addTag("ARTIST", sd->tag->items[i].value);
-				break;
-			case TAG_ITEM_ALBUM:
-				addTag("ALBUM", sd->tag->items[i].value);
-				break;
-			case TAG_ITEM_TITLE:
-				addTag("TITLE", sd->tag->items[i].value);
-				break;
-			}
-		}
-	}
-}
-
-static int initEncoder(ShoutData * sd)
-{
-	vorbis_info_init(&(sd->vi));
-
-	if (sd->quality >= -1.0) {
-		if (0 != vorbis_encode_init_vbr(&(sd->vi),
-						sd->audioFormat->channels,
-						sd->audioFormat->sampleRate,
-						sd->quality * 0.1)) {
-			ERROR("problem setting up vorbis encoder for shout\n");
-			vorbis_info_clear(&(sd->vi));
-			return -1;
-		}
-	} else {
-		if (0 != vorbis_encode_init(&(sd->vi),
-					    sd->audioFormat->channels,
-					    sd->audioFormat->sampleRate, -1.0,
-					    sd->bitrate * 1000, -1.0)) {
-			ERROR("problem setting up vorbis encoder for shout\n");
-			vorbis_info_clear(&(sd->vi));
-			return -1;
-		}
-	}
-
-	vorbis_analysis_init(&(sd->vd), &(sd->vi));
-	vorbis_block_init(&(sd->vd), &(sd->vb));
-
-	ogg_stream_init(&(sd->os), rand());
-
-	vorbis_comment_init(&(sd->vc));
-
-	return 0;
-}
-
-static int myShout_openShoutConn(AudioOutput * audioOutput)
-{
-	ShoutData *sd = (ShoutData *) audioOutput->data;
-	time_t t = time(NULL);
-
-	if (sd->connAttempts != 0 &&
-	    (t - sd->lastAttempt) < CONN_ATTEMPT_INTERVAL) {
-		return -1;
-	}
-
-	sd->connAttempts++;
-
-	if (sd->last_err == SHOUTERR_UNCONNECTED)
-		sd->last_err = shout_open(sd->shoutConn);
-	switch (sd->last_err) {
-	case SHOUTERR_SUCCESS:
-	case SHOUTERR_CONNECTED:
-		break;
-	case SHOUTERR_BUSY:
-		sd->last_err = shout_get_connected(sd->shoutConn);
-		if (sd->last_err == SHOUTERR_CONNECTED)
-			break;
-		return -1;
-	default:
-		sd->lastAttempt = t;
-		ERROR("problem opening connection to shout server %s:%i "
-		      "(attempt %i): %s\n",
-		      shout_get_host(sd->shoutConn),
-		      shout_get_port(sd->shoutConn),
-		      sd->connAttempts, shout_get_error(sd->shoutConn));
-		return -1;
-	}
-
-	if (initEncoder(sd) < 0) {
-		shout_close(sd->shoutConn);
-		return -1;
-	}
-
-	sd->shoutError = 0;
-
-	copyTagToVorbisComment(sd);
-
-	vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
-				  &(sd->header_comments),
-				  &(sd->header_codebooks));
-
-	ogg_stream_packetin(&(sd->os), &(sd->header_main));
-	ogg_stream_packetin(&(sd->os), &(sd->header_comments));
-	ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
-
-	sd->opened = 1;
-	sd->tagToSend = 0;
-
-	while (ogg_stream_flush(&(sd->os), &(sd->og))) {
-		if (write_page(sd) < 0) {
-			myShout_closeShoutConn(sd);
-			return -1;
-		}
-	}
-
-	sd->connAttempts = 0;
-
-	return 0;
-}
-
-static int myShout_openDevice(AudioOutput * audioOutput)
-{
-	ShoutData *sd = (ShoutData *) audioOutput->data;
-
-	audioOutput->open = 1;
-
-	if (sd->opened)
-		return 0;
-
-	if (myShout_openShoutConn(audioOutput) < 0) {
-		audioOutput->open = 0;
-		return -1;
-	}
-
-	return 0;
-}
-
-static void myShout_sendMetadata(ShoutData * sd)
-{
-	if (!sd->opened || !sd->tag)
-		return;
-
-	clearEncoder(sd);
-	if (initEncoder(sd) < 0)
-		return;
-
-	copyTagToVorbisComment(sd);
-
-	vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
-				  &(sd->header_comments),
-				  &(sd->header_codebooks));
-
-	ogg_stream_packetin(&(sd->os), &(sd->header_main));
-	ogg_stream_packetin(&(sd->os), &(sd->header_comments));
-	ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
-
-	while (ogg_stream_flush(&(sd->os), &(sd->og))) {
-		if (write_page(sd) < 0) {
-			myShout_closeShoutConn(sd);
-			return;
-		}
-	}
-
-	/*if(sd->tag) freeMpdTag(sd->tag);
-	   sd->tag = NULL; */
-	sd->tagToSend = 0;
-}
-
-static int myShout_play(AudioOutput * audioOutput, char *playChunk, int size)
-{
-	int i, j;
-	ShoutData *sd = (ShoutData *) audioOutput->data;
-	float **vorbbuf;
-	int samples;
-	int bytes = sd->audioFormat->bits / 8;
-
-	if (sd->opened && sd->tagToSend)
-		myShout_sendMetadata(sd);
-
-	if (!sd->opened) {
-		if (myShout_openShoutConn(audioOutput) < 0) {
-			return -1;
-		}
-	}
-
-	samples = size / (bytes * sd->audioFormat->channels);
-
-	/* this is for only 16-bit audio */
-
-	vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples);
-
-	for (i = 0; i < samples; i++) {
-		for (j = 0; j < sd->audioFormat->channels; j++) {
-			vorbbuf[j][i] = (*((mpd_sint16 *) playChunk)) / 32768.0;
-			playChunk += bytes;
-		}
-	}
-
-	vorbis_analysis_wrote(&(sd->vd), samples);
-
-	while (1 == vorbis_analysis_blockout(&(sd->vd), &(sd->vb))) {
-		vorbis_analysis(&(sd->vb), NULL);
-		vorbis_bitrate_addblock(&(sd->vb));
-
-		while (vorbis_bitrate_flushpacket(&(sd->vd), &(sd->op))) {
-			ogg_stream_packetin(&(sd->os), &(sd->op));
-		}
-	}
-
-	while (ogg_stream_pageout(&(sd->os), &(sd->og)) != 0) {
-		if (write_page(sd) < 0) {
-			myShout_closeShoutConn(sd);
-			return -1;
-		}
-	}
-
-	return 0;
-}
-
-static void myShout_setTag(AudioOutput * audioOutput, MpdTag * tag)
-{
-	ShoutData *sd = (ShoutData *) audioOutput->data;
-
-	if (sd->tag)
-		freeMpdTag(sd->tag);
-	sd->tag = NULL;
-	sd->tagToSend = 0;
-
-	if (!tag)
-		return;
-
-	sd->tag = mpdTagDup(tag);
-	sd->tagToSend = 1;
-}
-
-AudioOutputPlugin shoutPlugin = {
-	"shout",
-	NULL,
-	myShout_initDriver,
-	myShout_finishDriver,
-	myShout_openDevice,
-	myShout_play,
-	myShout_dropBufferedAudio,
-	myShout_closeDevice,
-	myShout_setTag,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
-#endif