renamed src/inputPlugins/ to src/decoder/

These plugins are not input plugins, they are decoder plugins.  No
CamelCase in the directory name.

15 files changed, 5609 insertions, 0 deletions

diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
new file mode 100644
index 000000000..db43e0003
--- /dev/null
+++ b/src/decoder/_flac_common.c
@@ -0,0 +1,323 @@
+/* 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
+ *
+ * Common data structures and functions used by FLAC and OggFLAC
+ * (c) 2005 by Eric Wong <normalperson@yhbt.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 "_flac_common.h"
+#include "../log.h"
+
+#include <FLAC/format.h>
+#include <FLAC/metadata.h>
+
+void init_FlacData(FlacData * data, struct decoder * decoder,
+		   InputStream * inStream)
+{
+	data->time = 0;
+	data->position = 0;
+	data->bitRate = 0;
+	data->decoder = decoder;
+	data->inStream = inStream;
+	data->replayGainInfo = NULL;
+	data->tag = NULL;
+}
+
+static int flacFindVorbisCommentFloat(const FLAC__StreamMetadata * block,
+				      const char *cmnt, float *fl)
+{
+	int offset =
+	    FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, cmnt);
+
+	if (offset >= 0) {
+		size_t pos = strlen(cmnt) + 1;	/* 1 is for '=' */
+		int len = block->data.vorbis_comment.comments[offset].length
+		    - pos;
+		if (len > 0) {
+			unsigned char tmp;
+			unsigned char *p = &(block->data.vorbis_comment.
+					     comments[offset].entry[pos]);
+			tmp = p[len];
+			p[len] = '\0';
+			*fl = (float)atof((char *)p);
+			p[len] = tmp;
+
+			return 1;
+		}
+	}
+
+	return 0;
+}
+
+/* replaygain stuff by AliasMrJones */
+static void flacParseReplayGain(const FLAC__StreamMetadata * block,
+				FlacData * data)
+{
+	int found = 0;
+
+	if (data->replayGainInfo)
+		freeReplayGainInfo(data->replayGainInfo);
+
+	data->replayGainInfo = newReplayGainInfo();
+
+	found |= flacFindVorbisCommentFloat(block, "replaygain_album_gain",
+					    &data->replayGainInfo->albumGain);
+	found |= flacFindVorbisCommentFloat(block, "replaygain_album_peak",
+					    &data->replayGainInfo->albumPeak);
+	found |= flacFindVorbisCommentFloat(block, "replaygain_track_gain",
+					    &data->replayGainInfo->trackGain);
+	found |= flacFindVorbisCommentFloat(block, "replaygain_track_peak",
+					    &data->replayGainInfo->trackPeak);
+
+	if (!found) {
+		freeReplayGainInfo(data->replayGainInfo);
+		data->replayGainInfo = NULL;
+	}
+}
+
+/* tracknumber is used in VCs, MPD uses "track" ..., all the other
+ * tag names match */
+static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber";
+static const char *VORBIS_COMMENT_DISC_KEY = "discnumber";
+
+static unsigned int commentMatchesAddToTag(const
+					   FLAC__StreamMetadata_VorbisComment_Entry
+					   * entry, unsigned int itemType,
+					   struct tag ** tag)
+{
+	const char *str;
+	size_t slen;
+	int vlen;
+
+	switch (itemType) {
+	case TAG_ITEM_TRACK:
+		str = VORBIS_COMMENT_TRACK_KEY;
+		break;
+	case TAG_ITEM_DISC:
+		str = VORBIS_COMMENT_DISC_KEY;
+		break;
+	default:
+		str = mpdTagItemKeys[itemType];
+	}
+	slen = strlen(str);
+	vlen = entry->length - slen - 1;
+
+	if ((vlen > 0) && (0 == strncasecmp(str, (char *)entry->entry, slen))
+	    && (*(entry->entry + slen) == '=')) {
+		if (!*tag)
+			*tag = tag_new();
+
+		tag_add_item_n(*tag, itemType,
+			       (char *)(entry->entry + slen + 1), vlen);
+
+		return 1;
+	}
+
+	return 0;
+}
+
+struct tag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
+					   struct tag * tag)
+{
+	unsigned int i, j;
+	FLAC__StreamMetadata_VorbisComment_Entry *comments;
+
+	comments = block->data.vorbis_comment.comments;
+
+	for (i = block->data.vorbis_comment.num_comments; i != 0; --i) {
+		for (j = TAG_NUM_OF_ITEM_TYPES; j--;) {
+			if (commentMatchesAddToTag(comments, j, &tag))
+				break;
+		}
+		comments++;
+	}
+
+	return tag;
+}
+
+void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
+			     FlacData * data)
+{
+	const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info);
+
+	switch (block->type) {
+	case FLAC__METADATA_TYPE_STREAMINFO:
+		data->audio_format.bits = (int8_t)si->bits_per_sample;
+		data->audio_format.sample_rate = si->sample_rate;
+		data->audio_format.channels = (int8_t)si->channels;
+		data->total_time = ((float)si->total_samples) / (si->sample_rate);
+		break;
+	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+		flacParseReplayGain(block, data);
+	default:
+		break;
+	}
+}
+
+void flac_error_common_cb(const char *plugin,
+			  const FLAC__StreamDecoderErrorStatus status,
+			  mpd_unused FlacData * data)
+{
+	if (decoder_get_command(data->decoder) == DECODE_COMMAND_STOP)
+		return;
+
+	switch (status) {
+	case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC:
+		ERROR("%s lost sync\n", plugin);
+		break;
+	case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER:
+		ERROR("bad %s header\n", plugin);
+		break;
+	case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH:
+		ERROR("%s crc mismatch\n", plugin);
+		break;
+	default:
+		ERROR("unknown %s error\n", plugin);
+	}
+}
+
+static void flac_convert_stereo16(int16_t *dest,
+				  const FLAC__int32 * const buf[],
+				  unsigned int position, unsigned int end)
+{
+	for (; position < end; ++position) {
+		*dest++ = buf[0][position];
+		*dest++ = buf[1][position];
+	}
+}
+
+static void
+flac_convert_16(int16_t *dest,
+		unsigned int num_channels,
+		const FLAC__int32 * const buf[],
+		unsigned int position, unsigned int end)
+{
+	unsigned int c_chan;
+
+	for (; position < end; ++position)
+		for (c_chan = 0; c_chan < num_channels; c_chan++)
+			*dest++ = buf[c_chan][position];
+}
+
+/**
+ * Note: this function also handles 24 bit files!
+ */
+static void
+flac_convert_32(int32_t *dest,
+		unsigned int num_channels,
+		const FLAC__int32 * const buf[],
+		unsigned int position, unsigned int end)
+{
+	unsigned int c_chan;
+
+	for (; position < end; ++position)
+		for (c_chan = 0; c_chan < num_channels; c_chan++)
+			*dest++ = buf[c_chan][position];
+}
+
+static void
+flac_convert_8(int8_t *dest,
+	       unsigned int num_channels,
+	       const FLAC__int32 * const buf[],
+	       unsigned int position, unsigned int end)
+{
+	unsigned int c_chan;
+
+	for (; position < end; ++position)
+		for (c_chan = 0; c_chan < num_channels; c_chan++)
+			*dest++ = buf[c_chan][position];
+}
+
+static void flac_convert(unsigned char *dest,
+			 unsigned int num_channels,
+			 unsigned int bytes_per_sample,
+			 const FLAC__int32 * const buf[],
+			 unsigned int position, unsigned int end)
+{
+	switch (bytes_per_sample) {
+	case 2:
+		if (num_channels == 2)
+			flac_convert_stereo16((int16_t*)dest, buf,
+					      position, end);
+		else
+			flac_convert_16((int16_t*)dest, num_channels, buf,
+					position, end);
+		break;
+
+	case 4:
+		flac_convert_32((int32_t*)dest, num_channels, buf,
+				position, end);
+		break;
+
+	case 1:
+		flac_convert_8((int8_t*)dest, num_channels, buf,
+			       position, end);
+		break;
+	}
+}
+
+FLAC__StreamDecoderWriteStatus
+flac_common_write(FlacData *data, const FLAC__Frame * frame,
+		  const FLAC__int32 *const buf[])
+{
+	unsigned int c_samp;
+	const unsigned int num_channels = frame->header.channels;
+	const unsigned int bytes_per_sample =
+		audio_format_sample_size(&data->audio_format);
+	const unsigned int bytes_per_channel =
+		bytes_per_sample * frame->header.channels;
+	const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
+	unsigned int num_samples;
+	enum decoder_command cmd;
+
+	if (bytes_per_sample != 1 && bytes_per_sample != 2 &&
+	    bytes_per_sample != 4)
+		/* exotic unsupported bit rate */
+		return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
+
+	for (c_samp = 0; c_samp < frame->header.blocksize;
+	     c_samp += num_samples) {
+		num_samples = frame->header.blocksize - c_samp;
+		if (num_samples > max_samples)
+			num_samples = max_samples;
+
+		flac_convert(data->chunk,
+			     num_channels, bytes_per_sample, buf,
+			     c_samp, c_samp + num_samples);
+
+		cmd = decoder_data(data->decoder, data->inStream,
+				   1, data->chunk,
+				   num_samples * bytes_per_channel,
+				   data->time, data->bitRate,
+				   data->replayGainInfo);
+		switch (cmd) {
+		case DECODE_COMMAND_NONE:
+		case DECODE_COMMAND_START:
+			break;
+
+		case DECODE_COMMAND_STOP:
+			return
+				FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
+
+		case DECODE_COMMAND_SEEK:
+			return
+				FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+		}
+	}
+
+	return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+}
diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h
new file mode 100644
index 000000000..45714b4bd
--- /dev/null
+++ b/src/decoder/_flac_common.h
@@ -0,0 +1,168 @@
+/* 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
+ *
+ * Common data structures and functions used by FLAC and OggFLAC
+ * (c) 2005 by Eric Wong <normalperson@yhbt.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
+ */
+
+#ifndef _FLAC_COMMON_H
+#define _FLAC_COMMON_H
+
+#include "../decoder_api.h"
+
+#include <FLAC/export.h>
+#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+#  include <FLAC/seekable_stream_decoder.h>
+#  define flac_decoder           FLAC__SeekableStreamDecoder
+#  define flac_new()             FLAC__seekable_stream_decoder_new()
+
+#  define flac_ogg_init(a,b,c,d,e,f,g,h,i,j) (0)
+
+#  define flac_get_decode_position(x,y) \
+                 FLAC__seekable_stream_decoder_get_decode_position(x,y)
+#  define flac_get_state(x)      FLAC__seekable_stream_decoder_get_state(x)
+#  define flac_process_single(x) FLAC__seekable_stream_decoder_process_single(x)
+#  define flac_process_metadata(x) \
+                 FLAC__seekable_stream_decoder_process_until_end_of_metadata(x)
+#  define flac_seek_absolute(x,y) \
+                 FLAC__seekable_stream_decoder_seek_absolute(x,y)
+#  define flac_finish(x)         FLAC__seekable_stream_decoder_finish(x)
+#  define flac_delete(x)         FLAC__seekable_stream_decoder_delete(x)
+
+#  define flac_decoder_eof       FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM
+
+typedef unsigned flac_read_status_size_t;
+#  define flac_read_status       FLAC__SeekableStreamDecoderReadStatus
+#  define flac_read_status_continue \
+                                 FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK
+#  define flac_read_status_eof   FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK
+#  define flac_read_status_abort \
+                               FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR
+
+#  define flac_seek_status       FLAC__SeekableStreamDecoderSeekStatus
+#  define flac_seek_status_ok    FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK
+#  define flac_seek_status_error FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR
+
+#  define flac_tell_status         FLAC__SeekableStreamDecoderTellStatus
+#  define flac_tell_status_ok      FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK
+#  define flac_tell_status_error \
+                                FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
+#  define flac_tell_status_unsupported \
+                                FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
+
+#  define flac_length_status       FLAC__SeekableStreamDecoderLengthStatus
+#  define flac_length_status_ok  FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK
+#  define flac_length_status_error \
+                              FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
+#  define flac_length_status_unsupported \
+                              FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
+
+#  ifdef HAVE_OGGFLAC
+#    include <OggFLAC/seekable_stream_decoder.h>
+#  endif
+#else /* FLAC_API_VERSION_CURRENT > 7 */
+
+/*
+ * OggFLAC support is handled by our flac_plugin already, and
+ * thus we *can* always have it if libFLAC was compiled with it
+ */
+#  include "_ogg_common.h"
+
+#  include <FLAC/stream_decoder.h>
+#  define flac_decoder           FLAC__StreamDecoder
+#  define flac_new()             FLAC__stream_decoder_new()
+
+#  define flac_init(a,b,c,d,e,f,g,h,i,j) \
+        (FLAC__stream_decoder_init_stream(a,b,c,d,e,f,g,h,i,j) \
+         == FLAC__STREAM_DECODER_INIT_STATUS_OK)
+#  define flac_ogg_init(a,b,c,d,e,f,g,h,i,j) \
+        (FLAC__stream_decoder_init_ogg_stream(a,b,c,d,e,f,g,h,i,j) \
+         == FLAC__STREAM_DECODER_INIT_STATUS_OK)
+
+#  define flac_get_decode_position(x,y) \
+                 FLAC__stream_decoder_get_decode_position(x,y)
+#  define flac_get_state(x)      FLAC__stream_decoder_get_state(x)
+#  define flac_process_single(x) FLAC__stream_decoder_process_single(x)
+#  define flac_process_metadata(x) \
+                          FLAC__stream_decoder_process_until_end_of_metadata(x)
+#  define flac_seek_absolute(x,y)  FLAC__stream_decoder_seek_absolute(x,y)
+#  define flac_finish(x)         FLAC__stream_decoder_finish(x)
+#  define flac_delete(x)         FLAC__stream_decoder_delete(x)
+
+#  define flac_decoder_eof       FLAC__STREAM_DECODER_END_OF_STREAM
+
+typedef size_t flac_read_status_size_t;
+#  define flac_read_status       FLAC__StreamDecoderReadStatus
+#  define flac_read_status_continue \
+                                 FLAC__STREAM_DECODER_READ_STATUS_CONTINUE
+#  define flac_read_status_eof   FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM
+#  define flac_read_status_abort FLAC__STREAM_DECODER_READ_STATUS_ABORT
+
+#  define flac_seek_status       FLAC__StreamDecoderSeekStatus
+#  define flac_seek_status_ok    FLAC__STREAM_DECODER_SEEK_STATUS_OK
+#  define flac_seek_status_error FLAC__STREAM_DECODER_SEEK_STATUS_ERROR
+#  define flac_seek_status_unsupported \
+                                 FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED
+
+#  define flac_tell_status         FLAC__StreamDecoderTellStatus
+#  define flac_tell_status_ok      FLAC__STREAM_DECODER_TELL_STATUS_OK
+#  define flac_tell_status_error   FLAC__STREAM_DECODER_TELL_STATUS_ERROR
+#  define flac_tell_status_unsupported \
+                                   FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED
+
+#  define flac_length_status       FLAC__StreamDecoderLengthStatus
+#  define flac_length_status_ok    FLAC__STREAM_DECODER_LENGTH_STATUS_OK
+#  define flac_length_status_error FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR
+#  define flac_length_status_unsupported \
+                                 FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED
+
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+#include <FLAC/metadata.h>
+
+#define FLAC_CHUNK_SIZE 4080
+
+typedef struct {
+	unsigned char chunk[FLAC_CHUNK_SIZE];
+	float time;
+	unsigned int bitRate;
+	struct audio_format audio_format;
+	float total_time;
+	FLAC__uint64 position;
+	struct decoder *decoder;
+	InputStream *inStream;
+	ReplayGainInfo *replayGainInfo;
+	struct tag *tag;
+} FlacData;
+
+/* initializes a given FlacData struct */
+void init_FlacData(FlacData * data, struct decoder * decoder,
+		   InputStream * inStream);
+void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
+			     FlacData * data);
+void flac_error_common_cb(const char *plugin,
+			  FLAC__StreamDecoderErrorStatus status,
+			  FlacData * data);
+
+struct tag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
+					   struct tag *tag);
+
+FLAC__StreamDecoderWriteStatus
+flac_common_write(FlacData *data, const FLAC__Frame * frame,
+		  const FLAC__int32 *const buf[]);
+
+#endif /* _FLAC_COMMON_H */
diff --git a/src/decoder/_ogg_common.c b/src/decoder/_ogg_common.c
new file mode 100644
index 000000000..841b2ad3f
--- /dev/null
+++ b/src/decoder/_ogg_common.c
@@ -0,0 +1,49 @@
+/* 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
+ *
+ * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC)
+ * (c) 2005 by Eric Wong <normalperson@yhbt.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 "_ogg_common.h"
+#include "_flac_common.h"
+#include "../utils.h"
+
+ogg_stream_type ogg_stream_type_detect(InputStream * inStream)
+{
+	/* oggflac detection based on code in ogg123 and this post
+	 * http://lists.xiph.org/pipermail/flac/2004-December/000393.html
+	 * ogg123 trunk still doesn't have this patch as of June 2005 */
+	unsigned char buf[41];
+	size_t r;
+
+	seekInputStream(inStream, 0, SEEK_SET);
+
+	r = decoder_read(NULL, inStream, buf, sizeof(buf));
+
+	if (r > 0)
+		seekInputStream(inStream, 0, SEEK_SET);
+
+	if (r >= 32 && memcmp(buf, "OggS", 4) == 0 && (
+				(memcmp(buf+29, "FLAC", 4) == 0
+					&& memcmp(buf+37, "fLaC", 4) == 0)
+				|| (memcmp(buf+28, "FLAC", 4) == 0)
+				|| (memcmp(buf+28, "fLaC", 4) == 0))) {
+		return FLAC;
+	}
+	return VORBIS;
+}
diff --git a/src/decoder/_ogg_common.h b/src/decoder/_ogg_common.h
new file mode 100644
index 000000000..7c9e7b630
--- /dev/null
+++ b/src/decoder/_ogg_common.h
@@ -0,0 +1,31 @@
+/* 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
+ *
+ * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC)
+ * (c) 2005 by Eric Wong <normalperson@yhbt.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
+ */
+
+#ifndef _OGG_COMMON_H
+#define _OGG_COMMON_H
+
+#include "../decoder_api.h"
+
+typedef enum _ogg_stream_type { VORBIS, FLAC } ogg_stream_type;
+
+ogg_stream_type ogg_stream_type_detect(InputStream * inStream);
+
+#endif /* _OGG_COMMON_H */
diff --git a/src/decoder/aac_plugin.c b/src/decoder/aac_plugin.c
new file mode 100644
index 000000000..7842bcc22
--- /dev/null
+++ b/src/decoder/aac_plugin.c
@@ -0,0 +1,602 @@
+/* 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 "../decoder_api.h"
+
+#define AAC_MAX_CHANNELS	6
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+#include <faad.h>
+
+/* all code here is either based on or copied from FAAD2's frontend code */
+typedef struct {
+	struct decoder *decoder;
+	InputStream *inStream;
+	size_t bytesIntoBuffer;
+	size_t bytesConsumed;
+	off_t fileOffset;
+	unsigned char *buffer;
+	int atEof;
+} AacBuffer;
+
+static void aac_buffer_shift(AacBuffer * b, size_t length)
+{
+	assert(length >= b->bytesConsumed);
+	assert(length <= b->bytesConsumed + b->bytesIntoBuffer);
+
+	memmove(b->buffer, b->buffer + length,
+		b->bytesConsumed + b->bytesIntoBuffer - length);
+
+	length -= b->bytesConsumed;
+	b->bytesConsumed = 0;
+	b->bytesIntoBuffer -= length;
+}
+
+static void fillAacBuffer(AacBuffer * b)
+{
+	size_t bread;
+
+	if (b->bytesIntoBuffer >= FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS)
+		/* buffer already full */
+		return;
+
+	aac_buffer_shift(b, b->bytesConsumed);
+
+	if (!b->atEof) {
+		size_t rest = FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS -
+			b->bytesIntoBuffer;
+
+		bread = decoder_read(b->decoder, b->inStream,
+				     (void *)(b->buffer + b->bytesIntoBuffer),
+				     rest);
+		if (bread == 0 && inputStreamAtEOF(b->inStream))
+			b->atEof = 1;
+		b->bytesIntoBuffer += bread;
+	}
+
+	if ((b->bytesIntoBuffer > 3 && memcmp(b->buffer, "TAG", 3) == 0) ||
+	    (b->bytesIntoBuffer > 11 &&
+	     memcmp(b->buffer, "LYRICSBEGIN", 11) == 0) ||
+	    (b->bytesIntoBuffer > 8 && memcmp(b->buffer, "APETAGEX", 8) == 0))
+		b->bytesIntoBuffer = 0;
+}
+
+static void advanceAacBuffer(AacBuffer * b, size_t bytes)
+{
+	b->fileOffset += bytes;
+	b->bytesConsumed = bytes;
+	b->bytesIntoBuffer -= bytes;
+}
+
+static int adtsSampleRates[] =
+    { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+	16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+/**
+ * Check whether the buffer head is an AAC frame, and return the frame
+ * length.  Returns 0 if it is not a frame.
+ */
+static size_t adts_check_frame(AacBuffer * b)
+{
+	if (b->bytesIntoBuffer <= 7)
+		return 0;
+
+	/* check syncword */
+	if (!((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)))
+		return 0;
+
+	return (((unsigned int)b->buffer[3] & 0x3) << 11) |
+		(((unsigned int)b->buffer[4]) << 3) |
+		(b->buffer[5] >> 5);
+}
+
+/**
+ * Find the next AAC frame in the buffer.  Returns 0 if no frame is
+ * found or if not enough data is available.
+ */
+static size_t adts_find_frame(AacBuffer * b)
+{
+	const unsigned char *p;
+	size_t frame_length;
+
+	while ((p = memchr(b->buffer, 0xff, b->bytesIntoBuffer)) != NULL) {
+		/* discard data before 0xff */
+		if (p > b->buffer)
+			aac_buffer_shift(b, p - b->buffer);
+
+		if (b->bytesIntoBuffer <= 7)
+			/* not enough data yet */
+			return 0;
+
+		/* is it a frame? */
+		frame_length = adts_check_frame(b);
+		if (frame_length > 0)
+			/* yes, it is */
+			return frame_length;
+
+		/* it's just some random 0xff byte; discard and and
+		   continue searching */
+		aac_buffer_shift(b, 1);
+	}
+
+	/* nothing at all; discard the whole buffer */
+	aac_buffer_shift(b, b->bytesIntoBuffer);
+	return 0;
+}
+
+static void adtsParse(AacBuffer * b, float *length)
+{
+	unsigned int frames, frameLength;
+	int sample_rate = 0;
+	float framesPerSec;
+
+	/* Read all frames to ensure correct time and bitrate */
+	for (frames = 0;; frames++) {
+		fillAacBuffer(b);
+
+		frameLength = adts_find_frame(b);
+		if (frameLength > 0) {
+			if (frames == 0) {
+				sample_rate = adtsSampleRates[(b->
+							       buffer[2] & 0x3c)
+							      >> 2];
+			}
+
+			if (frameLength > b->bytesIntoBuffer)
+				break;
+
+			advanceAacBuffer(b, frameLength);
+		} else
+			break;
+	}
+
+	framesPerSec = (float)sample_rate / 1024.0;
+	if (framesPerSec != 0)
+		*length = (float)frames / framesPerSec;
+}
+
+static void initAacBuffer(AacBuffer * b,
+			  struct decoder *decoder, InputStream * inStream)
+{
+	memset(b, 0, sizeof(AacBuffer));
+
+	b->decoder = decoder;
+	b->inStream = inStream;
+
+	b->buffer = xmalloc(FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+	memset(b->buffer, 0, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
+}
+
+static void aac_parse_header(AacBuffer * b, float *length)
+{
+	size_t fileread;
+	size_t tagsize;
+
+	if (length)
+		*length = -1;
+
+	fileread = b->inStream->size;
+
+	fillAacBuffer(b);
+
+	tagsize = 0;
+	if (b->bytesIntoBuffer >= 10 && !memcmp(b->buffer, "ID3", 3)) {
+		tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
+		    (b->buffer[8] << 7) | (b->buffer[9] << 0);
+
+		tagsize += 10;
+		advanceAacBuffer(b, tagsize);
+		fillAacBuffer(b);
+	}
+
+	if (length == NULL)
+		return;
+
+	if (b->bytesIntoBuffer >= 2 &&
+	    (b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
+		adtsParse(b, length);
+		seekInputStream(b->inStream, tagsize, SEEK_SET);
+
+		b->bytesIntoBuffer = 0;
+		b->bytesConsumed = 0;
+		b->fileOffset = tagsize;
+
+		fillAacBuffer(b);
+	} else if (memcmp(b->buffer, "ADIF", 4) == 0) {
+		int bitRate;
+		int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
+		bitRate =
+		    ((unsigned int)(b->
+				    buffer[4 +
+					   skipSize] & 0x0F) << 19) | ((unsigned
+									int)b->
+								       buffer[5
+									      +
+									      skipSize]
+								       << 11) |
+		    ((unsigned int)b->
+		     buffer[6 + skipSize] << 3) | ((unsigned int)b->buffer[7 +
+									   skipSize]
+						   & 0xE0);
+
+		if (fileread != 0 && bitRate != 0)
+			*length = fileread * 8.0 / bitRate;
+		else
+			*length = fileread;
+	}
+}
+
+static float getAacFloatTotalTime(char *file)
+{
+	AacBuffer b;
+	float length;
+	faacDecHandle decoder;
+	faacDecConfigurationPtr config;
+	uint32_t sample_rate;
+	unsigned char channels;
+	InputStream inStream;
+	long bread;
+
+	if (openInputStream(&inStream, file) < 0)
+		return -1;
+
+	initAacBuffer(&b, NULL, &inStream);
+	aac_parse_header(&b, &length);
+
+	if (length < 0) {
+		decoder = faacDecOpen();
+
+		config = faacDecGetCurrentConfiguration(decoder);
+		config->outputFormat = FAAD_FMT_16BIT;
+		faacDecSetConfiguration(decoder, config);
+
+		fillAacBuffer(&b);
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+		bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+				    &sample_rate, &channels);
+#else
+		bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+		if (bread >= 0 && sample_rate > 0 && channels > 0)
+			length = 0;
+
+		faacDecClose(decoder);
+	}
+
+	if (b.buffer)
+		free(b.buffer);
+	closeInputStream(&inStream);
+
+	return length;
+}
+
+static int getAacTotalTime(char *file)
+{
+	int file_time = -1;
+	float length;
+
+	if ((length = getAacFloatTotalTime(file)) >= 0)
+		file_time = length + 0.5;
+
+	return file_time;
+}
+
+static int aac_stream_decode(struct decoder * mpd_decoder,
+			     InputStream *inStream)
+{
+	float file_time;
+	float totalTime = 0;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	long bread;
+	struct audio_format audio_format;
+	uint32_t sample_rate;
+	unsigned char channels;
+	unsigned int sampleCount;
+	char *sampleBuffer;
+	size_t sampleBufferLen;
+	uint16_t bitRate = 0;
+	AacBuffer b;
+	int initialized = 0;
+
+	initAacBuffer(&b, mpd_decoder, inStream);
+
+	decoder = faacDecOpen();
+
+	config = faacDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+	config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+	config->dontUpSampleImplicitSBR = 0;
+#endif
+	faacDecSetConfiguration(decoder, config);
+
+	while (b.bytesIntoBuffer < FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS &&
+	       !b.atEof &&
+	       decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+	       	fillAacBuffer(&b);
+		adts_find_frame(&b);
+		fillAacBuffer(&b);
+		my_usleep(10000);
+	}
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+	bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+			    &sample_rate, &channels);
+#else
+	bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+	if (bread < 0) {
+		ERROR("Error not a AAC stream.\n");
+		faacDecClose(decoder);
+		if (b.buffer)
+			free(b.buffer);
+		return -1;
+	}
+
+	audio_format.bits = 16;
+
+	file_time = 0.0;
+
+	advanceAacBuffer(&b, bread);
+
+	while (1) {
+		fillAacBuffer(&b);
+		adts_find_frame(&b);
+		fillAacBuffer(&b);
+
+		if (b.bytesIntoBuffer == 0)
+			break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+		sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
+					     b.bytesIntoBuffer);
+#else
+		sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
+#endif
+
+		if (frameInfo.error > 0) {
+			ERROR("error decoding AAC stream\n");
+			ERROR("faad2 error: %s\n",
+			      faacDecGetErrorMessage(frameInfo.error));
+			break;
+		}
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+		sample_rate = frameInfo.samplerate;
+#endif
+
+		if (!initialized) {
+			audio_format.channels = frameInfo.channels;
+			audio_format.sample_rate = sample_rate;
+			decoder_initialized(mpd_decoder, &audio_format, totalTime);
+			initialized = 1;
+		}
+
+		advanceAacBuffer(&b, frameInfo.bytesconsumed);
+
+		sampleCount = (unsigned long)(frameInfo.samples);
+
+		if (sampleCount > 0) {
+			bitRate = frameInfo.bytesconsumed * 8.0 *
+			    frameInfo.channels * sample_rate /
+			    frameInfo.samples / 1000 + 0.5;
+			file_time +=
+			    (float)(frameInfo.samples) / frameInfo.channels /
+			    sample_rate;
+		}
+
+		sampleBufferLen = sampleCount * 2;
+
+		decoder_data(mpd_decoder, NULL, 0, sampleBuffer,
+			     sampleBufferLen, file_time,
+			     bitRate, NULL);
+		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+			decoder_seek_error(mpd_decoder);
+		} else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+			break;
+	}
+
+	decoder_flush(mpd_decoder);
+
+	faacDecClose(decoder);
+	if (b.buffer)
+		free(b.buffer);
+
+	if (!initialized)
+		return -1;
+
+	if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+		decoder_seek_error(mpd_decoder);
+	}
+
+	return 0;
+}
+
+
+static int aac_decode(struct decoder * mpd_decoder, char *path)
+{
+	float file_time;
+	float totalTime;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	long bread;
+	struct audio_format audio_format;
+	uint32_t sample_rate;
+	unsigned char channels;
+	unsigned int sampleCount;
+	char *sampleBuffer;
+	size_t sampleBufferLen;
+	/*float * seekTable;
+	   long seekTableEnd = -1;
+	   int seekPositionFound = 0; */
+	uint16_t bitRate = 0;
+	AacBuffer b;
+	InputStream inStream;
+	int initialized = 0;
+
+	if ((totalTime = getAacFloatTotalTime(path)) < 0)
+		return -1;
+
+	if (openInputStream(&inStream, path) < 0)
+		return -1;
+
+	initAacBuffer(&b, mpd_decoder, &inStream);
+	aac_parse_header(&b, NULL);
+
+	decoder = faacDecOpen();
+
+	config = faacDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+	config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+	config->dontUpSampleImplicitSBR = 0;
+#endif
+	faacDecSetConfiguration(decoder, config);
+
+	fillAacBuffer(&b);
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+	bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+			    &sample_rate, &channels);
+#else
+	bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
+#endif
+	if (bread < 0) {
+		ERROR("Error not a AAC stream.\n");
+		faacDecClose(decoder);
+		if (b.buffer)
+			free(b.buffer);
+		return -1;
+	}
+
+	audio_format.bits = 16;
+
+	file_time = 0.0;
+
+	advanceAacBuffer(&b, bread);
+
+	while (1) {
+		fillAacBuffer(&b);
+
+		if (b.bytesIntoBuffer == 0)
+			break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+		sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
+					     b.bytesIntoBuffer);
+#else
+		sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
+#endif
+
+		if (frameInfo.error > 0) {
+			ERROR("error decoding AAC file: %s\n", path);
+			ERROR("faad2 error: %s\n",
+			      faacDecGetErrorMessage(frameInfo.error));
+			break;
+		}
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+		sample_rate = frameInfo.samplerate;
+#endif
+
+		if (!initialized) {
+			audio_format.channels = frameInfo.channels;
+			audio_format.sample_rate = sample_rate;
+			decoder_initialized(mpd_decoder, &audio_format,
+					    totalTime);
+			initialized = 1;
+		}
+
+		advanceAacBuffer(&b, frameInfo.bytesconsumed);
+
+		sampleCount = (unsigned long)(frameInfo.samples);
+
+		if (sampleCount > 0) {
+			bitRate = frameInfo.bytesconsumed * 8.0 *
+			    frameInfo.channels * sample_rate /
+			    frameInfo.samples / 1000 + 0.5;
+			file_time +=
+			    (float)(frameInfo.samples) / frameInfo.channels /
+			    sample_rate;
+		}
+
+		sampleBufferLen = sampleCount * 2;
+
+		decoder_data(mpd_decoder, NULL, 0, sampleBuffer,
+			     sampleBufferLen, file_time,
+			     bitRate, NULL);
+		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+			decoder_seek_error(mpd_decoder);
+		} else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+			break;
+	}
+
+	decoder_flush(mpd_decoder);
+
+	faacDecClose(decoder);
+	if (b.buffer)
+		free(b.buffer);
+
+	if (!initialized)
+		return -1;
+
+	if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+		decoder_seek_error(mpd_decoder);
+	}
+
+	return 0;
+}
+
+static struct tag *aacTagDup(char *file)
+{
+	struct tag *ret = NULL;
+	int file_time = getAacTotalTime(file);
+
+	if (file_time >= 0) {
+		if ((ret = tag_id3_load(file)) == NULL)
+			ret = tag_new();
+		ret->time = file_time;
+	} else {
+		DEBUG("aacTagDup: Failed to get total song time from: %s\n",
+		      file);
+	}
+
+	return ret;
+}
+
+static const char *aac_suffixes[] = { "aac", NULL };
+static const char *aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL };
+
+struct decoder_plugin aacPlugin = {
+	.name = "aac",
+	.stream_decode = aac_stream_decode,
+	.file_decode = aac_decode,
+	.tag_dup = aacTagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+	.suffixes = aac_suffixes,
+	.mime_types = aac_mimeTypes
+};
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
new file mode 100644
index 000000000..99846e853
--- /dev/null
+++ b/src/decoder/audiofile_plugin.c
@@ -0,0 +1,147 @@
+/* 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
+ * 
+ * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.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 "../decoder_api.h"
+#include "../log.h"
+
+#include <sys/stat.h>
+#include <audiofile.h>
+
+/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
+#define CHUNK_SIZE		1020
+
+static int getAudiofileTotalTime(char *file)
+{
+	int total_time;
+	AFfilehandle af_fp = afOpenFile(file, "r", NULL);
+	if (af_fp == AF_NULL_FILEHANDLE) {
+		return -1;
+	}
+	total_time = (int)
+	    ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK)
+	     / afGetRate(af_fp, AF_DEFAULT_TRACK));
+	afCloseFile(af_fp);
+	return total_time;
+}
+
+static int audiofile_decode(struct decoder * decoder, char *path)
+{
+	int fs, frame_count;
+	AFfilehandle af_fp;
+	int bits;
+	struct audio_format audio_format;
+	float total_time;
+	uint16_t bitRate;
+	struct stat st;
+	int ret, current = 0;
+	char chunk[CHUNK_SIZE];
+
+	if (stat(path, &st) < 0) {
+		ERROR("failed to stat: %s\n", path);
+		return -1;
+	}
+
+	af_fp = afOpenFile(path, "r", NULL);
+	if (af_fp == AF_NULL_FILEHANDLE) {
+		ERROR("failed to open: %s\n", path);
+		return -1;
+	}
+
+	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
+	                         AF_SAMPFMT_TWOSCOMP, 16);
+	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
+	audio_format.bits = (uint8_t)bits;
+	audio_format.sample_rate =
+	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
+	audio_format.channels =
+	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+
+	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
+
+	total_time = ((float)frame_count / (float)audio_format.sample_rate);
+
+	bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
+
+	if (audio_format.bits != 8 && audio_format.bits != 16) {
+		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
+		      path, audio_format.bits);
+		afCloseFile(af_fp);
+		return -1;
+	}
+
+	fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
+
+	decoder_initialized(decoder, &audio_format, total_time);
+
+	do {
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+			decoder_clear(decoder);
+			current = decoder_seek_where(decoder) *
+				audio_format.sample_rate;
+			afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
+			decoder_command_finished(decoder);
+		}
+
+		ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
+				   CHUNK_SIZE / fs);
+		if (ret <= 0)
+			break;
+
+		current += ret;
+		decoder_data(decoder, NULL, 1,
+			     chunk, ret * fs,
+			     (float)current / (float)audio_format.sample_rate,
+			     bitRate, NULL);
+	} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
+
+	decoder_flush(decoder);
+
+	afCloseFile(af_fp);
+
+	return 0;
+}
+
+static struct tag *audiofileTagDup(char *file)
+{
+	struct tag *ret = NULL;
+	int total_time = getAudiofileTotalTime(file);
+
+	if (total_time >= 0) {
+		if (!ret)
+			ret = tag_new();
+		ret->time = total_time;
+	} else {
+		DEBUG
+		    ("audiofileTagDup: Failed to get total song time from: %s\n",
+		     file);
+	}
+
+	return ret;
+}
+
+static const char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL };
+
+struct decoder_plugin audiofilePlugin = {
+	.name = "audiofile",
+	.file_decode = audiofile_decode,
+	.tag_dup = audiofileTagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = audiofileSuffixes,
+};
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
new file mode 100644
index 000000000..6455cd1ce
--- /dev/null
+++ b/src/decoder/ffmpeg_plugin.c
@@ -0,0 +1,419 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2008 Viliam Mateicka <viliam.mateicka@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 "../decoder_api.h"
+#include "../log.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h>
+
+#ifdef OLD_FFMPEG_INCLUDES
+#include <avcodec.h>
+#include <avformat.h>
+#include <avio.h>
+#else
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#endif
+
+typedef struct {
+	int audioStream;
+	AVFormatContext *pFormatCtx;
+	AVCodecContext *aCodecCtx;
+	AVCodec *aCodec;
+	struct decoder *decoder;
+	InputStream *input;
+	struct tag *tag;
+} BasePtrs;
+
+typedef struct {
+	/** hack - see url_to_base() */
+	char url[8];
+
+	struct decoder *decoder;
+	InputStream *input;
+} FopsHelper;
+
+/**
+ * Convert a faked mpd:// URL to a FopsHelper structure.  This is a
+ * hack because ffmpeg does not provide a nice API for passing a
+ * user-defined pointer to mpdurl_open().
+ */
+static FopsHelper *url_to_base(const char *url)
+{
+	union {
+		const char *in;
+		FopsHelper *out;
+	} u = { .in = url };
+	return u.out;
+}
+
+static int mpdurl_open(URLContext *h, const char *filename,
+		       mpd_unused int flags)
+{
+	FopsHelper *base = url_to_base(filename);
+	h->priv_data = base;
+	h->is_streamed = (base->input->seekable ? 0 : 1);
+	return 0;
+}
+
+static int mpdurl_read(URLContext *h, unsigned char *buf, int size)
+{
+	int ret;
+	FopsHelper *base = (FopsHelper *) h->priv_data;
+	while (1) {
+		ret = readFromInputStream(base->input, (void *)buf, size);
+		if (ret == 0) {
+			DEBUG("ret 0\n");
+			if (inputStreamAtEOF(base->input) ||
+			    (base->decoder &&
+			     decoder_get_command(base->decoder) != DECODE_COMMAND_NONE)) {
+				DEBUG("eof stream\n");
+				return ret;
+			} else {
+				my_usleep(10000);
+			}
+		} else {
+			break;
+		}
+	}
+	return ret;
+}
+
+static int64_t mpdurl_seek(URLContext *h, int64_t pos, int whence)
+{
+	FopsHelper *base = (FopsHelper *) h->priv_data;
+	if (whence != AVSEEK_SIZE) { //only ftell
+		(void) seekInputStream(base->input, pos, whence);
+	}
+	return base->input->offset;
+}
+
+static int mpdurl_close(URLContext *h)
+{
+	FopsHelper *base = (FopsHelper *) h->priv_data;
+	if (base && base->input->seekable) {
+		(void) seekInputStream(base->input, 0, SEEK_SET);
+	}
+	h->priv_data = 0;
+	return 0;
+}
+
+static URLProtocol mpdurl_fileops = {
+	.name = "mpd",
+	.url_open = mpdurl_open,
+	.url_read = mpdurl_read,
+	.url_seek = mpdurl_seek,
+	.url_close = mpdurl_close,
+};
+
+static int ffmpeg_init(void)
+{
+	av_register_all();
+	register_protocol(&mpdurl_fileops);
+	return 0;
+}
+
+static int ffmpeg_helper(InputStream *input, int (*callback)(BasePtrs *ptrs),
+			 BasePtrs *ptrs)
+{
+	AVFormatContext *pFormatCtx;
+	AVCodecContext *aCodecCtx;
+	AVCodec *aCodec;
+	int ret, audioStream;
+	unsigned i;
+	FopsHelper fopshelp = {
+		.url = "mpd://X", /* only the mpd:// prefix matters */
+	};
+
+	fopshelp.input = input;
+	if (ptrs && ptrs->decoder) {
+		fopshelp.decoder = ptrs->decoder; //are we in decoding loop ?
+	} else {
+		fopshelp.decoder = NULL;
+	}
+
+	//ffmpeg works with ours "fileops" helper
+	if (av_open_input_file(&pFormatCtx, fopshelp.url, NULL, 0, NULL)!=0) {
+		ERROR("Open failed!\n");
+		return -1;
+	}
+
+	if (av_find_stream_info(pFormatCtx)<0) {
+		ERROR("Couldn't find stream info!\n");
+		return -1;
+	}
+
+	audioStream = -1;
+	for(i=0; i<pFormatCtx->nb_streams; i++) {
+		if (pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO &&
+		    audioStream < 0) {
+			audioStream=i;
+		}
+	}
+
+	if(audioStream==-1) {
+		ERROR("No audio stream inside!\n");
+		return -1;
+	}
+
+	aCodecCtx = pFormatCtx->streams[audioStream]->codec;
+	aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
+
+	if (!aCodec) {
+		ERROR("Unsupported audio codec!\n");
+		return -1;
+	}
+
+	if (avcodec_open(aCodecCtx, aCodec)<0) {
+		ERROR("Could not open codec!\n");
+		return -1;
+	}
+
+	if (callback) {
+		ptrs->audioStream = audioStream;
+		ptrs->pFormatCtx = pFormatCtx;
+		ptrs->aCodecCtx = aCodecCtx;
+		ptrs->aCodec = aCodec;
+
+		ret = (*callback)( ptrs );
+	} else {
+		ret = 0;
+		DEBUG("playable\n");
+	}
+
+	avcodec_close(aCodecCtx);
+	av_close_input_file(pFormatCtx);
+
+	return ret;
+}
+
+static bool ffmpeg_try_decode(InputStream *input)
+{
+	int ret;
+	if (input->seekable) {
+		ret = ffmpeg_helper(input, NULL, NULL);
+	} else {
+		ret = 0;
+	}
+	return (ret == -1 ? 0 : 1);
+}
+
+static int ffmpeg_decode_internal(BasePtrs *base)
+{
+	struct decoder *decoder = base->decoder;
+	AVCodecContext *aCodecCtx = base->aCodecCtx;
+	AVFormatContext *pFormatCtx = base->pFormatCtx;
+	AVPacket packet;
+	int len, audio_size;
+	int position;
+	struct audio_format audio_format;
+	int current, total_time;
+	uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
+
+	total_time = 0;
+
+	DEBUG("decoder_start\n");
+
+	if (aCodecCtx->channels > 2) {
+		aCodecCtx->channels = 2;
+	}
+
+	audio_format.bits = (uint8_t)16;
+	audio_format.sample_rate = (unsigned int)aCodecCtx->sample_rate;
+	audio_format.channels = aCodecCtx->channels;
+
+	//    frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
+	//    total_time = ((float)frame_count / (float)audio_format.sample_rate);
+
+	//there is some problem with this on some demux (mp3 at least)
+	if (pFormatCtx->duration != (int)AV_NOPTS_VALUE) {
+		total_time = pFormatCtx->duration / AV_TIME_BASE;
+	}
+
+	DEBUG("ffmpeg sample rate: %dHz %d channels\n",
+	      aCodecCtx->sample_rate, aCodecCtx->channels);
+
+	decoder_initialized(decoder, &audio_format, total_time);
+
+	position = 0;
+
+	DEBUG("duration:%d (%d secs)\n", (int) pFormatCtx->duration,
+	      (int) total_time);
+
+	do {
+
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+
+			DEBUG("seek\n");
+			decoder_clear(decoder);
+			current = decoder_seek_where(decoder) * AV_TIME_BASE;
+
+			if (av_seek_frame(pFormatCtx, -1, current , 0) < 0) {
+				WARNING("seek to %d failed\n", current);
+			}
+
+			decoder_command_finished(decoder);
+		}
+
+		if (av_read_frame(pFormatCtx, &packet) >= 0) {
+			if(packet.stream_index == base->audioStream) {
+
+				position = av_rescale_q(packet.pts, pFormatCtx->streams[base->audioStream]->time_base,
+							(AVRational){1, 1});
+
+				audio_size = sizeof(audio_buf);
+				len = avcodec_decode_audio2(aCodecCtx,
+							    (int16_t *)audio_buf,
+							    &audio_size,
+							    packet.data,
+							    packet.size);
+
+				if(len >= 0) {
+					if(audio_size >= 0) {
+						// DEBUG("sending data %d/%d\n", audio_size, len);
+
+						decoder_data(decoder, NULL, 1,
+							     audio_buf, audio_size,
+							     position, //(float)current / (float)audio_format.sample_rate,
+							     aCodecCtx->bit_rate / 1000, NULL);
+
+					}
+				} else {
+					WARNING("skiping frame!\n");
+				}
+			}
+			av_free_packet(&packet);
+		} else {
+			//end of file
+			break;
+		}
+	} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
+
+	decoder_flush(decoder);
+
+	DEBUG("decoder finish\n");
+
+	return 0;
+}
+
+static int ffmpeg_decode(struct decoder *decoder, InputStream *input)
+{
+	BasePtrs base;
+	int ret;
+
+	DEBUG("decode start\n");
+
+	base.input = input;
+	base.decoder = decoder;
+
+	ret = ffmpeg_helper(input, ffmpeg_decode_internal, &base);
+
+	DEBUG("decode finish\n");
+
+	return ret;
+}
+
+static int ffmpeg_tag_internal(BasePtrs *base)
+{
+	struct tag *tag = (struct tag *) base->tag;
+
+	if (base->pFormatCtx->duration != (int)AV_NOPTS_VALUE) {
+		tag->time = base->pFormatCtx->duration / AV_TIME_BASE;
+	} else {
+		tag->time = 0;
+	}
+	return 0;
+}
+
+//no tag reading in ffmpeg, check if playable
+static struct tag *ffmpeg_tag(char *file)
+{
+	InputStream input;
+	BasePtrs base;
+	int ret;
+	struct tag *tag = NULL;
+
+	if (openInputStream(&input, file) < 0) {
+		ERROR("failed to open %s\n", file);
+		return NULL;
+	}
+
+	tag = tag_new();
+
+	base.tag = tag;
+	ret = ffmpeg_helper(&input, ffmpeg_tag_internal, &base);
+
+	if (ret != 0) {
+		free(tag);
+		tag = NULL;
+	}
+
+	closeInputStream(&input);
+
+	return tag;
+}
+
+/**
+ * ffmpeg can decode almost everything from open codecs
+ * and also some of propietary codecs
+ * its hard to tell what can ffmpeg decode
+ * we can later put this into configure script
+ * to be sure ffmpeg is used to handle
+ * only that files
+ */
+
+static const char *ffmpeg_Suffixes[] = {
+	"wma", "asf", "wmv", "mpeg", "mpg", "avi", "vob", "mov", "qt", "swf", "rm", "swf",
+	"mp1", "mp2", "mp3", "mp4", "m4a", "flac", "ogg", "wav", "au", "aiff", "aif", "ac3", "aac", "mpc",
+	NULL
+};
+
+//not sure if this is correct...
+static const char *ffmpeg_Mimetypes[] = {
+	"video/x-ms-asf",
+	"audio/x-ms-wma",
+	"audio/x-ms-wax",
+	"video/x-ms-wmv",
+	"video/x-ms-wvx",
+	"video/x-ms-wm",
+	"video/x-ms-wmx",
+	"application/x-ms-wmz",
+	"application/x-ms-wmd",
+	"audio/mpeg",
+	NULL
+};
+
+struct decoder_plugin ffmpegPlugin = {
+	.name = "ffmpeg",
+	.init = ffmpeg_init,
+	.try_decode = ffmpeg_try_decode,
+	.stream_decode = ffmpeg_decode,
+	.tag_dup = ffmpeg_tag,
+	.stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = ffmpeg_Suffixes,
+	.mime_types = ffmpeg_Mimetypes
+};
diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c
new file mode 100644
index 000000000..7b9fce27d
--- /dev/null
+++ b/src/decoder/flac_plugin.c
@@ -0,0 +1,459 @@
+/* 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 "_flac_common.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+
+/* this code was based on flac123, from flac-tools */
+
+static flac_read_status flacRead(mpd_unused const flac_decoder * flacDec,
+                                  FLAC__byte buf[],
+				  flac_read_status_size_t *bytes,
+				  void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+	size_t r;
+
+	r = decoder_read(data->decoder, data->inStream, (void *)buf, *bytes);
+	*bytes = r;
+
+	if (r == 0) {
+		if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE ||
+		    inputStreamAtEOF(data->inStream))
+			return flac_read_status_eof;
+		else
+			return flac_read_status_abort;
+	}
+
+	return flac_read_status_continue;
+}
+
+static flac_seek_status flacSeek(mpd_unused const flac_decoder * flacDec,
+				 FLAC__uint64 offset,
+				 void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
+		return flac_seek_status_error;
+	}
+
+	return flac_seek_status_ok;
+}
+
+static flac_tell_status flacTell(mpd_unused const flac_decoder * flacDec,
+				 FLAC__uint64 * offset,
+				 void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	*offset = (long)(data->inStream->offset);
+
+	return flac_tell_status_ok;
+}
+
+static flac_length_status flacLength(mpd_unused const flac_decoder * flacDec,
+				     FLAC__uint64 * length,
+				     void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	*length = (size_t) (data->inStream->size);
+
+	return flac_length_status_ok;
+}
+
+static FLAC__bool flacEOF(mpd_unused const flac_decoder * flacDec, void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE &&
+		decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) ||
+		inputStreamAtEOF(data->inStream);
+}
+
+static void flacError(mpd_unused const flac_decoder *dec,
+		      FLAC__StreamDecoderErrorStatus status, void *fdata)
+{
+	flac_error_common_cb("flac", status, (FlacData *) fdata);
+}
+
+#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+static void flacPrintErroredState(FLAC__SeekableStreamDecoderState state)
+{
+	const char *str = ""; /* "" to silence compiler warning */
+	switch (state) {
+	case FLAC__SEEKABLE_STREAM_DECODER_OK:
+	case FLAC__SEEKABLE_STREAM_DECODER_SEEKING:
+	case FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM:
+		return;
+	case FLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
+		str = "allocation error";
+		break;
+	case FLAC__SEEKABLE_STREAM_DECODER_READ_ERROR:
+		str = "read error";
+		break;
+	case FLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR:
+		str = "seek error";
+		break;
+	case FLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR:
+		str = "seekable stream error";
+		break;
+	case FLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED:
+		str = "decoder already initialized";
+		break;
+	case FLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK:
+		str = "invalid callback";
+		break;
+	case FLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED:
+		str = "decoder uninitialized";
+	}
+	ERROR("flac %s\n", str);
+}
+
+static int flac_init(FLAC__SeekableStreamDecoder *dec,
+                     FLAC__SeekableStreamDecoderReadCallback read_cb,
+                     FLAC__SeekableStreamDecoderSeekCallback seek_cb,
+                     FLAC__SeekableStreamDecoderTellCallback tell_cb,
+                     FLAC__SeekableStreamDecoderLengthCallback length_cb,
+                     FLAC__SeekableStreamDecoderEofCallback eof_cb,
+                     FLAC__SeekableStreamDecoderWriteCallback write_cb,
+                     FLAC__SeekableStreamDecoderMetadataCallback metadata_cb,
+                     FLAC__SeekableStreamDecoderErrorCallback error_cb,
+                     void *data)
+{
+	int s = 1;
+	s &= FLAC__seekable_stream_decoder_set_read_callback(dec, read_cb);
+	s &= FLAC__seekable_stream_decoder_set_seek_callback(dec, seek_cb);
+	s &= FLAC__seekable_stream_decoder_set_tell_callback(dec, tell_cb);
+	s &= FLAC__seekable_stream_decoder_set_length_callback(dec, length_cb);
+	s &= FLAC__seekable_stream_decoder_set_eof_callback(dec, eof_cb);
+	s &= FLAC__seekable_stream_decoder_set_write_callback(dec, write_cb);
+	s &= FLAC__seekable_stream_decoder_set_metadata_callback(dec,
+	                                                         metadata_cb);
+	s &= FLAC__seekable_stream_decoder_set_metadata_respond(dec,
+	                                  FLAC__METADATA_TYPE_VORBIS_COMMENT);
+	s &= FLAC__seekable_stream_decoder_set_error_callback(dec, error_cb);
+	s &= FLAC__seekable_stream_decoder_set_client_data(dec, data);
+	if (!s || (FLAC__seekable_stream_decoder_init(dec) !=
+	           FLAC__SEEKABLE_STREAM_DECODER_OK))
+		return 0;
+	return 1;
+}
+#else /* FLAC_API_VERSION_CURRENT >= 7 */
+static void flacPrintErroredState(FLAC__StreamDecoderState state)
+{
+	const char *str = ""; /* "" to silence compiler warning */
+	switch (state) {
+	case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA:
+	case FLAC__STREAM_DECODER_READ_METADATA:
+	case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC:
+	case FLAC__STREAM_DECODER_READ_FRAME:
+	case FLAC__STREAM_DECODER_END_OF_STREAM:
+		return;
+	case FLAC__STREAM_DECODER_OGG_ERROR:
+		str = "error in the Ogg layer";
+		break;
+	case FLAC__STREAM_DECODER_SEEK_ERROR:
+		str = "seek error";
+		break;
+	case FLAC__STREAM_DECODER_ABORTED:
+		str = "decoder aborted by read";
+		break;
+	case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
+		str = "allocation error";
+		break;
+	case FLAC__STREAM_DECODER_UNINITIALIZED:
+		str = "decoder uninitialized";
+	}
+	ERROR("flac %s\n", str);
+}
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+static void flacMetadata(mpd_unused const flac_decoder * dec,
+			 const FLAC__StreamMetadata * block, void *vdata)
+{
+	flac_metadata_common_cb(block, (FlacData *) vdata);
+}
+
+static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
+                                                const FLAC__Frame * frame,
+						const FLAC__int32 * const buf[],
+						void *vdata)
+{
+	FLAC__uint32 samples = frame->header.blocksize;
+	FlacData *data = (FlacData *) vdata;
+	float timeChange;
+	FLAC__uint64 newPosition = 0;
+
+	timeChange = ((float)samples) / frame->header.sample_rate;
+	data->time += timeChange;
+
+	flac_get_decode_position(dec, &newPosition);
+	if (data->position && newPosition >= data->position) {
+		assert(timeChange >= 0);
+
+		data->bitRate =
+		    ((newPosition - data->position) * 8.0 / timeChange)
+		    / 1000 + 0.5;
+	}
+	data->position = newPosition;
+
+	return flac_common_write(data, frame, buf);
+}
+
+static struct tag *flacMetadataDup(char *file, int *vorbisCommentFound)
+{
+	struct tag *ret = NULL;
+	FLAC__Metadata_SimpleIterator *it;
+	FLAC__StreamMetadata *block = NULL;
+
+	*vorbisCommentFound = 0;
+
+	it = FLAC__metadata_simple_iterator_new();
+	if (!FLAC__metadata_simple_iterator_init(it, file, 1, 0)) {
+		const char *err;
+		FLAC_API FLAC__Metadata_SimpleIteratorStatus s;
+
+		s = FLAC__metadata_simple_iterator_status(it);
+
+		switch (s) { /* slightly more human-friendly messages: */
+		case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ILLEGAL_INPUT:
+			err = "illegal input";
+			break;
+		case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ERROR_OPENING_FILE:
+			err = "error opening file";
+			break;
+		case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_NOT_A_FLAC_FILE:
+			err = "not a FLAC file";
+			break;
+		default:
+			err = FLAC__Metadata_SimpleIteratorStatusString[s];
+		}
+		DEBUG("flacMetadataDup: Reading '%s' "
+		      "metadata gave the following error: %s\n",
+		      file, err);
+		FLAC__metadata_simple_iterator_delete(it);
+		return ret;
+	}
+
+	do {
+		block = FLAC__metadata_simple_iterator_get_block(it);
+		if (!block)
+			break;
+		if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
+			ret = copyVorbisCommentBlockToMpdTag(block, ret);
+
+			if (ret)
+				*vorbisCommentFound = 1;
+		} else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
+			if (!ret)
+				ret = tag_new();
+			ret->time = ((float)block->data.stream_info.
+				     total_samples) /
+			    block->data.stream_info.sample_rate + 0.5;
+		}
+		FLAC__metadata_object_delete(block);
+	} while (FLAC__metadata_simple_iterator_next(it));
+
+	FLAC__metadata_simple_iterator_delete(it);
+	return ret;
+}
+
+static struct tag *flacTagDup(char *file)
+{
+	struct tag *ret = NULL;
+	int foundVorbisComment = 0;
+
+	ret = flacMetadataDup(file, &foundVorbisComment);
+	if (!ret) {
+		DEBUG("flacTagDup: Failed to grab information from: %s\n",
+		      file);
+		return NULL;
+	}
+	if (!foundVorbisComment) {
+		struct tag *temp = tag_id3_load(file);
+		if (temp) {
+			temp->time = ret->time;
+			tag_free(ret);
+			ret = temp;
+		}
+	}
+
+	return ret;
+}
+
+static int flac_decode_internal(struct decoder * decoder,
+				InputStream * inStream, int is_ogg)
+{
+	flac_decoder *flacDec;
+	FlacData data;
+	const char *err = NULL;
+
+	if (!(flacDec = flac_new()))
+		return -1;
+	init_FlacData(&data, decoder, inStream);
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+        if(!FLAC__stream_decoder_set_metadata_respond(flacDec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
+        {
+                DEBUG(__FILE__": Failed to set metadata respond\n");
+        }
+#endif
+
+
+	if (is_ogg) {
+		if (!flac_ogg_init(flacDec, flacRead, flacSeek, flacTell,
+		                   flacLength, flacEOF, flacWrite, flacMetadata,
+			           flacError, (void *)&data)) {
+			err = "doing Ogg init()";
+			goto fail;
+		}
+	} else {
+		if (!flac_init(flacDec, flacRead, flacSeek, flacTell,
+		               flacLength, flacEOF, flacWrite, flacMetadata,
+			       flacError, (void *)&data)) {
+			err = "doing init()";
+			goto fail;
+		}
+		if (!flac_process_metadata(flacDec)) {
+			err = "problem reading metadata";
+			goto fail;
+		}
+	}
+
+	decoder_initialized(decoder, &data.audio_format, data.total_time);
+
+	while (1) {
+		if (!flac_process_single(flacDec))
+			break;
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+			FLAC__uint64 sampleToSeek = decoder_seek_where(decoder) *
+			    data.audio_format.sample_rate + 0.5;
+			if (flac_seek_absolute(flacDec, sampleToSeek)) {
+				decoder_clear(decoder);
+				data.time = ((float)sampleToSeek) /
+				    data.audio_format.sample_rate;
+				data.position = 0;
+				decoder_command_finished(decoder);
+			} else
+				decoder_seek_error(decoder);
+		} else if (flac_get_state(flacDec) == flac_decoder_eof)
+			break;
+	}
+	if (decoder_get_command(decoder) != DECODE_COMMAND_STOP) {
+		flacPrintErroredState(flac_get_state(flacDec));
+		flac_finish(flacDec);
+	}
+
+fail:
+	if (data.replayGainInfo)
+		freeReplayGainInfo(data.replayGainInfo);
+
+	if (flacDec)
+		flac_delete(flacDec);
+
+	if (err) {
+		ERROR("flac %s\n", err);
+		return -1;
+	}
+	return 0;
+}
+
+static int flac_decode(struct decoder * decoder, InputStream * inStream)
+{
+	return flac_decode_internal(decoder, inStream, 0);
+}
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 && \
+	!defined(HAVE_OGGFLAC)
+static struct tag *oggflac_tag_dup(char *file)
+{
+	struct tag *ret = NULL;
+	FLAC__Metadata_Iterator *it;
+	FLAC__StreamMetadata *block;
+	FLAC__Metadata_Chain *chain = FLAC__metadata_chain_new();
+
+	if (!(FLAC__metadata_chain_read_ogg(chain, file)))
+		goto out;
+	it = FLAC__metadata_iterator_new();
+	FLAC__metadata_iterator_init(it, chain);
+	do {
+		if (!(block = FLAC__metadata_iterator_get_block(it)))
+			break;
+		if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
+			ret = copyVorbisCommentBlockToMpdTag(block, ret);
+		} else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
+			if (!ret)
+				ret = tag_new();
+			ret->time = ((float)block->data.stream_info.
+				     total_samples) /
+			    block->data.stream_info.sample_rate + 0.5;
+		}
+	} while (FLAC__metadata_iterator_next(it));
+	FLAC__metadata_iterator_delete(it);
+out:
+	FLAC__metadata_chain_delete(chain);
+	return ret;
+}
+
+static int oggflac_decode(struct decoder *decoder, InputStream * inStream)
+{
+	return flac_decode_internal(decoder, inStream, 1);
+}
+
+static bool oggflac_try_decode(InputStream * inStream)
+{
+	return FLAC_API_SUPPORTS_OGG_FLAC &&
+		ogg_stream_type_detect(inStream) == FLAC;
+}
+
+static const char *oggflac_suffixes[] = { "ogg", "oga", NULL };
+static const char *oggflac_mime_types[] = { "audio/x-flac+ogg",
+					    "application/ogg",
+					    "application/x-ogg",
+					    NULL };
+
+struct decoder_plugin oggflacPlugin = {
+	.name = "oggflac",
+	.try_decode = oggflac_try_decode,
+	.stream_decode = oggflac_decode,
+	.tag_dup = oggflac_tag_dup,
+	.stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = oggflac_suffixes,
+	.mime_types = oggflac_mime_types
+};
+
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+static const char *flacSuffixes[] = { "flac", NULL };
+static const char *flac_mime_types[] = { "audio/x-flac",
+					 "application/x-flac",
+					 NULL };
+
+struct decoder_plugin flacPlugin = {
+	.name = "flac",
+	.stream_decode = flac_decode,
+	.tag_dup = flacTagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = flacSuffixes,
+	.mime_types = flac_mime_types
+};
diff --git a/src/decoder/mod_plugin.c b/src/decoder/mod_plugin.c
new file mode 100644
index 000000000..5916a24ab
--- /dev/null
+++ b/src/decoder/mod_plugin.c
@@ -0,0 +1,278 @@
+/* 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 "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <mikmod.h>
+
+/* this is largely copied from alsaplayer */
+
+#define MIKMOD_FRAME_SIZE	4096
+
+static BOOL mod_mpd_Init(void)
+{
+	return VC_Init();
+}
+
+static void mod_mpd_Exit(void)
+{
+	VC_Exit();
+}
+
+static void mod_mpd_Update(void)
+{
+}
+
+static BOOL mod_mpd_IsThere(void)
+{
+	return 1;
+}
+
+static char drv_name[] = "MPD";
+static char drv_version[] = "MPD Output Driver v0.1";
+
+#if (LIBMIKMOD_VERSION > 0x030106)
+static char drv_alias[] = "mpd";
+#endif
+
+static MDRIVER drv_mpd = {
+	NULL,
+	drv_name,
+	drv_version,
+	0,
+	255,
+#if (LIBMIKMOD_VERSION > 0x030106)
+	drv_alias,
+#if (LIBMIKMOD_VERSION >= 0x030200)
+	NULL,  /* CmdLineHelp */
+#endif
+	NULL,  /* CommandLine */
+#endif
+	mod_mpd_IsThere,
+	VC_SampleLoad,
+	VC_SampleUnload,
+	VC_SampleSpace,
+	VC_SampleLength,
+	mod_mpd_Init,
+	mod_mpd_Exit,
+	NULL,
+	VC_SetNumVoices,
+	VC_PlayStart,
+	VC_PlayStop,
+	mod_mpd_Update,
+	NULL,
+	VC_VoiceSetVolume,
+	VC_VoiceGetVolume,
+	VC_VoiceSetFrequency,
+	VC_VoiceGetFrequency,
+	VC_VoiceSetPanning,
+	VC_VoiceGetPanning,
+	VC_VoicePlay,
+	VC_VoiceStop,
+	VC_VoiceStopped,
+	VC_VoiceGetPosition,
+	VC_VoiceRealVolume
+};
+
+static int mod_mikModInitiated;
+static int mod_mikModInitError;
+
+static int mod_initMikMod(void)
+{
+	static char params[] = "";
+
+	if (mod_mikModInitError)
+		return -1;
+
+	if (!mod_mikModInitiated) {
+		mod_mikModInitiated = 1;
+
+		md_device = 0;
+		md_reverb = 0;
+
+		MikMod_RegisterDriver(&drv_mpd);
+		MikMod_RegisterAllLoaders();
+	}
+
+	md_pansep = 64;
+	md_mixfreq = 44100;
+	md_mode = (DMODE_SOFT_MUSIC | DMODE_INTERP | DMODE_STEREO |
+		   DMODE_16BITS);
+
+	if (MikMod_Init(params)) {
+		ERROR("Could not init MikMod: %s\n",
+		      MikMod_strerror(MikMod_errno));
+		mod_mikModInitError = 1;
+		return -1;
+	}
+
+	return 0;
+}
+
+static void mod_finishMikMod(void)
+{
+	MikMod_Exit();
+}
+
+typedef struct _mod_Data {
+	MODULE *moduleHandle;
+	SBYTE *audio_buffer;
+} mod_Data;
+
+static mod_Data *mod_open(char *path)
+{
+	MODULE *moduleHandle;
+	mod_Data *data;
+
+	if (!(moduleHandle = Player_Load(path, 128, 0)))
+		return NULL;
+
+	/* Prevent module from looping forever */
+	moduleHandle->loop = 0;
+
+	data = xmalloc(sizeof(mod_Data));
+
+	data->audio_buffer = xmalloc(MIKMOD_FRAME_SIZE);
+	data->moduleHandle = moduleHandle;
+
+	Player_Start(data->moduleHandle);
+
+	return data;
+}
+
+static void mod_close(mod_Data * data)
+{
+	Player_Stop();
+	Player_Free(data->moduleHandle);
+	free(data->audio_buffer);
+	free(data);
+}
+
+static int mod_decode(struct decoder * decoder, char *path)
+{
+	mod_Data *data;
+	struct audio_format audio_format;
+	float total_time = 0.0;
+	int ret;
+	float secPerByte;
+
+	if (mod_initMikMod() < 0)
+		return -1;
+
+	if (!(data = mod_open(path))) {
+		ERROR("failed to open mod: %s\n", path);
+		MikMod_Exit();
+		return -1;
+	}
+
+	audio_format.bits = 16;
+	audio_format.sample_rate = 44100;
+	audio_format.channels = 2;
+
+	secPerByte =
+	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
+		   (float)audio_format.sample_rate);
+
+	decoder_initialized(decoder, &audio_format, 0);
+
+	while (1) {
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+			decoder_seek_error(decoder);
+		}
+
+		if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
+			break;
+
+		if (!Player_Active())
+			break;
+
+		ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE);
+		total_time += ret * secPerByte;
+		decoder_data(decoder, NULL, 0,
+			     (char *)data->audio_buffer, ret,
+			     total_time, 0, NULL);
+	}
+
+	decoder_flush(decoder);
+
+	mod_close(data);
+
+	MikMod_Exit();
+
+	return 0;
+}
+
+static struct tag *modTagDup(char *file)
+{
+	struct tag *ret = NULL;
+	MODULE *moduleHandle;
+	char *title;
+
+	if (mod_initMikMod() < 0) {
+		DEBUG("modTagDup: Failed to initialize MikMod\n");
+		return NULL;
+	}
+
+	if (!(moduleHandle = Player_Load(file, 128, 0))) {
+		DEBUG("modTagDup: Failed to open file: %s\n", file);
+		MikMod_Exit();
+		return NULL;
+
+	}
+	Player_Free(moduleHandle);
+
+	ret = tag_new();
+
+	ret->time = 0;
+	title = xstrdup(Player_LoadTitle(file));
+	if (title)
+		tag_add_item(ret, TAG_ITEM_TITLE, title);
+
+	MikMod_Exit();
+
+	return ret;
+}
+
+static const char *modSuffixes[] = { "amf",
+	"dsm",
+	"far",
+	"gdm",
+	"imf",
+	"it",
+	"med",
+	"mod",
+	"mtm",
+	"s3m",
+	"stm",
+	"stx",
+	"ult",
+	"uni",
+	"xm",
+	NULL
+};
+
+struct decoder_plugin modPlugin = {
+	.name = "mod",
+	.finish = mod_finishMikMod,
+	.file_decode = mod_decode,
+	.tag_dup = modTagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = modSuffixes,
+};
diff --git a/src/decoder/mp3_plugin.c b/src/decoder/mp3_plugin.c
new file mode 100644
index 000000000..a0de30ba7
--- /dev/null
+++ b/src/decoder/mp3_plugin.c
@@ -0,0 +1,1086 @@
+/* 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 "../decoder_api.h"
+#include "../log.h"
+#include "../utils.h"
+#include "../conf.h"
+
+#include <mad.h>
+
+#ifdef HAVE_ID3TAG
+#include <id3tag.h>
+#endif
+
+#define FRAMES_CUSHION    2000
+
+#define READ_BUFFER_SIZE  40960
+
+enum mp3_action {
+	DECODE_SKIP = -3,
+	DECODE_BREAK = -2,
+	DECODE_CONT = -1,
+	DECODE_OK = 0
+};
+
+enum muteframe {
+	MUTEFRAME_NONE,
+	MUTEFRAME_SKIP,
+	MUTEFRAME_SEEK
+};
+
+/* the number of samples of silence the decoder inserts at start */
+#define DECODERDELAY 529
+
+#define DEFAULT_GAPLESS_MP3_PLAYBACK 1
+
+static int gaplessPlaybackEnabled;
+
+static inline int32_t
+mad_fixed_to_24_sample(mad_fixed_t sample)
+{
+	enum {
+		bits = 24,
+		MIN = -MAD_F_ONE,
+		MAX = MAD_F_ONE - 1
+	};
+
+	/* round */
+	sample = sample + (1L << (MAD_F_FRACBITS - bits));
+
+	/* clip */
+	if (sample > MAX)
+		sample = MAX;
+	else if (sample < MIN)
+		sample = MIN;
+
+	/* quantize */
+	return sample >> (MAD_F_FRACBITS + 1 - bits);
+}
+
+static void
+mad_fixed_to_24_buffer(int32_t *dest, const struct mad_synth *synth,
+		       unsigned int start, unsigned int end,
+		       unsigned int num_channels)
+{
+	unsigned int i, c;
+
+	for (i = start; i < end; ++i) {
+		for (c = 0; c < num_channels; ++c)
+			*dest++ = mad_fixed_to_24_sample(synth->pcm.samples[c][i]);
+	}
+}
+
+/* end of stolen stuff from mpg321 */
+
+static int mp3_plugin_init(void)
+{
+	gaplessPlaybackEnabled = getBoolConfigParam(CONF_GAPLESS_MP3_PLAYBACK,
+	                                            1);
+	if (gaplessPlaybackEnabled == CONF_BOOL_UNSET)
+		gaplessPlaybackEnabled = DEFAULT_GAPLESS_MP3_PLAYBACK;
+	return 1;
+}
+
+/* decoder stuff is based on madlld */
+
+#define MP3_DATA_OUTPUT_BUFFER_SIZE 2048
+
+typedef struct _mp3DecodeData {
+	struct mad_stream stream;
+	struct mad_frame frame;
+	struct mad_synth synth;
+	mad_timer_t timer;
+	unsigned char readBuffer[READ_BUFFER_SIZE];
+	int32_t outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
+	float totalTime;
+	float elapsedTime;
+	enum muteframe muteFrame;
+	long *frameOffset;
+	mad_timer_t *times;
+	unsigned long highestFrame;
+	unsigned long maxFrames;
+	unsigned long currentFrame;
+	unsigned int dropFramesAtStart;
+	unsigned int dropFramesAtEnd;
+	unsigned int dropSamplesAtStart;
+	unsigned int dropSamplesAtEnd;
+	int foundXing;
+	int foundFirstFrame;
+	int decodedFirstFrame;
+	unsigned long bitRate;
+	struct decoder *decoder;
+	InputStream *inStream;
+	enum mad_layer layer;
+} mp3DecodeData;
+
+static void initMp3DecodeData(mp3DecodeData * data, struct decoder *decoder,
+			      InputStream * inStream)
+{
+	data->muteFrame = MUTEFRAME_NONE;
+	data->highestFrame = 0;
+	data->maxFrames = 0;
+	data->frameOffset = NULL;
+	data->times = NULL;
+	data->currentFrame = 0;
+	data->dropFramesAtStart = 0;
+	data->dropFramesAtEnd = 0;
+	data->dropSamplesAtStart = 0;
+	data->dropSamplesAtEnd = 0;
+	data->foundXing = 0;
+	data->foundFirstFrame = 0;
+	data->decodedFirstFrame = 0;
+	data->decoder = decoder;
+	data->inStream = inStream;
+	data->layer = 0;
+
+	mad_stream_init(&data->stream);
+	mad_stream_options(&data->stream, MAD_OPTION_IGNORECRC);
+	mad_frame_init(&data->frame);
+	mad_synth_init(&data->synth);
+	mad_timer_reset(&data->timer);
+}
+
+static int seekMp3InputBuffer(mp3DecodeData * data, long offset)
+{
+	if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
+		return -1;
+	}
+
+	mad_stream_buffer(&data->stream, data->readBuffer, 0);
+	(data->stream).error = 0;
+
+	return 0;
+}
+
+static int fillMp3InputBuffer(mp3DecodeData * data)
+{
+	size_t readSize;
+	size_t remaining;
+	size_t readed;
+	unsigned char *readStart;
+
+	if ((data->stream).next_frame != NULL) {
+		remaining = (data->stream).bufend - (data->stream).next_frame;
+		memmove(data->readBuffer, (data->stream).next_frame, remaining);
+		readStart = (data->readBuffer) + remaining;
+		readSize = READ_BUFFER_SIZE - remaining;
+	} else {
+		readSize = READ_BUFFER_SIZE;
+		readStart = data->readBuffer, remaining = 0;
+	}
+
+	/* we've exhausted the read buffer, so give up!, these potential
+	 * mp3 frames are way too big, and thus unlikely to be mp3 frames */
+	if (readSize == 0)
+		return -1;
+
+	readed = decoder_read(data->decoder, data->inStream,
+			      readStart, readSize);
+	if (readed == 0)
+		return -1;
+
+	mad_stream_buffer(&data->stream, data->readBuffer, readed + remaining);
+	(data->stream).error = 0;
+
+	return 0;
+}
+
+#ifdef HAVE_ID3TAG
+static ReplayGainInfo *parseId3ReplayGainInfo(struct id3_tag *tag)
+{
+	int i;
+	char *key;
+	char *value;
+	struct id3_frame *frame;
+	int found = 0;
+	ReplayGainInfo *replayGainInfo;
+
+	replayGainInfo = newReplayGainInfo();
+
+	for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) {
+		if (frame->nfields < 3)
+			continue;
+
+		key = (char *)
+		    id3_ucs4_latin1duplicate(id3_field_getstring
+					     (&frame->fields[1]));
+		value = (char *)
+		    id3_ucs4_latin1duplicate(id3_field_getstring
+					     (&frame->fields[2]));
+
+		if (strcasecmp(key, "replaygain_track_gain") == 0) {
+			replayGainInfo->trackGain = atof(value);
+			found = 1;
+		} else if (strcasecmp(key, "replaygain_album_gain") == 0) {
+			replayGainInfo->albumGain = atof(value);
+			found = 1;
+		} else if (strcasecmp(key, "replaygain_track_peak") == 0) {
+			replayGainInfo->trackPeak = atof(value);
+			found = 1;
+		} else if (strcasecmp(key, "replaygain_album_peak") == 0) {
+			replayGainInfo->albumPeak = atof(value);
+			found = 1;
+		}
+
+		free(key);
+		free(value);
+	}
+
+	if (found)
+		return replayGainInfo;
+	freeReplayGainInfo(replayGainInfo);
+	return NULL;
+}
+#endif
+
+#ifdef HAVE_ID3TAG
+static void mp3_parseId3Tag(mp3DecodeData * data, size_t tagsize,
+			    struct tag ** mpdTag, ReplayGainInfo ** replayGainInfo)
+{
+	struct id3_tag *id3Tag = NULL;
+	id3_length_t count;
+	id3_byte_t const *id3_data;
+	id3_byte_t *allocated = NULL;
+	struct tag *tmpMpdTag;
+	ReplayGainInfo *tmpReplayGainInfo;
+
+	count = data->stream.bufend - data->stream.this_frame;
+
+	if (tagsize <= count) {
+		id3_data = data->stream.this_frame;
+		mad_stream_skip(&(data->stream), tagsize);
+	} else {
+		allocated = xmalloc(tagsize);
+		if (!allocated)
+			goto fail;
+
+		memcpy(allocated, data->stream.this_frame, count);
+		mad_stream_skip(&(data->stream), count);
+
+		while (count < tagsize) {
+			size_t len;
+
+			len = decoder_read(data->decoder, data->inStream,
+					   allocated + count, tagsize - count);
+			if (len == 0)
+				break;
+			else
+				count += len;
+		}
+
+		if (count != tagsize) {
+			DEBUG("mp3_decode: error parsing ID3 tag\n");
+			goto fail;
+		}
+
+		id3_data = allocated;
+	}
+
+	id3Tag = id3_tag_parse(id3_data, tagsize);
+	if (!id3Tag)
+		goto fail;
+
+	if (mpdTag) {
+		tmpMpdTag = tag_id3_import(id3Tag);
+		if (tmpMpdTag) {
+			if (*mpdTag)
+				tag_free(*mpdTag);
+			*mpdTag = tmpMpdTag;
+		}
+	}
+
+	if (replayGainInfo) {
+		tmpReplayGainInfo = parseId3ReplayGainInfo(id3Tag);
+		if (tmpReplayGainInfo) {
+			if (*replayGainInfo)
+				freeReplayGainInfo(*replayGainInfo);
+			*replayGainInfo = tmpReplayGainInfo;
+		}
+	}
+
+	id3_tag_delete(id3Tag);
+fail:
+	if (allocated)
+		free(allocated);
+}
+#endif
+
+static enum mp3_action
+decodeNextFrameHeader(mp3DecodeData * data, struct tag ** tag,
+		      ReplayGainInfo ** replayGainInfo)
+{
+	enum mad_layer layer;
+
+	if ((data->stream).buffer == NULL
+	    || (data->stream).error == MAD_ERROR_BUFLEN) {
+		if (fillMp3InputBuffer(data) < 0) {
+			return DECODE_BREAK;
+		}
+	}
+	if (mad_header_decode(&data->frame.header, &data->stream)) {
+#ifdef HAVE_ID3TAG
+		if ((data->stream).error == MAD_ERROR_LOSTSYNC &&
+		    (data->stream).this_frame) {
+			signed long tagsize = id3_tag_query((data->stream).
+							    this_frame,
+							    (data->stream).
+							    bufend -
+							    (data->stream).
+							    this_frame);
+
+			if (tagsize > 0) {
+				if (tag && !(*tag)) {
+					mp3_parseId3Tag(data, (size_t)tagsize,
+							tag, replayGainInfo);
+				} else {
+					mad_stream_skip(&(data->stream),
+							tagsize);
+				}
+				return DECODE_CONT;
+			}
+		}
+#endif
+		if (MAD_RECOVERABLE((data->stream).error)) {
+			return DECODE_SKIP;
+		} else {
+			if ((data->stream).error == MAD_ERROR_BUFLEN)
+				return DECODE_CONT;
+			else {
+				ERROR("unrecoverable frame level error "
+				      "(%s).\n",
+				      mad_stream_errorstr(&data->stream));
+				return DECODE_BREAK;
+			}
+		}
+	}
+
+	layer = data->frame.header.layer;
+	if (!data->layer) {
+		if (layer != MAD_LAYER_II && layer != MAD_LAYER_III) {
+			/* Only layer 2 and 3 have been tested to work */
+			return DECODE_SKIP;
+		}
+		data->layer = layer;
+	} else if (layer != data->layer) {
+		/* Don't decode frames with a different layer than the first */
+		return DECODE_SKIP;
+	}
+
+	return DECODE_OK;
+}
+
+static enum mp3_action
+decodeNextFrame(mp3DecodeData * data)
+{
+	if ((data->stream).buffer == NULL
+	    || (data->stream).error == MAD_ERROR_BUFLEN) {
+		if (fillMp3InputBuffer(data) < 0) {
+			return DECODE_BREAK;
+		}
+	}
+	if (mad_frame_decode(&data->frame, &data->stream)) {
+#ifdef HAVE_ID3TAG
+		if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
+			signed long tagsize = id3_tag_query((data->stream).
+							    this_frame,
+							    (data->stream).
+							    bufend -
+							    (data->stream).
+							    this_frame);
+			if (tagsize > 0) {
+				mad_stream_skip(&(data->stream), tagsize);
+				return DECODE_CONT;
+			}
+		}
+#endif
+		if (MAD_RECOVERABLE((data->stream).error)) {
+			return DECODE_SKIP;
+		} else {
+			if ((data->stream).error == MAD_ERROR_BUFLEN)
+				return DECODE_CONT;
+			else {
+				ERROR("unrecoverable frame level error "
+				      "(%s).\n",
+				      mad_stream_errorstr(&data->stream));
+				return DECODE_BREAK;
+			}
+		}
+	}
+
+	return DECODE_OK;
+}
+
+/* xing stuff stolen from alsaplayer, and heavily modified by jat */
+#define XI_MAGIC (('X' << 8) | 'i')
+#define NG_MAGIC (('n' << 8) | 'g')
+#define IN_MAGIC (('I' << 8) | 'n')
+#define FO_MAGIC (('f' << 8) | 'o')
+
+enum xing_magic {
+	XING_MAGIC_XING, /* VBR */
+	XING_MAGIC_INFO  /* CBR */
+};
+
+struct xing {
+	long flags;             /* valid fields (see below) */
+	unsigned long frames;   /* total number of frames */
+	unsigned long bytes;    /* total number of bytes */
+	unsigned char toc[100]; /* 100-point seek table */
+	long scale;             /* VBR quality */
+	enum xing_magic magic;  /* header magic */
+};
+
+enum {
+	XING_FRAMES = 0x00000001L,
+	XING_BYTES  = 0x00000002L,
+	XING_TOC    = 0x00000004L,
+	XING_SCALE  = 0x00000008L
+};
+
+struct version {
+	unsigned major;
+	unsigned minor;
+};
+
+struct lame {
+	char encoder[10];       /* 9 byte encoder name/version ("LAME3.97b") */
+	struct version version; /* struct containing just the version */
+	float peak;             /* replaygain peak */
+	float trackGain;        /* replaygain track gain */
+	float albumGain;        /* replaygain album gain */
+	int encoderDelay;       /* # of added samples at start of mp3 */
+	int encoderPadding;     /* # of added samples at end of mp3 */
+	int crc;                /* CRC of the first 190 bytes of this frame */
+};
+
+static int parse_xing(struct xing *xing, struct mad_bitptr *ptr, int *oldbitlen)
+{
+	unsigned long bits;
+	int bitlen;
+	int bitsleft;
+	int i;
+
+	bitlen = *oldbitlen;
+
+	if (bitlen < 16) goto fail;
+	bits = mad_bit_read(ptr, 16);
+	bitlen -= 16;
+
+	if (bits == XI_MAGIC) {
+		if (bitlen < 16) goto fail;
+		if (mad_bit_read(ptr, 16) != NG_MAGIC) goto fail;
+		bitlen -= 16;
+		xing->magic = XING_MAGIC_XING;
+	} else if (bits == IN_MAGIC) {
+		if (bitlen < 16) goto fail;
+		if (mad_bit_read(ptr, 16) != FO_MAGIC) goto fail;
+		bitlen -= 16;
+		xing->magic = XING_MAGIC_INFO;
+	}
+	else if (bits == NG_MAGIC) xing->magic = XING_MAGIC_XING;
+	else if (bits == FO_MAGIC) xing->magic = XING_MAGIC_INFO;
+	else goto fail;
+
+	if (bitlen < 32) goto fail;
+	xing->flags = mad_bit_read(ptr, 32);
+	bitlen -= 32;
+
+	if (xing->flags & XING_FRAMES) {
+		if (bitlen < 32) goto fail;
+		xing->frames = mad_bit_read(ptr, 32);
+		bitlen -= 32;
+	}
+
+	if (xing->flags & XING_BYTES) {
+		if (bitlen < 32) goto fail;
+		xing->bytes = mad_bit_read(ptr, 32);
+		bitlen -= 32;
+	}
+
+	if (xing->flags & XING_TOC) {
+		if (bitlen < 800) goto fail;
+		for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(ptr, 8);
+		bitlen -= 800;
+	}
+
+	if (xing->flags & XING_SCALE) {
+		if (bitlen < 32) goto fail;
+		xing->scale = mad_bit_read(ptr, 32);
+		bitlen -= 32;
+	}
+
+	/* Make sure we consume no less than 120 bytes (960 bits) in hopes that
+	 * the LAME tag is found there, and not right after the Xing header */
+	bitsleft = 960 - ((*oldbitlen) - bitlen);
+	if (bitsleft < 0) goto fail;
+	else if (bitsleft > 0) {
+		mad_bit_read(ptr, bitsleft);
+		bitlen -= bitsleft;
+	}
+
+	*oldbitlen = bitlen;
+
+	return 1;
+fail:
+	xing->flags = 0;
+	return 0;
+}
+
+static int parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen)
+{
+	int adj = 0;
+	int name;
+	int orig;
+	int sign;
+	int gain;
+	int i;
+
+	/* Unlike the xing header, the lame tag has a fixed length.  Fail if
+	 * not all 36 bytes (288 bits) are there. */
+	if (*bitlen < 288)
+		return 0;
+
+	for (i = 0; i < 9; i++)
+		lame->encoder[i] = (char)mad_bit_read(ptr, 8);
+	lame->encoder[9] = '\0';
+
+	*bitlen -= 72;
+
+	/* This is technically incorrect, since the encoder might not be lame.
+	 * But there's no other way to determine if this is a lame tag, and we
+	 * wouldn't want to go reading a tag that's not there. */
+	if (prefixcmp(lame->encoder, "LAME"))
+		return 0;
+
+	if (sscanf(lame->encoder+4, "%u.%u",
+	           &lame->version.major, &lame->version.minor) != 2)
+		return 0;
+
+	DEBUG("detected LAME version %i.%i (\"%s\")\n",
+	      lame->version.major, lame->version.minor, lame->encoder);
+
+	/* The reference volume was changed from the 83dB used in the
+	 * ReplayGain spec to 89dB in lame 3.95.1.  Bump the gain for older
+	 * versions, since everyone else uses 89dB instead of 83dB.
+	 * Unfortunately, lame didn't differentiate between 3.95 and 3.95.1, so
+	 * it's impossible to make the proper adjustment for 3.95.
+	 * Fortunately, 3.95 was only out for about a day before 3.95.1 was
+	 * released. -- tmz */
+	if (lame->version.major < 3 ||
+	    (lame->version.major == 3 && lame->version.minor < 95))
+		adj = 6;
+
+	mad_bit_read(ptr, 16);
+
+	lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */
+	DEBUG("LAME peak found: %f\n", lame->peak);
+
+	lame->trackGain = 0;
+	name = mad_bit_read(ptr, 3); /* gain name */
+	orig = mad_bit_read(ptr, 3); /* gain originator */
+	sign = mad_bit_read(ptr, 1); /* sign bit */
+	gain = mad_bit_read(ptr, 9); /* gain*10 */
+	if (gain && name == 1 && orig != 0) {
+		lame->trackGain = ((sign ? -gain : gain) / 10.0) + adj;
+		DEBUG("LAME track gain found: %f\n", lame->trackGain);
+	}
+
+	/* tmz reports that this isn't currently written by any version of lame
+	 * (as of 3.97).  Since we have no way of testing it, don't use it.
+	 * Wouldn't want to go blowing someone's ears just because we read it
+	 * wrong. :P -- jat */
+	lame->albumGain = 0;
+#if 0
+	name = mad_bit_read(ptr, 3); /* gain name */
+	orig = mad_bit_read(ptr, 3); /* gain originator */
+	sign = mad_bit_read(ptr, 1); /* sign bit */
+	gain = mad_bit_read(ptr, 9); /* gain*10 */
+	if (gain && name == 2 && orig != 0) {
+		lame->albumGain = ((sign ? -gain : gain) / 10.0) + adj;
+		DEBUG("LAME album gain found: %f\n", lame->trackGain);
+	}
+#else
+	mad_bit_read(ptr, 16);
+#endif
+
+	mad_bit_read(ptr, 16);
+
+	lame->encoderDelay = mad_bit_read(ptr, 12);
+	lame->encoderPadding = mad_bit_read(ptr, 12);
+
+	DEBUG("encoder delay is %i, encoder padding is %i\n",
+	      lame->encoderDelay, lame->encoderPadding);
+
+	mad_bit_read(ptr, 80);
+
+	lame->crc = mad_bit_read(ptr, 16);
+
+	*bitlen -= 216;
+
+	return 1;
+}
+
+static int decodeFirstFrame(mp3DecodeData * data,
+                            struct tag ** tag, ReplayGainInfo ** replayGainInfo)
+{
+	struct decoder *decoder = data->decoder;
+	struct xing xing;
+	struct lame lame;
+	struct mad_bitptr ptr;
+	int bitlen;
+	int ret;
+
+	/* stfu gcc */
+	memset(&xing, 0, sizeof(struct xing));
+	xing.flags = 0;
+
+	while (1) {
+		while ((ret = decodeNextFrameHeader(data, tag, replayGainInfo)) == DECODE_CONT &&
+		       (!decoder || decoder_get_command(decoder) == DECODE_COMMAND_NONE));
+		if (ret == DECODE_BREAK ||
+		    (decoder && decoder_get_command(decoder) != DECODE_COMMAND_NONE))
+			return -1;
+		if (ret == DECODE_SKIP) continue;
+
+		while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
+		       (!decoder || decoder_get_command(decoder) == DECODE_COMMAND_NONE));
+		if (ret == DECODE_BREAK ||
+		    (decoder && decoder_get_command(decoder) != DECODE_COMMAND_NONE))
+			return -1;
+		if (ret == DECODE_OK) break;
+	}
+
+	ptr = data->stream.anc_ptr;
+	bitlen = data->stream.anc_bitlen;
+
+	/*
+	 * Attempt to calulcate the length of the song from filesize
+	 */
+	{
+		size_t offset = data->inStream->offset;
+		mad_timer_t duration = data->frame.header.duration;
+		float frameTime = ((float)mad_timer_count(duration,
+		                   MAD_UNITS_MILLISECONDS)) / 1000;
+
+		if (data->stream.this_frame != NULL)
+			offset -= data->stream.bufend - data->stream.this_frame;
+		else
+			offset -= data->stream.bufend - data->stream.buffer;
+
+		if (data->inStream->size >= offset) {
+			data->totalTime = ((data->inStream->size - offset) *
+			                   8.0) / (data->frame).header.bitrate;
+			data->maxFrames = data->totalTime / frameTime +
+			                  FRAMES_CUSHION;
+		} else {
+			data->maxFrames = FRAMES_CUSHION;
+			data->totalTime = 0;
+		}
+	}
+	/*
+	 * if an xing tag exists, use that!
+	 */
+	if (parse_xing(&xing, &ptr, &bitlen)) {
+		data->foundXing = 1;
+		data->muteFrame = MUTEFRAME_SKIP;
+
+		if ((xing.flags & XING_FRAMES) && xing.frames) {
+			mad_timer_t duration = data->frame.header.duration;
+			mad_timer_multiply(&duration, xing.frames);
+			data->totalTime = ((float)mad_timer_count(duration, MAD_UNITS_MILLISECONDS)) / 1000;
+			data->maxFrames = xing.frames;
+		}
+
+		if (parse_lame(&lame, &ptr, &bitlen)) {
+			if (gaplessPlaybackEnabled &&
+			    data->inStream->seekable) {
+				data->dropSamplesAtStart = lame.encoderDelay +
+				                           DECODERDELAY;
+				data->dropSamplesAtEnd = lame.encoderPadding;
+			}
+
+			/* Album gain isn't currently used.  See comment in
+			 * parse_lame() for details. -- jat */
+			if (replayGainInfo && !*replayGainInfo &&
+			    lame.trackGain) {
+				*replayGainInfo = newReplayGainInfo();
+				(*replayGainInfo)->trackGain = lame.trackGain;
+				(*replayGainInfo)->trackPeak = lame.peak;
+			}
+		}
+	} 
+
+	if (!data->maxFrames) return -1;
+
+	if (data->maxFrames > 8 * 1024 * 1024) {
+		ERROR("mp3 file header indicates too many frames: %lu",
+		      data->maxFrames);
+		return -1;
+	}
+
+	data->frameOffset = xmalloc(sizeof(long) * data->maxFrames);
+	data->times = xmalloc(sizeof(mad_timer_t) * data->maxFrames);
+
+	return 0;
+}
+
+static void mp3DecodeDataFinalize(mp3DecodeData * data)
+{
+	mad_synth_finish(&data->synth);
+	mad_frame_finish(&data->frame);
+	mad_stream_finish(&data->stream);
+
+	if (data->frameOffset) free(data->frameOffset);
+	if (data->times) free(data->times);
+}
+
+/* this is primarily used for getting total time for tags */
+static int getMp3TotalTime(char *file)
+{
+	InputStream inStream;
+	mp3DecodeData data;
+	int ret;
+
+	if (openInputStream(&inStream, file) < 0)
+		return -1;
+	initMp3DecodeData(&data, NULL, &inStream);
+	if (decodeFirstFrame(&data, NULL, NULL) < 0)
+		ret = -1;
+	else
+		ret = data.totalTime + 0.5;
+	mp3DecodeDataFinalize(&data);
+	closeInputStream(&inStream);
+
+	return ret;
+}
+
+static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
+				  struct decoder * decoder, struct tag ** tag,
+				  ReplayGainInfo ** replayGainInfo)
+{
+	initMp3DecodeData(data, decoder, inStream);
+	*tag = NULL;
+	if (decodeFirstFrame(data, tag, replayGainInfo) < 0) {
+		mp3DecodeDataFinalize(data);
+		if (tag && *tag)
+			tag_free(*tag);
+		return -1;
+	}
+
+	return 0;
+}
+
+static enum mp3_action
+mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
+{
+	struct decoder *decoder = data->decoder;
+	unsigned int pcm_length, max_samples;
+	unsigned int i;
+	int ret;
+	int skip;
+
+	if (data->currentFrame >= data->highestFrame) {
+		mad_timer_add(&data->timer, (data->frame).header.duration);
+		data->bitRate = (data->frame).header.bitrate;
+		if (data->currentFrame >= data->maxFrames) {
+			data->currentFrame = data->maxFrames - 1;
+		} else {
+			data->highestFrame++;
+		}
+		data->frameOffset[data->currentFrame] = data->inStream->offset;
+		if (data->stream.this_frame != NULL) {
+			data->frameOffset[data->currentFrame] -=
+			    data->stream.bufend - data->stream.this_frame;
+		} else {
+			data->frameOffset[data->currentFrame] -=
+			    data->stream.bufend - data->stream.buffer;
+		}
+		data->times[data->currentFrame] = data->timer;
+	} else {
+		data->timer = data->times[data->currentFrame];
+	}
+	data->currentFrame++;
+	data->elapsedTime =
+	    ((float)mad_timer_count(data->timer, MAD_UNITS_MILLISECONDS)) /
+	    1000;
+
+	switch (data->muteFrame) {
+	case MUTEFRAME_SKIP:
+		data->muteFrame = MUTEFRAME_NONE;
+		break;
+	case MUTEFRAME_SEEK:
+		if (decoder_seek_where(decoder) <= data->elapsedTime) {
+			decoder_clear(decoder);
+			data->muteFrame = MUTEFRAME_NONE;
+			decoder_command_finished(decoder);
+		}
+		break;
+	case MUTEFRAME_NONE:
+		mad_synth_frame(&data->synth, &data->frame);
+
+		if (!data->foundFirstFrame) {
+			unsigned int samplesPerFrame = (data->synth).pcm.length;
+			data->dropFramesAtStart = data->dropSamplesAtStart / samplesPerFrame;
+			data->dropFramesAtEnd = data->dropSamplesAtEnd / samplesPerFrame;
+			data->dropSamplesAtStart = data->dropSamplesAtStart % samplesPerFrame;
+			data->dropSamplesAtEnd = data->dropSamplesAtEnd % samplesPerFrame;
+			data->foundFirstFrame = 1;
+		}
+
+		if (data->dropFramesAtStart > 0) {
+			data->dropFramesAtStart--;
+			break;
+		} else if ((data->dropFramesAtEnd > 0) && 
+		           (data->currentFrame == (data->maxFrames + 1 - data->dropFramesAtEnd))) {
+			/* stop decoding, effectively dropping all remaining
+			 * frames */
+			return DECODE_BREAK;
+		}
+
+		if (data->inStream->metaTitle) {
+			struct tag *tag = tag_new();
+			if (data->inStream->metaName) {
+				tag_add_item(tag, TAG_ITEM_NAME,
+					     data->inStream->metaName);
+			}
+			tag_add_item(tag, TAG_ITEM_TITLE,
+					data->inStream->metaTitle);
+			free(data->inStream->metaTitle);
+			data->inStream->metaTitle = NULL;
+			tag_free(tag);
+		}
+
+		if (!data->decodedFirstFrame) {
+			i = data->dropSamplesAtStart;
+			data->decodedFirstFrame = 1;
+		} else
+			i = 0;
+
+		pcm_length = data->synth.pcm.length;
+		if (data->dropSamplesAtEnd &&
+		    (data->currentFrame == data->maxFrames - data->dropFramesAtEnd)) {
+			if (data->dropSamplesAtEnd >= pcm_length)
+				pcm_length = 0;
+			else
+				pcm_length -= data->dropSamplesAtEnd;
+		}
+
+		max_samples = sizeof(data->outputBuffer) /
+			sizeof(data->outputBuffer[0]) /
+			MAD_NCHANNELS(&(data->frame).header);
+
+		while (i < pcm_length) {
+			enum decoder_command cmd;
+			unsigned int num_samples = pcm_length - i;
+			if (num_samples > max_samples)
+				num_samples = max_samples;
+
+			i += num_samples;
+
+			mad_fixed_to_24_buffer(data->outputBuffer,
+					       &data->synth,
+					       i - num_samples, i,
+					       MAD_NCHANNELS(&(data->frame).header));
+			num_samples *= MAD_NCHANNELS(&(data->frame).header);
+
+			cmd = decoder_data(decoder, data->inStream,
+					   data->inStream->seekable,
+					   data->outputBuffer,
+					   sizeof(data->outputBuffer[0]) * num_samples,
+					   data->elapsedTime,
+					   data->bitRate / 1000,
+					   (replayGainInfo != NULL) ? *replayGainInfo : NULL);
+			if (cmd == DECODE_COMMAND_STOP)
+				return DECODE_BREAK;
+		}
+
+		if (data->dropSamplesAtEnd &&
+		    (data->currentFrame == data->maxFrames - data->dropFramesAtEnd))
+			/* stop decoding, effectively dropping
+			 * all remaining samples */
+			return DECODE_BREAK;
+
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+		    data->inStream->seekable) {
+			unsigned long j = 0;
+			data->muteFrame = MUTEFRAME_SEEK;
+			while (j < data->highestFrame &&
+			       decoder_seek_where(decoder) >
+			       ((float)mad_timer_count(data->times[j],
+						       MAD_UNITS_MILLISECONDS))
+			       / 1000) {
+				j++;
+			}
+			if (j < data->highestFrame) {
+				if (seekMp3InputBuffer(data,
+						       data->frameOffset[j]) ==
+				    0) {
+					decoder_clear(decoder);
+					data->currentFrame = j;
+					decoder_command_finished(decoder);
+				} else
+					decoder_seek_error(decoder);
+				data->muteFrame = MUTEFRAME_NONE;
+			}
+		} else if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+			   !data->inStream->seekable) {
+			decoder_seek_error(decoder);
+		}
+	}
+
+	while (1) {
+		skip = 0;
+		while ((ret =
+			decodeNextFrameHeader(data, NULL,
+					      replayGainInfo)) == DECODE_CONT
+		       && decoder_get_command(decoder) == DECODE_COMMAND_NONE) ;
+		if (ret == DECODE_BREAK || decoder_get_command(decoder) != DECODE_COMMAND_NONE)
+			break;
+		else if (ret == DECODE_SKIP)
+			skip = 1;
+		if (data->muteFrame == MUTEFRAME_NONE) {
+			while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
+			       decoder_get_command(decoder) == DECODE_COMMAND_NONE) ;
+			if (ret == DECODE_BREAK ||
+			    decoder_get_command(decoder) != DECODE_COMMAND_NONE)
+				break;
+		}
+		if (!skip && ret == DECODE_OK)
+			break;
+	}
+
+	switch (decoder_get_command(decoder)) {
+	case DECODE_COMMAND_NONE:
+	case DECODE_COMMAND_START:
+		break;
+
+	case DECODE_COMMAND_STOP:
+		return DECODE_BREAK;
+
+	case DECODE_COMMAND_SEEK:
+		return DECODE_CONT;
+	}
+
+	return ret;
+}
+
+static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
+					     struct audio_format * af)
+{
+	af->bits = 24;
+	af->sample_rate = (data->frame).header.samplerate;
+	af->channels = MAD_NCHANNELS(&(data->frame).header);
+}
+
+static int mp3_decode(struct decoder * decoder, InputStream * inStream)
+{
+	mp3DecodeData data;
+	struct tag *tag = NULL;
+	ReplayGainInfo *replayGainInfo = NULL;
+	struct audio_format audio_format;
+
+	if (openMp3FromInputStream(inStream, &data, decoder,
+				   &tag, &replayGainInfo) < 0) {
+		if (decoder_get_command(decoder) == DECODE_COMMAND_NONE) {
+			ERROR
+			    ("Input does not appear to be a mp3 bit stream.\n");
+			return -1;
+		}
+		return 0;
+	}
+
+	initAudioFormatFromMp3DecodeData(&data, &audio_format);
+
+	if (inStream->metaTitle) {
+		if (tag)
+			tag_free(tag);
+		tag = tag_new();
+		tag_add_item(tag, TAG_ITEM_TITLE, inStream->metaTitle);
+		free(inStream->metaTitle);
+		inStream->metaTitle = NULL;
+		if (inStream->metaName) {
+			tag_add_item(tag, TAG_ITEM_NAME, inStream->metaName);
+		}
+		tag_free(tag);
+	} else if (tag) {
+		if (inStream->metaName) {
+			tag_clear_items_by_type(tag, TAG_ITEM_NAME);
+			tag_add_item(tag, TAG_ITEM_NAME, inStream->metaName);
+		}
+		tag_free(tag);
+	} else if (inStream->metaName) {
+		tag = tag_new();
+		if (inStream->metaName) {
+			tag_add_item(tag, TAG_ITEM_NAME, inStream->metaName);
+		}
+		tag_free(tag);
+	}
+
+	decoder_initialized(decoder, &audio_format, data.totalTime);
+
+	while (mp3Read(&data, &replayGainInfo) != DECODE_BREAK) ;
+
+	if (replayGainInfo)
+		freeReplayGainInfo(replayGainInfo);
+
+	if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+	    data.muteFrame == MUTEFRAME_SEEK) {
+		decoder_clear(decoder);
+		decoder_command_finished(decoder);
+	}
+
+	decoder_flush(decoder);
+	mp3DecodeDataFinalize(&data);
+
+	return 0;
+}
+
+static struct tag *mp3_tagDup(char *file)
+{
+	struct tag *ret = NULL;
+	int total_time;
+
+	ret = tag_id3_load(file);
+
+	total_time = getMp3TotalTime(file);
+
+	if (total_time >= 0) {
+		if (!ret)
+			ret = tag_new();
+		ret->time = total_time;
+	} else {
+		DEBUG("mp3_tagDup: Failed to get total song time from: %s\n",
+		      file);
+	}
+
+	return ret;
+}
+
+static const char *mp3_suffixes[] = { "mp3", "mp2", NULL };
+static const char *mp3_mimeTypes[] = { "audio/mpeg", NULL };
+
+struct decoder_plugin mp3Plugin = {
+	.name = "mp3",
+	.init = mp3_plugin_init,
+	.stream_decode = mp3_decode,
+	.tag_dup = mp3_tagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+	.suffixes = mp3_suffixes,
+	.mime_types = mp3_mimeTypes
+};
diff --git a/src/decoder/mp4_plugin.c b/src/decoder/mp4_plugin.c
new file mode 100644
index 000000000..4a613744e
--- /dev/null
+++ b/src/decoder/mp4_plugin.c
@@ -0,0 +1,423 @@
+/* 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 "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include "mp4ff.h"
+
+#include <limits.h>
+#include <faad.h>
+/* all code here is either based on or copied from FAAD2's frontend code */
+
+static int mp4_getAACTrack(mp4ff_t * infile)
+{
+	/* find AAC track */
+	int i, rc;
+	int numTracks = mp4ff_total_tracks(infile);
+
+	for (i = 0; i < numTracks; i++) {
+		unsigned char *buff = NULL;
+		unsigned int buff_size = 0;
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
+		mp4AudioSpecificConfig mp4ASC;
+#else
+		unsigned long dummy1_32;
+		unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
+		    dummy7_8, dummy8_8;
+#endif
+
+		mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
+
+		if (buff) {
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
+			rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+#else
+			rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
+						 &dummy3_8, &dummy4_8,
+						 &dummy5_8, &dummy6_8,
+						 &dummy7_8, &dummy8_8);
+#endif
+			free(buff);
+			if (rc < 0)
+				continue;
+			return i;
+		}
+	}
+
+	/* can't decode this */
+	return -1;
+}
+
+static uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer,
+					    uint32_t length)
+{
+	return readFromInputStream((InputStream *) inStream, buffer, length);
+}
+
+static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
+{
+	return seekInputStream((InputStream *) inStream, position, SEEK_SET);
+}
+
+static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
+{
+	mp4ff_t *mp4fh;
+	mp4ff_callback_t *mp4cb;
+	int32_t track;
+	float file_time, total_time;
+	int32_t scale;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	struct audio_format audio_format;
+	unsigned char *mp4Buffer;
+	unsigned int mp4BufferSize;
+	uint32_t sample_rate;
+	unsigned char channels;
+	long sampleId;
+	long numSamples;
+	long dur;
+	unsigned int sampleCount;
+	char *sampleBuffer;
+	size_t sampleBufferLen;
+	unsigned int initial = 1;
+	float *seekTable;
+	long seekTableEnd = -1;
+	bool seekPositionFound = false;
+	long offset;
+	uint16_t bitRate = 0;
+	bool seeking = false;
+	double seek_where = 0;
+	bool initialized = false;
+
+	mp4cb = xmalloc(sizeof(mp4ff_callback_t));
+	mp4cb->read = mp4_inputStreamReadCallback;
+	mp4cb->seek = mp4_inputStreamSeekCallback;
+	mp4cb->user_data = inStream;
+
+	mp4fh = mp4ff_open_read(mp4cb);
+	if (!mp4fh) {
+		ERROR("Input does not appear to be a mp4 stream.\n");
+		free(mp4cb);
+		return -1;
+	}
+
+	track = mp4_getAACTrack(mp4fh);
+	if (track < 0) {
+		ERROR("No AAC track found in mp4 stream.\n");
+		mp4ff_close(mp4fh);
+		free(mp4cb);
+		return -1;
+	}
+
+	decoder = faacDecOpen();
+
+	config = faacDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+	config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+	config->dontUpSampleImplicitSBR = 0;
+#endif
+	faacDecSetConfiguration(decoder, config);
+
+	audio_format.bits = 16;
+
+	mp4Buffer = NULL;
+	mp4BufferSize = 0;
+	mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
+
+	if (faacDecInit2
+	    (decoder, mp4Buffer, mp4BufferSize, &sample_rate, &channels) < 0) {
+		ERROR("Error not a AAC stream.\n");
+		faacDecClose(decoder);
+		mp4ff_close(mp4fh);
+		free(mp4cb);
+		return -1;
+	}
+
+	audio_format.sample_rate = sample_rate;
+	audio_format.channels = channels;
+	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+	scale = mp4ff_time_scale(mp4fh, track);
+
+	if (mp4Buffer)
+		free(mp4Buffer);
+
+	if (scale < 0) {
+		ERROR("Error getting audio format of mp4 AAC track.\n");
+		faacDecClose(decoder);
+		mp4ff_close(mp4fh);
+		free(mp4cb);
+		return -1;
+	}
+	total_time = ((float)file_time) / scale;
+
+	numSamples = mp4ff_num_samples(mp4fh, track);
+	if (numSamples > (long)(INT_MAX / sizeof(float))) {
+		 ERROR("Integer overflow.\n");
+		 faacDecClose(decoder);
+		 mp4ff_close(mp4fh);
+		 free(mp4cb);
+		 return -1;
+	}
+
+	file_time = 0.0;
+
+	seekTable = xmalloc(sizeof(float) * numSamples);
+
+	for (sampleId = 0; sampleId < numSamples; sampleId++) {
+		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+			seeking = true;
+			seek_where = decoder_seek_where(mpd_decoder);
+		}
+
+		if (seeking && seekTableEnd > 1 &&
+		    seekTable[seekTableEnd] >= seek_where) {
+			int i = 2;
+			while (seekTable[i] < seek_where)
+				i++;
+			sampleId = i - 1;
+			file_time = seekTable[sampleId];
+		}
+
+		dur = mp4ff_get_sample_duration(mp4fh, track, sampleId);
+		offset = mp4ff_get_sample_offset(mp4fh, track, sampleId);
+
+		if (sampleId > seekTableEnd) {
+			seekTable[sampleId] = file_time;
+			seekTableEnd = sampleId;
+		}
+
+		if (sampleId == 0)
+			dur = 0;
+		if (offset > dur)
+			dur = 0;
+		else
+			dur -= offset;
+		file_time += ((float)dur) / scale;
+
+		if (seeking && file_time > seek_where)
+			seekPositionFound = true;
+
+		if (seeking && seekPositionFound) {
+			seekPositionFound = false;
+			decoder_clear(mpd_decoder);
+			seeking = 0;
+			decoder_command_finished(mpd_decoder);
+		}
+
+		if (seeking)
+			continue;
+
+		if (mp4ff_read_sample(mp4fh, track, sampleId, &mp4Buffer,
+				      &mp4BufferSize) == 0)
+			break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+		sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer,
+					     mp4BufferSize);
+#else
+		sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer);
+#endif
+
+		if (mp4Buffer)
+			free(mp4Buffer);
+		if (frameInfo.error > 0) {
+			ERROR("faad2 error: %s\n",
+			      faacDecGetErrorMessage(frameInfo.error));
+			break;
+		}
+
+		if (!initialized) {
+			channels = frameInfo.channels;
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+			scale = frameInfo.samplerate;
+#endif
+			audio_format.sample_rate = scale;
+			audio_format.channels = frameInfo.channels;
+			decoder_initialized(mpd_decoder, &audio_format,
+					    total_time);
+			initialized = true;
+		}
+
+		if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
+			dur = frameInfo.samples / channels;
+			offset = 0;
+		}
+
+		sampleCount = (unsigned long)(dur * channels);
+
+		if (sampleCount > 0) {
+			initial = 0;
+			bitRate = frameInfo.bytesconsumed * 8.0 *
+			    frameInfo.channels * scale /
+			    frameInfo.samples / 1000 + 0.5;
+		}
+
+		sampleBufferLen = sampleCount * 2;
+
+		sampleBuffer += offset * channels * 2;
+
+		decoder_data(mpd_decoder, inStream, 1, sampleBuffer,
+			     sampleBufferLen, file_time,
+			     bitRate, NULL);
+		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+			break;
+	}
+
+	free(seekTable);
+	faacDecClose(decoder);
+	mp4ff_close(mp4fh);
+	free(mp4cb);
+
+	if (!initialized)
+		return -1;
+
+	if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK && seeking) {
+		decoder_clear(mpd_decoder);
+		decoder_command_finished(mpd_decoder);
+	}
+	decoder_flush(mpd_decoder);
+
+	return 0;
+}
+
+static struct tag *mp4DataDup(char *file, int *mp4MetadataFound)
+{
+	struct tag *ret = NULL;
+	InputStream inStream;
+	mp4ff_t *mp4fh;
+	mp4ff_callback_t *callback;
+	int32_t track;
+	int32_t file_time;
+	int32_t scale;
+	int i;
+
+	*mp4MetadataFound = 0;
+
+	if (openInputStream(&inStream, file) < 0) {
+		DEBUG("mp4DataDup: Failed to open file: %s\n", file);
+		return NULL;
+	}
+
+	callback = xmalloc(sizeof(mp4ff_callback_t));
+	callback->read = mp4_inputStreamReadCallback;
+	callback->seek = mp4_inputStreamSeekCallback;
+	callback->user_data = &inStream;
+
+	mp4fh = mp4ff_open_read(callback);
+	if (!mp4fh) {
+		free(callback);
+		closeInputStream(&inStream);
+		return NULL;
+	}
+
+	track = mp4_getAACTrack(mp4fh);
+	if (track < 0) {
+		mp4ff_close(mp4fh);
+		closeInputStream(&inStream);
+		free(callback);
+		return NULL;
+	}
+
+	ret = tag_new();
+	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+	scale = mp4ff_time_scale(mp4fh, track);
+	if (scale < 0) {
+		mp4ff_close(mp4fh);
+		closeInputStream(&inStream);
+		free(callback);
+		tag_free(ret);
+		return NULL;
+	}
+	ret->time = ((float)file_time) / scale + 0.5;
+
+	for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
+		char *item;
+		char *value;
+
+		mp4ff_meta_get_by_index(mp4fh, i, &item, &value);
+
+		if (0 == strcasecmp("artist", item)) {
+			tag_add_item(ret, TAG_ITEM_ARTIST, value);
+			*mp4MetadataFound = 1;
+		} else if (0 == strcasecmp("title", item)) {
+			tag_add_item(ret, TAG_ITEM_TITLE, value);
+			*mp4MetadataFound = 1;
+		} else if (0 == strcasecmp("album", item)) {
+			tag_add_item(ret, TAG_ITEM_ALBUM, value);
+			*mp4MetadataFound = 1;
+		} else if (0 == strcasecmp("track", item)) {
+			tag_add_item(ret, TAG_ITEM_TRACK, value);
+			*mp4MetadataFound = 1;
+		} else if (0 == strcasecmp("disc", item)) {	/* Is that the correct id? */
+			tag_add_item(ret, TAG_ITEM_DISC, value);
+			*mp4MetadataFound = 1;
+		} else if (0 == strcasecmp("genre", item)) {
+			tag_add_item(ret, TAG_ITEM_GENRE, value);
+			*mp4MetadataFound = 1;
+		} else if (0 == strcasecmp("date", item)) {
+			tag_add_item(ret, TAG_ITEM_DATE, value);
+			*mp4MetadataFound = 1;
+		}
+
+		free(item);
+		free(value);
+	}
+
+	mp4ff_close(mp4fh);
+	closeInputStream(&inStream);
+
+	return ret;
+}
+
+static struct tag *mp4TagDup(char *file)
+{
+	struct tag *ret = NULL;
+	int mp4MetadataFound = 0;
+
+	ret = mp4DataDup(file, &mp4MetadataFound);
+	if (!ret)
+		return NULL;
+	if (!mp4MetadataFound) {
+		struct tag *temp = tag_id3_load(file);
+		if (temp) {
+			temp->time = ret->time;
+			tag_free(ret);
+			ret = temp;
+		}
+	}
+
+	return ret;
+}
+
+static const char *mp4_suffixes[] = { "m4a", "mp4", NULL };
+static const char *mp4_mimeTypes[] = { "audio/mp4", "audio/m4a", NULL };
+
+struct decoder_plugin mp4Plugin = {
+	.name = "mp4",
+	.stream_decode = mp4_decode,
+	.tag_dup = mp4TagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+	.suffixes = mp4_suffixes,
+	.mime_types = mp4_mimeTypes,
+};
diff --git a/src/decoder/mpc_plugin.c b/src/decoder/mpc_plugin.c
new file mode 100644
index 000000000..fb1b0b56c
--- /dev/null
+++ b/src/decoder/mpc_plugin.c
@@ -0,0 +1,308 @@
+/* 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 "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+
+#include <mpcdec/mpcdec.h>
+
+typedef struct _MpcCallbackData {
+	InputStream *inStream;
+	struct decoder *decoder;
+} MpcCallbackData;
+
+static mpc_int32_t mpc_read_cb(void *vdata, void *ptr, mpc_int32_t size)
+{
+	MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+	return decoder_read(data->decoder, data->inStream, ptr, size);
+}
+
+static mpc_bool_t mpc_seek_cb(void *vdata, mpc_int32_t offset)
+{
+	MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+	return seekInputStream(data->inStream, offset, SEEK_SET) < 0 ? 0 : 1;
+}
+
+static mpc_int32_t mpc_tell_cb(void *vdata)
+{
+	MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+	return (long)(data->inStream->offset);
+}
+
+static mpc_bool_t mpc_canseek_cb(void *vdata)
+{
+	MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+	return data->inStream->seekable;
+}
+
+static mpc_int32_t mpc_getsize_cb(void *vdata)
+{
+	MpcCallbackData *data = (MpcCallbackData *) vdata;
+
+	return data->inStream->size;
+}
+
+/* this _looks_ performance-critical, don't de-inline -- eric */
+static inline int16_t convertSample(MPC_SAMPLE_FORMAT sample)
+{
+	/* only doing 16-bit audio for now */
+	int32_t val;
+
+	const int clip_min = -1 << (16 - 1);
+	const int clip_max = (1 << (16 - 1)) - 1;
+
+#ifdef MPC_FIXED_POINT
+	const int shift = 16 - MPC_FIXED_POINT_SCALE_SHIFT;
+
+	if (sample > 0) {
+		sample <<= shift;
+	} else if (shift < 0) {
+		sample >>= -shift;
+	}
+	val = sample;
+#else
+	const int float_scale = 1 << (16 - 1);
+
+	val = sample * float_scale;
+#endif
+
+	if (val < clip_min)
+		val = clip_min;
+	else if (val > clip_max)
+		val = clip_max;
+
+	return val;
+}
+
+static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
+{
+	mpc_decoder decoder;
+	mpc_reader reader;
+	mpc_streaminfo info;
+	struct audio_format audio_format;
+
+	MpcCallbackData data;
+
+	MPC_SAMPLE_FORMAT sample_buffer[MPC_DECODER_BUFFER_LENGTH];
+
+	int eof = 0;
+	long ret;
+#define MPC_CHUNK_SIZE 4096
+	char chunk[MPC_CHUNK_SIZE];
+	int chunkpos = 0;
+	long bitRate = 0;
+	int16_t *s16 = (int16_t *) chunk;
+	unsigned long samplePos = 0;
+	mpc_uint32_t vbrUpdateAcc;
+	mpc_uint32_t vbrUpdateBits;
+	float total_time;
+	int i;
+	ReplayGainInfo *replayGainInfo = NULL;
+
+	data.inStream = inStream;
+	data.decoder = mpd_decoder;
+
+	reader.read = mpc_read_cb;
+	reader.seek = mpc_seek_cb;
+	reader.tell = mpc_tell_cb;
+	reader.get_size = mpc_getsize_cb;
+	reader.canseek = mpc_canseek_cb;
+	reader.data = &data;
+
+	mpc_streaminfo_init(&info);
+
+	if ((ret = mpc_streaminfo_read(&info, &reader)) != ERROR_CODE_OK) {
+		if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) {
+			ERROR("Not a valid musepack stream\n");
+			return -1;
+		}
+		return 0;
+	}
+
+	mpc_decoder_setup(&decoder, &reader);
+
+	if (!mpc_decoder_initialize(&decoder, &info)) {
+		if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) {
+			ERROR("Not a valid musepack stream\n");
+			return -1;
+		}
+		return 0;
+	}
+
+	audio_format.bits = 16;
+	audio_format.channels = info.channels;
+	audio_format.sample_rate = info.sample_freq;
+
+	replayGainInfo = newReplayGainInfo();
+	replayGainInfo->albumGain = info.gain_album * 0.01;
+	replayGainInfo->albumPeak = info.peak_album / 32767.0;
+	replayGainInfo->trackGain = info.gain_title * 0.01;
+	replayGainInfo->trackPeak = info.peak_title / 32767.0;
+
+	decoder_initialized(mpd_decoder, &audio_format,
+			    mpc_streaminfo_get_length(&info));
+
+	while (!eof) {
+		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+			samplePos = decoder_seek_where(mpd_decoder) *
+				audio_format.sample_rate;
+			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
+				decoder_clear(mpd_decoder);
+				s16 = (int16_t *) chunk;
+				chunkpos = 0;
+				decoder_command_finished(mpd_decoder);
+			} else
+				decoder_seek_error(mpd_decoder);
+		}
+
+		vbrUpdateAcc = 0;
+		vbrUpdateBits = 0;
+		ret = mpc_decoder_decode(&decoder, sample_buffer,
+					 &vbrUpdateAcc, &vbrUpdateBits);
+
+		if (ret <= 0 || decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) {
+			eof = 1;
+			break;
+		}
+
+		samplePos += ret;
+
+		/* ret is in samples, and we have stereo */
+		ret *= 2;
+
+		for (i = 0; i < ret; i++) {
+			/* 16 bit audio again */
+			*s16 = convertSample(sample_buffer[i]);
+			chunkpos += 2;
+			s16++;
+
+			if (chunkpos >= MPC_CHUNK_SIZE) {
+				total_time = ((float)samplePos) /
+				    audio_format.sample_rate;
+
+				bitRate = vbrUpdateBits *
+				    audio_format.sample_rate / 1152 / 1000;
+
+				decoder_data(mpd_decoder, inStream,
+					     inStream->seekable,
+					     chunk, chunkpos,
+					     total_time,
+					     bitRate, replayGainInfo);
+
+				chunkpos = 0;
+				s16 = (int16_t *) chunk;
+				if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) {
+					eof = 1;
+					break;
+				}
+			}
+		}
+	}
+
+	if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP &&
+	    chunkpos > 0) {
+		total_time = ((float)samplePos) / audio_format.sample_rate;
+
+		bitRate =
+		    vbrUpdateBits * audio_format.sample_rate / 1152 / 1000;
+
+		decoder_data(mpd_decoder, NULL, inStream->seekable,
+			     chunk, chunkpos, total_time, bitRate,
+			     replayGainInfo);
+	}
+
+	decoder_flush(mpd_decoder);
+
+	freeReplayGainInfo(replayGainInfo);
+
+	return 0;
+}
+
+static float mpcGetTime(char *file)
+{
+	InputStream inStream;
+	float total_time = -1;
+
+	mpc_reader reader;
+	mpc_streaminfo info;
+	MpcCallbackData data;
+
+	data.inStream = &inStream;
+	data.decoder = NULL;
+
+	reader.read = mpc_read_cb;
+	reader.seek = mpc_seek_cb;
+	reader.tell = mpc_tell_cb;
+	reader.get_size = mpc_getsize_cb;
+	reader.canseek = mpc_canseek_cb;
+	reader.data = &data;
+
+	mpc_streaminfo_init(&info);
+
+	if (openInputStream(&inStream, file) < 0) {
+		DEBUG("mpcGetTime: Failed to open file: %s\n", file);
+		return -1;
+	}
+
+	if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) {
+		closeInputStream(&inStream);
+		return -1;
+	}
+
+	total_time = mpc_streaminfo_get_length(&info);
+
+	closeInputStream(&inStream);
+
+	return total_time;
+}
+
+static struct tag *mpcTagDup(char *file)
+{
+	struct tag *ret = NULL;
+	float total_time = mpcGetTime(file);
+
+	if (total_time < 0) {
+		DEBUG("mpcTagDup: Failed to get Songlength of file: %s\n",
+		      file);
+		return NULL;
+	}
+
+	ret = tag_ape_load(file);
+	if (!ret)
+		ret = tag_id3_load(file);
+	if (!ret)
+		ret = tag_new();
+	ret->time = total_time;
+
+	return ret;
+}
+
+static const char *mpcSuffixes[] = { "mpc", NULL };
+
+struct decoder_plugin mpcPlugin = {
+	.name = "mpc",
+	.stream_decode = mpc_decode,
+	.tag_dup = mpcTagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = mpcSuffixes,
+};
diff --git a/src/decoder/oggflac_plugin.c b/src/decoder/oggflac_plugin.c
new file mode 100644
index 000000000..091b00988
--- /dev/null
+++ b/src/decoder/oggflac_plugin.c
@@ -0,0 +1,355 @@
+/* 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
+ *
+ * OggFLAC support (half-stolen from flac_plugin.c :))
+ * (c) 2005 by Eric Wong <normalperson@yhbt.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 "_flac_common.h"
+#include "_ogg_common.h"
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <OggFLAC/seekable_stream_decoder.h>
+
+static void oggflac_cleanup(FlacData * data,
+			    OggFLAC__SeekableStreamDecoder * decoder)
+{
+	if (data->replayGainInfo)
+		freeReplayGainInfo(data->replayGainInfo);
+	if (decoder)
+		OggFLAC__seekable_stream_decoder_delete(decoder);
+}
+
+static OggFLAC__SeekableStreamDecoderReadStatus of_read_cb(mpd_unused const
+							   OggFLAC__SeekableStreamDecoder
+							   * decoder,
+							   FLAC__byte buf[],
+							   unsigned *bytes,
+							   void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+	size_t r;
+
+	r = decoder_read(data->decoder, data->inStream, (void *)buf, *bytes);
+	*bytes = r;
+
+	if (r == 0 && !inputStreamAtEOF(data->inStream) &&
+	    decoder_get_command(data->decoder) == DECODE_COMMAND_NONE)
+		return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR;
+
+	return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK;
+}
+
+static OggFLAC__SeekableStreamDecoderSeekStatus of_seek_cb(mpd_unused const
+							   OggFLAC__SeekableStreamDecoder
+							   * decoder,
+							   FLAC__uint64 offset,
+							   void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
+		return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR;
+	}
+
+	return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK;
+}
+
+static OggFLAC__SeekableStreamDecoderTellStatus of_tell_cb(mpd_unused const
+							   OggFLAC__SeekableStreamDecoder
+							   * decoder,
+							   FLAC__uint64 *
+							   offset, void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	*offset = (long)(data->inStream->offset);
+
+	return OggFLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK;
+}
+
+static OggFLAC__SeekableStreamDecoderLengthStatus of_length_cb(mpd_unused const
+							       OggFLAC__SeekableStreamDecoder
+							       * decoder,
+							       FLAC__uint64 *
+							       length,
+							       void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	*length = (size_t) (data->inStream->size);
+
+	return OggFLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK;
+}
+
+static FLAC__bool of_EOF_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * decoder,
+			    void *fdata)
+{
+	FlacData *data = (FlacData *) fdata;
+
+	return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE &&
+		decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) ||
+		inputStreamAtEOF(data->inStream);
+}
+
+static void of_error_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * decoder,
+			FLAC__StreamDecoderErrorStatus status, void *fdata)
+{
+	flac_error_common_cb("oggflac", status, (FlacData *) fdata);
+}
+
+static void oggflacPrintErroredState(OggFLAC__SeekableStreamDecoderState state)
+{
+	switch (state) {
+	case OggFLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
+		ERROR("oggflac allocation error\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_READ_ERROR:
+		ERROR("oggflac read error\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR:
+		ERROR("oggflac seek error\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR:
+		ERROR("oggflac seekable stream error\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED:
+		ERROR("oggflac decoder already initialized\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK:
+		ERROR("invalid oggflac callback\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED:
+		ERROR("oggflac decoder uninitialized\n");
+		break;
+	case OggFLAC__SEEKABLE_STREAM_DECODER_OK:
+	case OggFLAC__SEEKABLE_STREAM_DECODER_SEEKING:
+	case OggFLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM:
+		break;
+	}
+}
+
+static FLAC__StreamDecoderWriteStatus oggflacWrite(mpd_unused const
+						   OggFLAC__SeekableStreamDecoder
+						   * decoder,
+						   const FLAC__Frame * frame,
+						   const FLAC__int32 *
+						   const buf[], void *vdata)
+{
+	FlacData *data = (FlacData *) vdata;
+	FLAC__uint32 samples = frame->header.blocksize;
+	float timeChange;
+
+	timeChange = ((float)samples) / frame->header.sample_rate;
+	data->time += timeChange;
+
+	return flac_common_write(data, frame, buf);
+}
+
+/* used by TagDup */
+static void of_metadata_dup_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * decoder,
+			       const FLAC__StreamMetadata * block, void *vdata)
+{
+	FlacData *data = (FlacData *) vdata;
+
+	switch (block->type) {
+	case FLAC__METADATA_TYPE_STREAMINFO:
+		if (!data->tag)
+			data->tag = tag_new();
+		data->tag->time = ((float)block->data.stream_info.
+				   total_samples) /
+		    block->data.stream_info.sample_rate + 0.5;
+		return;
+	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+		copyVorbisCommentBlockToMpdTag(block, data->tag);
+	default:
+		break;
+	}
+}
+
+/* used by decode */
+static void of_metadata_decode_cb(mpd_unused const OggFLAC__SeekableStreamDecoder * dec,
+				  const FLAC__StreamMetadata * block,
+				  void *vdata)
+{
+	flac_metadata_common_cb(block, (FlacData *) vdata);
+}
+
+static OggFLAC__SeekableStreamDecoder
+    * full_decoder_init_and_read_metadata(FlacData * data,
+					  unsigned int metadata_only)
+{
+	OggFLAC__SeekableStreamDecoder *decoder = NULL;
+	unsigned int s = 1;
+
+	if (!(decoder = OggFLAC__seekable_stream_decoder_new()))
+		return NULL;
+
+	if (metadata_only) {
+		s &= OggFLAC__seekable_stream_decoder_set_metadata_callback
+		    (decoder, of_metadata_dup_cb);
+		s &= OggFLAC__seekable_stream_decoder_set_metadata_respond
+		    (decoder, FLAC__METADATA_TYPE_STREAMINFO);
+	} else {
+		s &= OggFLAC__seekable_stream_decoder_set_metadata_callback
+		    (decoder, of_metadata_decode_cb);
+	}
+
+	s &= OggFLAC__seekable_stream_decoder_set_read_callback(decoder,
+								of_read_cb);
+	s &= OggFLAC__seekable_stream_decoder_set_seek_callback(decoder,
+								of_seek_cb);
+	s &= OggFLAC__seekable_stream_decoder_set_tell_callback(decoder,
+								of_tell_cb);
+	s &= OggFLAC__seekable_stream_decoder_set_length_callback(decoder,
+								  of_length_cb);
+	s &= OggFLAC__seekable_stream_decoder_set_eof_callback(decoder,
+							       of_EOF_cb);
+	s &= OggFLAC__seekable_stream_decoder_set_write_callback(decoder,
+								 oggflacWrite);
+	s &= OggFLAC__seekable_stream_decoder_set_metadata_respond(decoder,
+								   FLAC__METADATA_TYPE_VORBIS_COMMENT);
+	s &= OggFLAC__seekable_stream_decoder_set_error_callback(decoder,
+								 of_error_cb);
+	s &= OggFLAC__seekable_stream_decoder_set_client_data(decoder,
+							      (void *)data);
+
+	if (!s) {
+		ERROR("oggflac problem before init()\n");
+		goto fail;
+	}
+	if (OggFLAC__seekable_stream_decoder_init(decoder) !=
+	    OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
+		ERROR("oggflac problem doing init()\n");
+		goto fail;
+	}
+	if (!OggFLAC__seekable_stream_decoder_process_until_end_of_metadata
+	    (decoder)) {
+		ERROR("oggflac problem reading metadata\n");
+		goto fail;
+	}
+
+	return decoder;
+
+fail:
+	oggflacPrintErroredState(OggFLAC__seekable_stream_decoder_get_state
+				 (decoder));
+	OggFLAC__seekable_stream_decoder_delete(decoder);
+	return NULL;
+}
+
+/* public functions: */
+static struct tag *oggflac_TagDup(char *file)
+{
+	InputStream inStream;
+	OggFLAC__SeekableStreamDecoder *decoder;
+	FlacData data;
+
+	if (openInputStream(&inStream, file) < 0)
+		return NULL;
+	if (ogg_stream_type_detect(&inStream) != FLAC) {
+		closeInputStream(&inStream);
+		return NULL;
+	}
+
+	init_FlacData(&data, NULL, &inStream);
+
+	/* errors here won't matter,
+	 * data.tag will be set or unset, that's all we care about */
+	decoder = full_decoder_init_and_read_metadata(&data, 1);
+
+	oggflac_cleanup(&data, decoder);
+	closeInputStream(&inStream);
+
+	return data.tag;
+}
+
+static bool oggflac_try_decode(InputStream * inStream)
+{
+	if (!inStream->seekable)
+		/* we cannot seek after the detection, so don't bother
+		   checking */
+		return true;
+
+	return ogg_stream_type_detect(inStream) == FLAC;
+}
+
+static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
+{
+	OggFLAC__SeekableStreamDecoder *decoder = NULL;
+	FlacData data;
+	int ret = 0;
+
+	init_FlacData(&data, mpd_decoder, inStream);
+
+	if (!(decoder = full_decoder_init_and_read_metadata(&data, 0))) {
+		ret = -1;
+		goto fail;
+	}
+
+	decoder_initialized(mpd_decoder, &data.audio_format, data.total_time);
+
+	while (1) {
+		OggFLAC__seekable_stream_decoder_process_single(decoder);
+		if (OggFLAC__seekable_stream_decoder_get_state(decoder) !=
+		    OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
+			break;
+		}
+		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+			FLAC__uint64 sampleToSeek = decoder_seek_where(mpd_decoder) *
+			    data.audio_format.sample_rate + 0.5;
+			if (OggFLAC__seekable_stream_decoder_seek_absolute
+			    (decoder, sampleToSeek)) {
+				decoder_clear(mpd_decoder);
+				data.time = ((float)sampleToSeek) /
+				    data.audio_format.sample_rate;
+				data.position = 0;
+				decoder_command_finished(mpd_decoder);
+			} else
+				decoder_seek_error(mpd_decoder);
+		}
+	}
+
+	if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+		oggflacPrintErroredState
+		    (OggFLAC__seekable_stream_decoder_get_state(decoder));
+		OggFLAC__seekable_stream_decoder_finish(decoder);
+	}
+
+fail:
+	oggflac_cleanup(&data, decoder);
+
+	return ret;
+}
+
+static const char *oggflac_Suffixes[] = { "ogg", "oga",NULL };
+static const char *oggflac_mime_types[] = { "audio/x-flac+ogg",
+					    "application/ogg",
+					    "application/x-ogg",
+					    NULL };
+
+struct decoder_plugin oggflacPlugin = {
+	.name = "oggflac",
+	.try_decode = oggflac_try_decode,
+	.stream_decode = oggflac_decode,
+	.tag_dup = oggflac_TagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = oggflac_Suffixes,
+	.mime_types = oggflac_mime_types
+};
diff --git a/src/decoder/oggvorbis_plugin.c b/src/decoder/oggvorbis_plugin.c
new file mode 100644
index 000000000..0eecb783f
--- /dev/null
+++ b/src/decoder/oggvorbis_plugin.c
@@ -0,0 +1,387 @@
+/* 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
+ */
+
+/* TODO 'ogg' should probably be replaced with 'oggvorbis' in all instances */
+
+#include "_ogg_common.h"
+#include "../utils.h"
+#include "../log.h"
+
+#ifndef HAVE_TREMOR
+#include <vorbis/vorbisfile.h>
+#else
+#include <tremor/ivorbisfile.h>
+/* Macros to make Tremor's API look like libogg. Tremor always
+   returns host-byte-order 16-bit signed data, and uses integer
+   milliseconds where libogg uses double seconds. 
+*/
+#define ov_read(VF, BUFFER, LENGTH, BIGENDIANP, WORD, SGNED, BITSTREAM) \
+        ov_read(VF, BUFFER, LENGTH, BITSTREAM)
+#define ov_time_total(VF, I) ((double)ov_time_total(VF, I)/1000)
+#define ov_time_tell(VF) ((double)ov_time_tell(VF)/1000)
+#define ov_time_seek_page(VF, S) (ov_time_seek_page(VF, (S)*1000))
+#endif /* HAVE_TREMOR */
+
+#ifdef WORDS_BIGENDIAN
+#define OGG_DECODE_USE_BIGENDIAN	1
+#else
+#define OGG_DECODE_USE_BIGENDIAN	0
+#endif
+
+typedef struct _OggCallbackData {
+	InputStream *inStream;
+	struct decoder *decoder;
+} OggCallbackData;
+
+static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
+{
+	size_t ret;
+	OggCallbackData *data = (OggCallbackData *) vdata;
+
+	ret = decoder_read(data->decoder, data->inStream, ptr, size * nmemb);
+
+	errno = 0;
+	/*if(ret<0) errno = ((InputStream *)inStream)->error; */
+
+	return ret / size;
+}
+
+static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence)
+{
+	const OggCallbackData *data = (const OggCallbackData *) vdata;
+	if(decoder_get_command(data->decoder) == DECODE_COMMAND_STOP)
+		return -1;
+	return seekInputStream(data->inStream, offset, whence);
+}
+
+/* TODO: check Ogg libraries API and see if we can just not have this func */
+static int ogg_close_cb(mpd_unused void *vdata)
+{
+	return 0;
+}
+
+static long ogg_tell_cb(void *vdata)
+{
+	const OggCallbackData *data = (const OggCallbackData *) vdata;
+
+	return (long)(data->inStream->offset);
+}
+
+static const char *ogg_parseComment(const char *comment, const char *needle)
+{
+	int len = strlen(needle);
+
+	if (strncasecmp(comment, needle, len) == 0 && *(comment + len) == '=') {
+		return comment + len + 1;
+	}
+
+	return NULL;
+}
+
+static void ogg_getReplayGainInfo(char **comments, ReplayGainInfo ** infoPtr)
+{
+	const char *temp;
+	int found = 0;
+
+	if (*infoPtr)
+		freeReplayGainInfo(*infoPtr);
+	*infoPtr = newReplayGainInfo();
+
+	while (*comments) {
+		if ((temp =
+		     ogg_parseComment(*comments, "replaygain_track_gain"))) {
+			(*infoPtr)->trackGain = atof(temp);
+			found = 1;
+		} else if ((temp = ogg_parseComment(*comments,
+						    "replaygain_album_gain"))) {
+			(*infoPtr)->albumGain = atof(temp);
+			found = 1;
+		} else if ((temp = ogg_parseComment(*comments,
+						    "replaygain_track_peak"))) {
+			(*infoPtr)->trackPeak = atof(temp);
+			found = 1;
+		} else if ((temp = ogg_parseComment(*comments,
+						    "replaygain_album_peak"))) {
+			(*infoPtr)->albumPeak = atof(temp);
+			found = 1;
+		}
+
+		comments++;
+	}
+
+	if (!found) {
+		freeReplayGainInfo(*infoPtr);
+		*infoPtr = NULL;
+	}
+}
+
+static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber";
+static const char *VORBIS_COMMENT_DISC_KEY = "discnumber";
+
+static unsigned int ogg_parseCommentAddToTag(char *comment,
+					     unsigned int itemType,
+					     struct tag ** tag)
+{
+	const char *needle;
+	unsigned int len;
+	switch (itemType) {
+	case TAG_ITEM_TRACK:
+		needle = VORBIS_COMMENT_TRACK_KEY;
+		break;
+	case TAG_ITEM_DISC:
+		needle = VORBIS_COMMENT_DISC_KEY;
+		break;
+	default:
+		needle = mpdTagItemKeys[itemType];
+	}
+	len = strlen(needle);
+
+	if (strncasecmp(comment, needle, len) == 0 && *(comment + len) == '=') {
+		if (!*tag)
+			*tag = tag_new();
+
+		tag_add_item(*tag, itemType, comment + len + 1);
+
+		return 1;
+	}
+
+	return 0;
+}
+
+static struct tag *oggCommentsParse(char **comments)
+{
+	struct tag *tag = NULL;
+
+	while (*comments) {
+		int j;
+		for (j = TAG_NUM_OF_ITEM_TYPES; --j >= 0;) {
+			if (ogg_parseCommentAddToTag(*comments, j, &tag))
+				break;
+		}
+		comments++;
+	}
+
+	return tag;
+}
+
+static void putOggCommentsIntoOutputBuffer(char *streamName,
+					   char **comments)
+{
+	struct tag *tag;
+
+	tag = oggCommentsParse(comments);
+	if (!tag && streamName) {
+		tag = tag_new();
+	}
+	if (!tag)
+		return;
+
+	if (streamName) {
+		tag_clear_items_by_type(tag, TAG_ITEM_NAME);
+		tag_add_item(tag, TAG_ITEM_NAME, streamName);
+	}
+
+	tag_free(tag);
+}
+
+/* public */
+static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
+{
+	OggVorbis_File vf;
+	ov_callbacks callbacks;
+	OggCallbackData data;
+	struct audio_format audio_format;
+	int current_section;
+	int prev_section = -1;
+	long ret;
+#define OGG_CHUNK_SIZE 4096
+	char chunk[OGG_CHUNK_SIZE];
+	int chunkpos = 0;
+	long bitRate = 0;
+	long test;
+	ReplayGainInfo *replayGainInfo = NULL;
+	char **comments;
+	const char *errorStr;
+	int initialized = 0;
+
+	data.inStream = inStream;
+	data.decoder = decoder;
+
+	callbacks.read_func = ogg_read_cb;
+	callbacks.seek_func = ogg_seek_cb;
+	callbacks.close_func = ogg_close_cb;
+	callbacks.tell_func = ogg_tell_cb;
+	if ((ret = ov_open_callbacks(&data, &vf, NULL, 0, callbacks)) < 0) {
+		if (decoder_get_command(decoder) != DECODE_COMMAND_NONE)
+			return 0;
+
+		switch (ret) {
+		case OV_EREAD:
+			errorStr = "read error";
+			break;
+		case OV_ENOTVORBIS:
+			errorStr = "not vorbis stream";
+			break;
+		case OV_EVERSION:
+			errorStr = "vorbis version mismatch";
+			break;
+		case OV_EBADHEADER:
+			errorStr = "invalid vorbis header";
+			break;
+		case OV_EFAULT:
+			errorStr = "internal logic error";
+			break;
+		default:
+			errorStr = "unknown error";
+			break;
+		}
+		ERROR("Error decoding Ogg Vorbis stream: %s\n",
+		      errorStr);
+		return -1;
+	}
+	audio_format.bits = 16;
+
+	while (1) {
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+			double seek_where = decoder_seek_where(decoder);
+			if (0 == ov_time_seek_page(&vf, seek_where)) {
+				decoder_clear(decoder);
+				chunkpos = 0;
+				decoder_command_finished(decoder);
+			} else
+				decoder_seek_error(decoder);
+		}
+		ret = ov_read(&vf, chunk + chunkpos,
+			      OGG_CHUNK_SIZE - chunkpos,
+			      OGG_DECODE_USE_BIGENDIAN, 2, 1, &current_section);
+		if (current_section != prev_section) {
+			/*printf("new song!\n"); */
+			vorbis_info *vi = ov_info(&vf, -1);
+			audio_format.channels = vi->channels;
+			audio_format.sample_rate = vi->rate;
+			if (!initialized) {
+				float total_time = ov_time_total(&vf, -1);
+				if (total_time < 0)
+					total_time = 0;
+				decoder_initialized(decoder, &audio_format,
+						    total_time);
+				initialized = 1;
+			}
+			comments = ov_comment(&vf, -1)->user_comments;
+			putOggCommentsIntoOutputBuffer(inStream->metaName,
+						       comments);
+			ogg_getReplayGainInfo(comments, &replayGainInfo);
+		}
+
+		prev_section = current_section;
+
+		if (ret <= 0) {
+			if (ret == OV_HOLE) /* bad packet */
+				ret = 0;
+			else /* break on EOF or other error */
+				break;
+		}
+
+		chunkpos += ret;
+
+		if (chunkpos >= OGG_CHUNK_SIZE) {
+			if ((test = ov_bitrate_instant(&vf)) > 0) {
+				bitRate = test / 1000;
+			}
+			decoder_data(decoder, inStream,
+				     inStream->seekable,
+				     chunk, chunkpos,
+				     ov_pcm_tell(&vf) / audio_format.sample_rate,
+				     bitRate, replayGainInfo);
+			chunkpos = 0;
+			if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
+				break;
+		}
+	}
+
+	if (decoder_get_command(decoder) == DECODE_COMMAND_NONE &&
+	    chunkpos > 0) {
+		decoder_data(decoder, NULL, inStream->seekable,
+			     chunk, chunkpos,
+			     ov_time_tell(&vf), bitRate,
+			     replayGainInfo);
+	}
+
+	if (replayGainInfo)
+		freeReplayGainInfo(replayGainInfo);
+
+	ov_clear(&vf);
+
+	decoder_flush(decoder);
+
+	return 0;
+}
+
+static struct tag *oggvorbis_TagDup(char *file)
+{
+	struct tag *ret;
+	FILE *fp;
+	OggVorbis_File vf;
+
+	fp = fopen(file, "r");
+	if (!fp) {
+		DEBUG("oggvorbis_TagDup: Failed to open file: '%s', %s\n",
+		      file, strerror(errno));
+		return NULL;
+	}
+	if (ov_open(fp, &vf, NULL, 0) < 0) {
+		fclose(fp);
+		return NULL;
+	}
+
+	ret = oggCommentsParse(ov_comment(&vf, -1)->user_comments);
+
+	if (!ret)
+		ret = tag_new();
+	ret->time = (int)(ov_time_total(&vf, -1) + 0.5);
+
+	ov_clear(&vf);
+
+	return ret;
+}
+
+static bool oggvorbis_try_decode(InputStream * inStream)
+{
+	if (!inStream->seekable)
+		/* we cannot seek after the detection, so don't bother
+		   checking */
+		return true;
+
+	return ogg_stream_type_detect(inStream) == VORBIS;
+}
+
+static const char *oggvorbis_Suffixes[] = { "ogg","oga", NULL };
+static const char *oggvorbis_MimeTypes[] = { "application/ogg",
+					     "audio/x-vorbis+ogg",
+					     "application/x-ogg",
+					     NULL };
+
+struct decoder_plugin oggvorbisPlugin = {
+	.name = "oggvorbis",
+	.try_decode = oggvorbis_try_decode,
+	.stream_decode = oggvorbis_decode,
+	.tag_dup = oggvorbis_TagDup,
+	.stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+	.suffixes = oggvorbis_Suffixes,
+	.mime_types = oggvorbis_MimeTypes
+};
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
new file mode 100644
index 000000000..14b7e5f69
--- /dev/null
+++ b/src/decoder/wavpack_plugin.c
@@ -0,0 +1,574 @@
+/* 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
+ * 
+ * WavPack support added by Laszlo Ashin <kodest@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 "../decoder_api.h"
+#include "../utils.h"
+#include "../log.h"
+#include "../path.h"
+
+#include <wavpack/wavpack.h>
+
+/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
+#define CHUNK_SIZE		1020
+
+#define ERRORLEN 80
+
+static struct {
+	const char *name;
+	int type;
+} tagtypes[] = {
+	{ "artist",     TAG_ITEM_ARTIST },
+	{ "album",      TAG_ITEM_ALBUM },
+	{ "title",      TAG_ITEM_TITLE },
+	{ "track",      TAG_ITEM_TRACK },
+	{ "name",       TAG_ITEM_NAME },
+	{ "genre",      TAG_ITEM_GENRE },
+	{ "date",       TAG_ITEM_DATE },
+	{ "composer",   TAG_ITEM_COMPOSER },
+	{ "performer",  TAG_ITEM_PERFORMER },
+	{ "comment",    TAG_ITEM_COMMENT },
+	{ "disc",       TAG_ITEM_DISC },
+	{ NULL,         0 }
+};
+
+/*
+ * This function has been borrowed from the tiny player found on
+ * wavpack.com. Modifications were required because mpd only handles
+ * max 16 bit samples.
+ */
+static void format_samples_int(int Bps, void *buffer, uint32_t samcnt)
+{
+	int32_t temp;
+	uchar *dst = (uchar *)buffer;
+	int32_t *src = (int32_t *)buffer;
+
+	switch (Bps) {
+		case 1:
+			while (samcnt--)
+				*dst++ = *src++;
+			break;
+		case 2:
+			while (samcnt--) {
+				temp = *src++;
+#ifdef WORDS_BIGENDIAN
+				*dst++ = (uchar)(temp >> 8);
+				*dst++ = (uchar)(temp); 
+#else
+				*dst++ = (uchar)(temp); 
+				*dst++ = (uchar)(temp >> 8);
+#endif
+			}
+			break;
+		case 3:
+			/* downscale to 16 bits */
+			while (samcnt--) {
+				temp = *src++;
+#ifdef WORDS_BIGENDIAN
+				*dst++ = (uchar)(temp >> 16);
+				*dst++ = (uchar)(temp >> 8);
+#else
+				*dst++ = (uchar)(temp >> 8);
+				*dst++ = (uchar)(temp >> 16);
+#endif
+			}
+			break;
+		case 4:
+			/* downscale to 16 bits */
+			while (samcnt--) {
+				temp = *src++;
+#ifdef WORDS_BIGENDIAN
+				*dst++ = (uchar)(temp >> 24);
+				*dst++ = (uchar)(temp >> 16);
+
+#else
+				*dst++ = (uchar)(temp >> 16);
+				*dst++ = (uchar)(temp >> 24);
+#endif
+			}
+			break;
+	}
+}
+
+/*
+ * This function converts floating point sample data to 16 bit integer.
+ */
+static void format_samples_float(mpd_unused int Bps, void *buffer,
+				 uint32_t samcnt)
+{
+	int16_t *dst = (int16_t *)buffer;
+	float *src = (float *)buffer;
+
+	while (samcnt--) {
+		*dst++ = (int16_t)(*src++);
+	}
+}
+
+/*
+ * This does the main decoding thing.
+ * Requires an already opened WavpackContext.
+ */
+static void wavpack_decode(struct decoder * decoder,
+                           WavpackContext *wpc, int canseek,
+                           ReplayGainInfo *replayGainInfo)
+{
+	struct audio_format audio_format;
+	void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
+	char chunk[CHUNK_SIZE];
+	float file_time;
+	int samplesreq, samplesgot;
+	int allsamples;
+	int position, outsamplesize;
+	int Bps;
+
+	audio_format.sample_rate = WavpackGetSampleRate(wpc);
+	audio_format.channels = WavpackGetReducedChannels(wpc);
+	audio_format.bits = WavpackGetBitsPerSample(wpc);
+
+	if (audio_format.bits > 16)
+		audio_format.bits = 16;
+
+	if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
+		format_samples = format_samples_float;
+	else
+		format_samples = format_samples_int;
+/*
+	if ((WavpackGetMode(wpc) & MODE_WVC) == MODE_WVC)
+		ERROR("decoding WITH wvc!!!\n");
+	else
+		ERROR("decoding without wvc\n");
+*/
+	allsamples = WavpackGetNumSamples(wpc);
+	Bps = WavpackGetBytesPerSample(wpc);
+
+	outsamplesize = Bps;
+	if (outsamplesize > 2)
+		outsamplesize = 2;
+	outsamplesize *= audio_format.channels;
+
+	samplesreq = sizeof(chunk) / (4 * audio_format.channels);
+
+	decoder_initialized(decoder, &audio_format,
+			    (float)allsamples / audio_format.sample_rate);
+
+	position = 0;
+
+	do {
+		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+			if (canseek) {
+				int where;
+
+				decoder_clear(decoder);
+
+				where = decoder_seek_where(decoder) *
+					audio_format.sample_rate;
+				if (WavpackSeekSample(wpc, where)) {
+					position = where;
+					decoder_command_finished(decoder);
+				} else
+					decoder_seek_error(decoder);
+			} else {
+				decoder_seek_error(decoder);
+			}
+		}
+
+		if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
+			break;
+
+		samplesgot = WavpackUnpackSamples(wpc,
+		                                  (int32_t *)chunk, samplesreq);
+		if (samplesgot > 0) {
+			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
+			              1000 + 0.5);
+			position += samplesgot;
+			file_time = (float)position / audio_format.sample_rate;
+
+			format_samples(Bps, chunk,
+			               samplesgot * audio_format.channels);
+
+			decoder_data(decoder, NULL, 0, chunk,
+				     samplesgot * outsamplesize,
+				     file_time, bitrate,
+				     replayGainInfo);
+		}
+	} while (samplesgot == samplesreq);
+
+	decoder_flush(decoder);
+}
+
+static char *wavpack_tag(WavpackContext *wpc, char *key)
+{
+	char *value = NULL;
+	int size;
+
+	size = WavpackGetTagItem(wpc, key, NULL, 0);
+	if (size > 0) {
+		size++;
+		value = xmalloc(size);
+		if (!value)
+			return NULL;
+		WavpackGetTagItem(wpc, key, value, size);
+	}
+
+	return value;
+}
+
+static ReplayGainInfo *wavpack_replaygain(WavpackContext *wpc)
+{
+	static char replaygain_track_gain[] = "replaygain_track_gain";
+	static char replaygain_album_gain[] = "replaygain_album_gain";
+	static char replaygain_track_peak[] = "replaygain_track_peak";
+	static char replaygain_album_peak[] = "replaygain_album_peak";
+	ReplayGainInfo *replayGainInfo;
+	int found = 0;
+	char *value;
+
+	replayGainInfo = newReplayGainInfo();
+
+	value = wavpack_tag(wpc, replaygain_track_gain);
+	if (value) {
+		replayGainInfo->trackGain = atof(value);
+		free(value);
+		found = 1;
+	}
+
+	value = wavpack_tag(wpc, replaygain_album_gain);
+	if (value) {
+		replayGainInfo->albumGain = atof(value);
+		free(value);
+		found = 1;
+	}
+
+	value = wavpack_tag(wpc, replaygain_track_peak);
+	if (value) {
+		replayGainInfo->trackPeak = atof(value);
+		free(value);
+		found = 1;
+	}
+
+	value = wavpack_tag(wpc, replaygain_album_peak);
+	if (value) {
+		replayGainInfo->albumPeak = atof(value);
+		free(value);
+		found = 1;
+	}
+
+
+	if (found)
+		return replayGainInfo;
+
+	freeReplayGainInfo(replayGainInfo);
+
+	return NULL;
+}
+
+/*
+ * Reads metainfo from the specified file.
+ */
+static struct tag *wavpack_tagdup(char *fname)
+{
+	WavpackContext *wpc;
+	struct tag *tag;
+	char error[ERRORLEN];
+	char *s;
+	int ssize;
+	int i, j;
+
+	wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0);
+	if (wpc == NULL) {
+		ERROR("failed to open WavPack file \"%s\": %s\n", fname, error);
+		return NULL;
+	}
+
+	tag = tag_new();
+	tag->time =
+		(float)WavpackGetNumSamples(wpc) / WavpackGetSampleRate(wpc);
+
+	ssize = 0;
+	s = NULL;
+
+	for (i = 0; tagtypes[i].name != NULL; ++i) {
+		j = WavpackGetTagItem(wpc, tagtypes[i].name, NULL, 0);
+		if (j > 0) {
+			++j;
+
+			if (s == NULL) {
+				s = xmalloc(j);
+				if (s == NULL) break;
+				ssize = j;
+			} else if (j > ssize) {
+				char *t = (char *)xrealloc(s, j);
+				if (t == NULL) break;
+				ssize = j;
+				s = t;
+			}
+
+			WavpackGetTagItem(wpc, tagtypes[i].name, s, j);
+			tag_add_item(tag, tagtypes[i].type, s);
+		}
+	}
+
+	if (s != NULL)
+		free(s);
+
+	WavpackCloseFile(wpc);
+
+	return tag;
+}
+
+/*
+ * mpd InputStream <=> WavpackStreamReader wrapper callbacks
+ */
+
+/* This struct is needed for per-stream last_byte storage. */
+typedef struct {
+	struct decoder *decoder;
+	InputStream *is;
+	/* Needed for push_back_byte() */
+	int last_byte;
+} InputStreamPlus;
+
+static int32_t read_bytes(void *id, void *data, int32_t bcount)
+{
+	InputStreamPlus *isp = (InputStreamPlus *)id;
+	uint8_t *buf = (uint8_t *)data;
+	int32_t i = 0;
+
+	if (isp->last_byte != EOF) {
+		*buf++ = isp->last_byte;
+		isp->last_byte = EOF;
+		--bcount;
+		++i;
+	}
+	return i + decoder_read(isp->decoder, isp->is, buf, bcount);
+}
+
+static uint32_t get_pos(void *id)
+{
+	return ((InputStreamPlus *)id)->is->offset;
+}
+
+static int set_pos_abs(void *id, uint32_t pos)
+{
+	return seekInputStream(((InputStreamPlus *)id)->is, pos, SEEK_SET);
+}
+
+static int set_pos_rel(void *id, int32_t delta, int mode)
+{
+	return seekInputStream(((InputStreamPlus *)id)->is, delta, mode);
+}
+
+static int push_back_byte(void *id, int c)
+{
+	((InputStreamPlus *)id)->last_byte = c;
+	return 1;
+}
+
+static uint32_t get_length(void *id)
+{
+	return ((InputStreamPlus *)id)->is->size;
+}
+
+static int can_seek(void *id)
+{
+	return ((InputStreamPlus *)id)->is->seekable;
+}
+
+static WavpackStreamReader mpd_is_reader = {
+	.read_bytes = read_bytes,
+	.get_pos = get_pos,
+	.set_pos_abs = set_pos_abs,
+	.set_pos_rel = set_pos_rel,
+	.push_back_byte = push_back_byte,
+	.get_length = get_length,
+	.can_seek = can_seek,
+	.write_bytes = NULL /* no need to write edited tags */
+};
+
+static void
+initInputStreamPlus(InputStreamPlus *isp, struct decoder *decoder,
+		    InputStream *is)
+{
+	isp->decoder = decoder;
+	isp->is = is;
+	isp->last_byte = EOF;
+}
+
+/*
+ * Tries to decode the specified stream, and gives true if managed to do it.
+ */
+static bool wavpack_trydecode(InputStream *is)
+{
+	char error[ERRORLEN];
+	WavpackContext *wpc;
+	InputStreamPlus isp;
+
+	initInputStreamPlus(&isp, NULL, is);
+	wpc = WavpackOpenFileInputEx(&mpd_is_reader, &isp, NULL, error,
+	                             OPEN_STREAMING, 0);
+	if (wpc == NULL)
+		return false;
+
+	WavpackCloseFile(wpc);
+	/* Seek it back in order to play from the first byte. */
+	seekInputStream(is, 0, SEEK_SET);
+
+	return true;
+}
+
+static int wavpack_open_wvc(struct decoder *decoder,
+			    InputStream *is_wvc)
+{
+	char tmp[MPD_PATH_MAX];
+	const char *utf8url;
+	size_t len;
+	char *wvc_url = NULL;
+	int ret;
+
+	/*
+	 * As we use dc->utf8url, this function will be bad for
+	 * single files. utf8url is not absolute file path :/
+	 */
+	utf8url = decoder_get_url(decoder, tmp);
+	if (utf8url == NULL)
+		return 0;
+
+	len = strlen(utf8url);
+	if (!len)
+		return 0;
+
+	wvc_url = (char *)xmalloc(len + 2); /* +2: 'c' and EOS */
+	if (wvc_url == NULL)
+		return 0;
+
+	memcpy(wvc_url, utf8url, len);
+	wvc_url[len] = 'c';
+	wvc_url[len + 1] = '\0';
+
+	ret = openInputStream(is_wvc, wvc_url);
+	free(wvc_url);
+
+	if (ret)
+		return 0;
+
+	/*
+	 * And we try to buffer in order to get know
+	 * about a possible 404 error.
+	 */
+	for (;;) {
+		if (inputStreamAtEOF(is_wvc)) {
+			/*
+			 * EOF is reached even without
+			 * a single byte is read...
+			 * So, this is not good :/
+			 */
+			closeInputStream(is_wvc);
+			return 0;
+		}
+
+		if (bufferInputStream(is_wvc) >= 0)
+			return 1;
+
+		if (decoder_get_command(decoder) != DECODE_COMMAND_NONE) {
+			closeInputStream(is_wvc);
+			return 0;
+		}
+
+		/* Save some CPU */
+		my_usleep(1000);
+	}
+}
+
+/*
+ * Decodes a stream.
+ */
+static int wavpack_streamdecode(struct decoder * decoder, InputStream *is)
+{
+	char error[ERRORLEN];
+	WavpackContext *wpc;
+	InputStream is_wvc;
+	int open_flags = OPEN_2CH_MAX | OPEN_NORMALIZE /*| OPEN_STREAMING*/;
+	InputStreamPlus isp, isp_wvc;
+
+	if (wavpack_open_wvc(decoder, &is_wvc)) {
+		initInputStreamPlus(&isp_wvc, decoder, &is_wvc);
+		open_flags |= OPEN_WVC;
+	}
+
+	initInputStreamPlus(&isp, decoder, is);
+	wpc = WavpackOpenFileInputEx(&mpd_is_reader, &isp, &isp_wvc, error,
+				     open_flags, 15);
+
+	if (wpc == NULL) {
+		ERROR("failed to open WavPack stream: %s\n", error);
+		return -1;
+	}
+
+	wavpack_decode(decoder, wpc, can_seek(&isp), NULL);
+
+	WavpackCloseFile(wpc);
+	if (open_flags & OPEN_WVC)
+		closeInputStream(&is_wvc);
+	closeInputStream(is);
+
+	return 0;
+}
+
+/*
+ * Decodes a file.
+ */
+static int wavpack_filedecode(struct decoder * decoder, char *fname)
+{
+	char error[ERRORLEN];
+	WavpackContext *wpc;
+	ReplayGainInfo *replayGainInfo;
+
+	wpc = WavpackOpenFileInput(fname, error,
+	                           OPEN_TAGS | OPEN_WVC |
+	                           OPEN_2CH_MAX | OPEN_NORMALIZE, 15);
+	if (wpc == NULL) {
+		ERROR("failed to open WavPack file \"%s\": %s\n", fname, error);
+		return -1;
+	}
+
+	replayGainInfo = wavpack_replaygain(wpc);
+
+	wavpack_decode(decoder, wpc, 1, replayGainInfo);
+
+	if (replayGainInfo)
+		freeReplayGainInfo(replayGainInfo);
+
+	WavpackCloseFile(wpc);
+
+	return 0;
+}
+
+static char const *wavpackSuffixes[] = { "wv", NULL };
+static char const *wavpackMimeTypes[] = { "audio/x-wavpack", NULL };
+
+struct decoder_plugin wavpackPlugin = {
+	.name = "wavpack",
+	.try_decode = wavpack_trydecode,
+	.stream_decode = wavpack_streamdecode,
+	.file_decode = wavpack_filedecode,
+	.tag_dup = wavpack_tagdup,
+	.stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+	.suffixes = wavpackSuffixes,
+	.mime_types = wavpackMimeTypes
+};