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, 0 insertions, 5609 deletions

diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
deleted file mode 100644
index db43e0003..000000000
--- a/src/inputPlugins/_flac_common.c
+++ /dev/null
@@ -1,323 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * 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/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h
deleted file mode 100644
index 45714b4bd..000000000
--- a/src/inputPlugins/_flac_common.h
+++ /dev/null
@@ -1,168 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * 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/inputPlugins/_ogg_common.c b/src/inputPlugins/_ogg_common.c
deleted file mode 100644
index 841b2ad3f..000000000
--- a/src/inputPlugins/_ogg_common.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * 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/inputPlugins/_ogg_common.h b/src/inputPlugins/_ogg_common.h
deleted file mode 100644
index 7c9e7b630..000000000
--- a/src/inputPlugins/_ogg_common.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * 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/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
deleted file mode 100644
index 7842bcc22..000000000
--- a/src/inputPlugins/aac_plugin.c
+++ /dev/null
@@ -1,602 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- * 
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../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/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
deleted file mode 100644
index 99846e853..000000000
--- a/src/inputPlugins/audiofile_plugin.c
+++ /dev/null
@@ -1,147 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- * 
- * 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/inputPlugins/ffmpeg_plugin.c b/src/inputPlugins/ffmpeg_plugin.c
deleted file mode 100644
index 6455cd1ce..000000000
--- a/src/inputPlugins/ffmpeg_plugin.c
+++ /dev/null
@@ -1,419 +0,0 @@
-/* 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/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
deleted file mode 100644
index 7b9fce27d..000000000
--- a/src/inputPlugins/flac_plugin.c
+++ /dev/null
@@ -1,459 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "_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/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
deleted file mode 100644
index 5916a24ab..000000000
--- a/src/inputPlugins/mod_plugin.c
+++ /dev/null
@@ -1,278 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- * 
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../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/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
deleted file mode 100644
index a0de30ba7..000000000
--- a/src/inputPlugins/mp3_plugin.c
+++ /dev/null
@@ -1,1086 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../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/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
deleted file mode 100644
index 4a613744e..000000000
--- a/src/inputPlugins/mp4_plugin.c
+++ /dev/null
@@ -1,423 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- * 
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../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/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
deleted file mode 100644
index fb1b0b56c..000000000
--- a/src/inputPlugins/mpc_plugin.c
+++ /dev/null
@@ -1,308 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../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/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
deleted file mode 100644
index 091b00988..000000000
--- a/src/inputPlugins/oggflac_plugin.c
+++ /dev/null
@@ -1,355 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * 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/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
deleted file mode 100644
index 0eecb783f..000000000
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ /dev/null
@@ -1,387 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-/* 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/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
deleted file mode 100644
index 14b7e5f69..000000000
--- a/src/inputPlugins/wavpack_plugin.c
+++ /dev/null
@@ -1,574 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- * 
- * 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
-};