29 files changed, 3336 insertions, 2569 deletions

diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index 7c8fe9875..8dd22a253 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -21,7 +21,11 @@
  * Common data structures and functions used by FLAC and OggFLAC
  */
 
+#include "config.h"
 #include "_flac_common.h"
+#include "flac_metadata.h"
+#include "flac_pcm.h"
+#include "audio_check.h"
 
 #include <glib.h>
 
@@ -31,186 +35,104 @@ void
 flac_data_init(struct flac_data *data, struct decoder * decoder,
 	       struct input_stream *input_stream)
 {
-	data->time = 0;
+	pcm_buffer_init(&data->buffer);
+
+	data->unsupported = false;
+	data->initialized = false;
+	data->total_frames = 0;
+	data->first_frame = 0;
+	data->next_frame = 0;
+
 	data->position = 0;
-	data->bit_rate = 0;
 	data->decoder = decoder;
 	data->input_stream = input_stream;
-	data->replay_gain_info = NULL;
 	data->tag = NULL;
 }
 
-static void
-flac_find_float_comment(const FLAC__StreamMetadata *block,
-			const char *cmnt, float *fl, bool *found_r)
+void
+flac_data_deinit(struct flac_data *data)
 {
-	int offset;
-	size_t pos;
-	int len;
-	unsigned char tmp, *p;
-
-	offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0,
-								     cmnt);
-	if (offset < 0)
-		return;
-
-	pos = strlen(cmnt) + 1; /* 1 is for '=' */
-	len = block->data.vorbis_comment.comments[offset].length - pos;
-	if (len <= 0)
-		return;
-
-	p = &block->data.vorbis_comment.comments[offset].entry[pos];
-	tmp = p[len];
-	p[len] = '\0';
-	*fl = (float)atof((char *)p);
-	p[len] = tmp;
-
-	*found_r = true;
-}
+	pcm_buffer_deinit(&data->buffer);
 
-static void
-flac_parse_replay_gain(const FLAC__StreamMetadata *block,
-		       struct flac_data *data)
-{
-	bool found = false;
-
-	if (data->replay_gain_info)
-		replay_gain_info_free(data->replay_gain_info);
-
-	data->replay_gain_info = replay_gain_info_new();
-
-	flac_find_float_comment(block, "replaygain_album_gain",
-				&data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain,
-				&found);
-	flac_find_float_comment(block, "replaygain_album_peak",
-				&data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak,
-				&found);
-	flac_find_float_comment(block, "replaygain_track_gain",
-				&data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain,
-				&found);
-	flac_find_float_comment(block, "replaygain_track_peak",
-				&data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak,
-				&found);
-
-	if (!found) {
-		replay_gain_info_free(data->replay_gain_info);
-		data->replay_gain_info = NULL;
-	}
+	if (data->tag != NULL)
+		tag_free(data->tag);
 }
 
-/**
- * Checks if the specified name matches the entry's name, and if yes,
- * returns the comment value (not null-temrinated).
- */
-static const char *
-flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry,
-		   const char *name, const char *char_tnum, size_t *length_r)
+static enum sample_format
+flac_sample_format(unsigned bits_per_sample)
 {
-	size_t name_length = strlen(name);
-	size_t char_tnum_length = 0;
-	const char *comment = (const char*)entry->entry;
-
-	if (entry->length <= name_length ||
-	    g_ascii_strncasecmp(comment, name, name_length) != 0)
-		return NULL;
-
-	if (char_tnum != NULL) {
-		char_tnum_length = strlen(char_tnum);
-		if (entry->length > name_length + char_tnum_length + 2 &&
-		    comment[name_length] == '[' &&
-		    g_ascii_strncasecmp(comment + name_length + 1,
-					char_tnum, char_tnum_length) == 0 &&
-		    comment[name_length + char_tnum_length + 1] == ']')
-			name_length = name_length + char_tnum_length + 2;
-		else if (entry->length > name_length + char_tnum_length &&
-			 g_ascii_strncasecmp(comment + name_length,
-					     char_tnum, char_tnum_length) == 0)
-			name_length = name_length + char_tnum_length;
-	}
+	switch (bits_per_sample) {
+	case 8:
+		return SAMPLE_FORMAT_S8;
 
-	if (comment[name_length] == '=') {
-		*length_r = entry->length - name_length - 1;
-		return comment + name_length + 1;
-	}
+	case 16:
+		return SAMPLE_FORMAT_S16;
 
-	return NULL;
-}
+	case 24:
+		return SAMPLE_FORMAT_S24_P32;
 
-/**
- * Check if the comment's name equals the passed name, and if so, copy
- * the comment value into the tag.
- */
-static bool
-flac_copy_comment(struct tag *tag,
-		  const FLAC__StreamMetadata_VorbisComment_Entry *entry,
-		  const char *name, enum tag_type tag_type,
-		  const char *char_tnum)
-{
-	const char *value;
-	size_t value_length;
+	case 32:
+		return SAMPLE_FORMAT_S32;
 
-	value = flac_comment_value(entry, name, char_tnum, &value_length);
-	if (value != NULL) {
-		tag_add_item_n(tag, tag_type, value, value_length);
-		return true;
+	default:
+		return SAMPLE_FORMAT_UNDEFINED;
 	}
-
-	return false;
 }
 
-/* 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 void
-flac_parse_comment(struct tag *tag, const char *char_tnum,
-		   const FLAC__StreamMetadata_VorbisComment_Entry *entry)
+flac_got_stream_info(struct flac_data *data,
+		     const FLAC__StreamMetadata_StreamInfo *stream_info)
 {
-	assert(tag != NULL);
-
-	if (flac_copy_comment(tag, entry, VORBIS_COMMENT_TRACK_KEY,
-			      TAG_ITEM_TRACK, char_tnum) ||
-	    flac_copy_comment(tag, entry, VORBIS_COMMENT_DISC_KEY,
-			      TAG_ITEM_DISC, char_tnum) ||
-	    flac_copy_comment(tag, entry, "album artist",
-			      TAG_ITEM_ALBUM_ARTIST, char_tnum))
+	if (data->initialized || data->unsupported)
 		return;
 
-	for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i)
-		if (flac_copy_comment(tag, entry,
-				      tag_item_names[i], i, char_tnum))
-			return;
-}
+	GError *error = NULL;
+	if (!audio_format_init_checked(&data->audio_format,
+				       stream_info->sample_rate,
+				       flac_sample_format(stream_info->bits_per_sample),
+				       stream_info->channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		data->unsupported = true;
+		return;
+	}
 
-void
-flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
-			    const FLAC__StreamMetadata *block)
-{
-	FLAC__StreamMetadata_VorbisComment_Entry *comments =
-		block->data.vorbis_comment.comments;
+	data->frame_size = audio_format_frame_size(&data->audio_format);
 
-	for (unsigned i = block->data.vorbis_comment.num_comments; i > 0; --i)
-		flac_parse_comment(tag, char_tnum, comments++);
+	if (data->total_frames == 0)
+		data->total_frames = stream_info->total_samples;
+
+	data->initialized = true;
 }
 
 void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 			     struct flac_data *data)
 {
-	const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info);
+	if (data->unsupported)
+		return;
+
+	struct replay_gain_info rgi;
+	char *mixramp_start;
+	char *mixramp_end;
+	float replay_gain_db = 0;
 
 	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);
+		flac_got_stream_info(data, &block->data.stream_info);
 		break;
+
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
-		flac_parse_replay_gain(block, data);
+		if (flac_parse_replay_gain(&rgi, block))
+			replay_gain_db = decoder_replay_gain(data->decoder, &rgi);
+		if (flac_parse_mixramp(&mixramp_start, &mixramp_end, block)) {
+			g_debug("setting mixramp_tags");
+			decoder_mixramp(data->decoder, replay_gain_db,
+					mixramp_start, mixramp_end);
+		}
 
 		if (data->tag != NULL)
-			flac_vorbis_comments_to_tag(data->tag, NULL, block);
+			flac_vorbis_comments_to_tag(data->tag, NULL,
+						    &block->data.vorbis_comment);
 
 	default:
 		break;
@@ -239,187 +161,82 @@ void flac_error_common_cb(const char *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!
+ * This function attempts to call decoder_initialized() in case there
+ * was no STREAMINFO block.  This is allowed for nonseekable streams,
+ * where the server sends us only a part of the file, without
+ * providing the STREAMINFO block from the beginning of the file
+ * (e.g. when seeking with SqueezeBox Server).
  */
-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;
+static bool
+flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header)
+{
+	if (data->unsupported)
+		return false;
+
+	GError *error = NULL;
+	if (!audio_format_init_checked(&data->audio_format,
+				       header->sample_rate,
+				       flac_sample_format(header->bits_per_sample),
+				       header->channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		data->unsupported = true;
+		return false;
+	}
 
-	for (; position < end; ++position)
-		for (c_chan = 0; c_chan < num_channels; c_chan++)
-			*dest++ = buf[c_chan][position];
-}
+	data->frame_size = audio_format_frame_size(&data->audio_format);
 
-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;
+	decoder_initialized(data->decoder, &data->audio_format,
+			    data->input_stream->seekable,
+			    (float)data->total_frames /
+			    (float)data->audio_format.sample_rate);
 
-	case 4:
-		flac_convert_32((int32_t*)dest, num_channels, buf,
-				position, end);
-		break;
+	data->initialized = true;
 
-	case 1:
-		flac_convert_8((int8_t*)dest, num_channels, buf,
-			       position, end);
-		break;
-	}
+	return true;
 }
 
 FLAC__StreamDecoderWriteStatus
 flac_common_write(struct flac_data *data, const FLAC__Frame * frame,
-		  const FLAC__int32 *const buf[])
+		  const FLAC__int32 *const buf[],
+		  FLAC__uint64 nbytes)
 {
-	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;
+	void *buffer;
+	unsigned bit_rate;
 
-	if (bytes_per_sample != 1 && bytes_per_sample != 2 &&
-	    bytes_per_sample != 4)
-		/* exotic unsupported bit rate */
+	if (!data->initialized && !flac_got_first_frame(data, &frame->header))
 		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->input_stream,
-				   data->chunk,
-				   num_samples * bytes_per_channel,
-				   data->time, data->bit_rate,
-				   data->replay_gain_info);
-		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;
-}
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+	size_t buffer_size = frame->header.blocksize * data->frame_size;
+	buffer = pcm_buffer_get(&data->buffer, buffer_size);
 
-char*
-flac_cue_track(	const char* pathname,
-		const unsigned int tnum)
-{
-	FLAC__bool success;
-	FLAC__StreamMetadata* cs;
-
-	success = FLAC__metadata_get_cuesheet(pathname, &cs);
-	if (!success)
-		return NULL;
-
-	assert(cs != NULL);
-
-	if (cs->data.cue_sheet.num_tracks <= 1)
-	{
-		FLAC__metadata_object_delete(cs);
-		return NULL;
-	}
+	flac_convert(buffer, frame->header.channels,
+		     data->audio_format.format, buf,
+		     0, frame->header.blocksize);
 
-	if (tnum > 0 && tnum < cs->data.cue_sheet.num_tracks)
-	{
-		char* track = g_strdup_printf("track_%03u.flac", tnum);
+	if (nbytes > 0)
+		bit_rate = nbytes * 8 * frame->header.sample_rate /
+			(1000 * frame->header.blocksize);
+	else
+		bit_rate = 0;
+
+	cmd = decoder_data(data->decoder, data->input_stream,
+			   buffer, buffer_size,
+			   bit_rate);
+	data->next_frame += frame->header.blocksize;
+	switch (cmd) {
+	case DECODE_COMMAND_NONE:
+	case DECODE_COMMAND_START:
+		break;
 
-		FLAC__metadata_object_delete(cs);
+	case DECODE_COMMAND_STOP:
+		return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
 
-		return track;
-	}
-	else
-	{
-		FLAC__metadata_object_delete(cs);
-		return NULL;
+	case DECODE_COMMAND_SEEK:
+		return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
 	}
-}
 
-unsigned int
-flac_vtrack_tnum(const char* fname)
-{
-	/* find last occurrence of '_' in fname
-	 * which is hopefully something like track_xxx.flac
-	 * another/better way would be to use tag struct
-	 */
-	char* ptr = strrchr(fname, '_');
-	if (ptr == NULL)
-		return 0;
-
-	// copy ascii tracknumber to int
-	return (unsigned int)strtol(++ptr, NULL, 10);
+	return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
 }
-
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h
index 68de7e969..5c59ee123 100644
--- a/src/decoder/_flac_common.h
+++ b/src/decoder/_flac_common.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -24,136 +24,62 @@
 #ifndef MPD_FLAC_COMMON_H
 #define MPD_FLAC_COMMON_H
 
-#include "../decoder_api.h"
-#include "config.h"
+#include "decoder_api.h"
+#include "pcm_buffer.h"
 
 #include <glib.h>
 
+#include <FLAC/stream_decoder.h>
+#include <FLAC/metadata.h>
+
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "flac"
 
-#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 */
+struct flac_data {
+	struct pcm_buffer buffer;
+
+	/**
+	 * The size of one frame in the output buffer.
+	 */
+	unsigned frame_size;
+
+	/**
+	 * Has decoder_initialized() been called yet?
+	 */
+	bool initialized;
+
+	/**
+	 * Does the FLAC file contain an unsupported audio format?
+	 */
+	bool unsupported;
+
+	/**
+	 * The validated audio format of the FLAC file.  This
+	 * attribute is defined if "initialized" is true.
+	 */
+	struct audio_format audio_format;
 
-/*
- * 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 */
+	/**
+	 * The total number of frames in this song.  The decoder
+	 * plugin may initialize this attribute to override the value
+	 * provided by libFLAC (e.g. for sub songs from a CUE sheet).
+	 */
+	FLAC__uint64 total_frames;
 
-#include <FLAC/metadata.h>
+	/**
+	 * The number of the first frame in this song.  This is only
+	 * non-zero if playing sub songs from a CUE sheet.
+	 */
+	FLAC__uint64 first_frame;
 
-#define FLAC_CHUNK_SIZE 4080
+	/**
+	 * The number of the next frame which is going to be decoded.
+	 */
+	FLAC__uint64 next_frame;
 
-struct flac_data {
-	unsigned char chunk[FLAC_CHUNK_SIZE];
-	float time;
-	unsigned int bit_rate;
-	struct audio_format audio_format;
-	float total_time;
 	FLAC__uint64 position;
 	struct decoder *decoder;
 	struct input_stream *input_stream;
-	struct replay_gain_info *replay_gain_info;
 	struct tag *tag;
 };
 
@@ -162,6 +88,9 @@ void
 flac_data_init(struct flac_data *data, struct decoder * decoder,
 	       struct input_stream *input_stream);
 
+void
+flac_data_deinit(struct flac_data *data);
+
 void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 			     struct flac_data *data);
 
@@ -169,23 +98,9 @@ void flac_error_common_cb(const char *plugin,
 			  FLAC__StreamDecoderErrorStatus status,
 			  struct flac_data *data);
 
-void
-flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
-			    const FLAC__StreamMetadata *block);
-
 FLAC__StreamDecoderWriteStatus
 flac_common_write(struct flac_data *data, const FLAC__Frame * frame,
-		  const FLAC__int32 *const buf[]);
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-
-char*
-flac_cue_track(		const char* pathname,
-			const unsigned int tnum);
-
-unsigned int
-flac_vtrack_tnum(	const char*);
-
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+		  const FLAC__int32 *const buf[],
+		  FLAC__uint64 nbytes);
 
 #endif /* _FLAC_COMMON_H */
diff --git a/src/decoder/_ogg_common.c b/src/decoder/_ogg_common.c
index 6c6553422..bd0650ac4 100644
--- a/src/decoder/_ogg_common.c
+++ b/src/decoder/_ogg_common.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -21,8 +21,8 @@
  * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC)
  */
 
+#include "config.h"
 #include "_ogg_common.h"
-#include "../utils.h"
 
 ogg_stream_type ogg_stream_type_detect(struct input_stream *inStream)
 {
diff --git a/src/decoder/_ogg_common.h b/src/decoder/_ogg_common.h
index e650c366d..f8446c69c 100644
--- a/src/decoder/_ogg_common.h
+++ b/src/decoder/_ogg_common.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -24,7 +24,7 @@
 #ifndef MPD_OGG_COMMON_H
 #define MPD_OGG_COMMON_H
 
-#include "../decoder_api.h"
+#include "decoder_api.h"
 
 typedef enum _ogg_stream_type { VORBIS, FLAC } ogg_stream_type;
 
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_decoder_plugin.c
index f66d90dc1..3026f3cc7 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,7 +17,9 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
+#include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
 
 #include <audiofile.h>
 #include <af_vfs.h>
@@ -45,10 +47,20 @@ static int audiofile_get_duration(const char *file)
 }
 
 static ssize_t
-audiofile_file_read(AFvirtualfile *vfile, void *data, size_t nbytes)
+audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length)
 {
 	struct input_stream *is = (struct input_stream *) vfile->closure;
-	return input_stream_read(is, data, nbytes);
+	GError *error = NULL;
+	size_t nbytes;
+
+	nbytes = input_stream_read(is, data, length, &error);
+	if (nbytes == 0 && error != NULL) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return -1;
+	}
+
+	return nbytes;
 }
 
 static long
@@ -78,7 +90,7 @@ audiofile_file_seek(AFvirtualfile *vfile, long offset, int is_relative)
 {
 	struct input_stream *is = (struct input_stream *) vfile->closure;
 	int whence = (is_relative ? SEEK_CUR : SEEK_SET);
-	if (input_stream_seek(is, offset, whence)) {
+	if (input_stream_seek(is, offset, whence, NULL)) {
 		return is->offset;
 	} else {
 		return -1;
@@ -99,17 +111,56 @@ setup_virtual_fops(struct input_stream *stream)
 	return vf;
 }
 
+static enum sample_format
+audiofile_bits_to_sample_format(int bits)
+{
+	switch (bits) {
+	case 8:
+		return SAMPLE_FORMAT_S8;
+
+	case 16:
+		return SAMPLE_FORMAT_S16;
+
+	case 24:
+		return SAMPLE_FORMAT_S24_P32;
+
+	case 32:
+		return SAMPLE_FORMAT_S32;
+	}
+
+	return SAMPLE_FORMAT_UNDEFINED;
+}
+
+static enum sample_format
+audiofile_setup_sample_format(AFfilehandle af_fp)
+{
+	int fs, bits;
+
+	afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
+	if (!audio_valid_sample_format(audiofile_bits_to_sample_format(bits))) {
+		g_debug("input file has %d bit samples, converting to 16",
+			bits);
+		bits = 16;
+	}
+
+	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
+	                         AF_SAMPFMT_TWOSCOMP, bits);
+	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
+
+	return audiofile_bits_to_sample_format(bits);
+}
+
 static void
 audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 {
+	GError *error = NULL;
 	AFvirtualfile *vf;
 	int fs, frame_count;
 	AFfilehandle af_fp;
-	int bits;
 	struct audio_format audio_format;
 	float total_time;
 	uint16_t bit_rate;
-	int ret, current = 0;
+	int ret;
 	char chunk[CHUNK_SIZE];
 	enum decoder_command cmd;
 
@@ -126,26 +177,13 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	if (!audio_valid_sample_format(bits)) {
-		g_debug("input file has %d bit samples, converting to 16",
-			bits);
-		bits = 16;
-	}
-
-	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
-	                         AF_SAMPFMT_TWOSCOMP, bits);
-	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);
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate, audio_format.bits,
-			  audio_format.channels);
+	if (!audio_format_init_checked(&audio_format,
+				       afGetRate(af_fp, AF_DEFAULT_TRACK),
+				       audiofile_setup_sample_format(af_fp),
+				       afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		afCloseFile(af_fp);
 		return;
 	}
@@ -166,17 +204,14 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 		if (ret <= 0)
 			break;
 
-		current += ret;
 		cmd = decoder_data(decoder, NULL,
 				   chunk, ret * fs,
-				   (float)current /
-				   (float)audio_format.sample_rate,
-				   bit_rate, NULL);
+				   bit_rate);
 
 		if (cmd == DECODE_COMMAND_SEEK) {
-			current = decoder_seek_where(decoder) *
+			AFframecount frame = decoder_seek_where(decoder) *
 				audio_format.sample_rate;
-			afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
+			afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame);
 
 			decoder_command_finished(decoder);
 			cmd = DECODE_COMMAND_NONE;
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_decoder_plugin.c
index 7b2806a4c..8f932ad58 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,9 +17,10 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
-#include "decoder_buffer.h"
 #include "config.h"
+#include "decoder_api.h"
+#include "decoder_buffer.h"
+#include "audio_check.h"
 
 #define AAC_MAX_CHANNELS	6
 
@@ -37,6 +38,15 @@ static const unsigned adts_sample_rates[] =
 };
 
 /**
+ * The GLib quark used for errors reported by this plugin.
+ */
+static inline GQuark
+faad_decoder_quark(void)
+{
+	return g_quark_from_static_string("faad");
+}
+
+/**
  * Check whether the buffer head is an AAC frame, and return the frame
  * length.  Returns 0 if it is not a frame.
  */
@@ -195,7 +205,7 @@ faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
 		/* obtain the duration from the ADTS header */
 		float song_length = adts_song_duration(buffer);
 
-		input_stream_seek(is, tagsize, SEEK_SET);
+		input_stream_seek(is, tagsize, SEEK_SET, NULL);
 
 		data = decoder_buffer_read(buffer, &length);
 		if (data != NULL)
@@ -232,7 +242,7 @@ faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
  */
 static bool
 faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
-		  struct audio_format *audio_format)
+		  struct audio_format *audio_format, GError **error_r)
 {
 	union {
 		/* deconst hack for libfaad */
@@ -247,32 +257,33 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
 	/* neaacdec.h declares all arguments as "unsigned long", but
 	   internally expects uint32_t pointers.  To avoid gcc
 	   warnings, use this workaround. */
-	unsigned long *sample_rate_r = (unsigned long *)(void *)&sample_rate;
+	unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
 #else
-	uint32_t *sample_rate_r = &sample_rate;
+	uint32_t *sample_rate_p = &sample_rate;
 #endif
 
 	u.in = decoder_buffer_read(buffer, &length);
-	if (u.in == NULL)
+	if (u.in == NULL) {
+		g_set_error(error_r, faad_decoder_quark(), 0,
+			    "Empty file");
 		return false;
+	}
 
 	nbytes = faacDecInit(decoder, u.out,
 #ifdef HAVE_FAAD_BUFLEN_FUNCS
 			     length,
 #endif
-			     sample_rate_r, &channels);
-	if (nbytes < 0)
+			     sample_rate_p, &channels);
+	if (nbytes < 0) {
+		g_set_error(error_r, faad_decoder_quark(), 0,
+			    "Not an AAC stream");
 		return false;
+	}
 
 	decoder_buffer_consume(buffer, nbytes);
 
-	*audio_format = (struct audio_format){
-		.bits = 16,
-		.channels = channels,
-		.sample_rate = sample_rate,
-	};
-
-	return true;
+	return audio_format_init_checked(audio_format, sample_rate,
+					 SAMPLE_FORMAT_S16, channels, error_r);
 }
 
 /**
@@ -311,20 +322,16 @@ faad_decoder_decode(faacDecHandle decoder, struct decoder_buffer *buffer,
  * file is invalid.
  */
 static float
-faad_get_file_time_float(const char *file)
+faad_get_file_time_float(struct input_stream *is)
 {
 	struct decoder_buffer *buffer;
 	float length;
 	faacDecHandle decoder;
 	faacDecConfigurationPtr config;
-	struct input_stream is;
-
-	if (!input_stream_open(&is, file))
-		return -1;
 
-	buffer = decoder_buffer_new(NULL, &is,
+	buffer = decoder_buffer_new(NULL, is,
 				    FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
-	length = faad_song_duration(buffer, &is);
+	length = faad_song_duration(buffer, is);
 
 	if (length < 0) {
 		bool ret;
@@ -338,15 +345,14 @@ faad_get_file_time_float(const char *file)
 
 		decoder_buffer_fill(buffer);
 
-		ret = faad_decoder_init(decoder, buffer, &audio_format);
-		if (ret && audio_format_valid(&audio_format))
+		ret = faad_decoder_init(decoder, buffer, &audio_format, NULL);
+		if (ret)
 			length = 0;
 
 		faacDecClose(decoder);
 	}
 
 	decoder_buffer_free(buffer);
-	input_stream_close(&is);
 
 	return length;
 }
@@ -357,12 +363,12 @@ faad_get_file_time_float(const char *file)
  * file is invalid.
  */
 static int
-faad_get_file_time(const char *file)
+faad_get_file_time(struct input_stream *is)
 {
 	int file_time = -1;
 	float length;
 
-	if ((length = faad_get_file_time_float(file)) >= 0)
+	if ((length = faad_get_file_time_float(is)) >= 0)
 		file_time = length + 0.5;
 
 	return file_time;
@@ -371,7 +377,7 @@ faad_get_file_time(const char *file)
 static void
 faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 {
-	float file_time;
+	GError *error = NULL;
 	float total_time = 0;
 	faacDecHandle decoder;
 	struct audio_format audio_format;
@@ -408,15 +414,10 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 
 	/* initialize it */
 
-	ret = faad_decoder_init(decoder, buffer, &audio_format);
+	ret = faad_decoder_init(decoder, buffer, &audio_format, &error);
 	if (!ret) {
-		g_warning("Error not a AAC stream.\n");
-		faacDecClose(decoder);
-		return;
-	}
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("invalid audio format\n");
+		g_warning("%s", error->message);
+		g_error_free(error);
 		faacDecClose(decoder);
 		return;
 	}
@@ -427,8 +428,6 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 
 	/* the decoder loop */
 
-	file_time = 0.0;
-
 	do {
 		size_t frame_size;
 		const void *decoded;
@@ -474,16 +473,13 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 			bit_rate = frame_info.bytesconsumed * 8.0 *
 			    frame_info.channels * audio_format.sample_rate /
 			    frame_info.samples / 1000 + 0.5;
-			file_time +=
-			    (float)(frame_info.samples) / frame_info.channels /
-			    audio_format.sample_rate;
 		}
 
 		/* send PCM samples to MPD */
 
 		cmd = decoder_data(mpd_decoder, is, decoded,
-				   (size_t)frame_info.samples * 2, file_time,
-				   bit_rate, NULL);
+				   (size_t)frame_info.samples * 2,
+				   bit_rate);
 	} while (cmd != DECODE_COMMAND_STOP);
 
 	/* cleanup */
@@ -492,15 +488,13 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 }
 
 static struct tag *
-faad_tag_dup(const char *file)
+faad_stream_tag(struct input_stream *is)
 {
-	int file_time = faad_get_file_time(file);
+	int file_time = faad_get_file_time(is);
 	struct tag *tag;
 
-	if (file_time < 0) {
-		g_debug("Failed to get total song time from: %s", file);
+	if (file_time < 0)
 		return NULL;
-	}
 
 	tag = tag_new();
 	tag->time = file_time;
@@ -515,7 +509,7 @@ static const char *const faad_mime_types[] = {
 const struct decoder_plugin faad_decoder_plugin = {
 	.name = "faad",
 	.stream_decode = faad_stream_decode,
-	.tag_dup = faad_tag_dup,
+	.stream_tag = faad_stream_tag,
 	.suffixes = faad_suffixes,
 	.mime_types = faad_mime_types,
 };
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_decoder_plugin.c
index 10894b633..f9d4eb8a9 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,9 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
 #include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
 
 #include <glib.h>
 
@@ -39,19 +40,46 @@
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavformat/avio.h>
+#include <libavutil/log.h>
 #endif
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "ffmpeg"
 
-struct ffmpeg_context {
-	int audio_stream;
-	AVFormatContext *format_context;
-	AVCodecContext *codec_context;
-	struct decoder *decoder;
-	struct input_stream *input;
-	struct tag *tag;
-};
+#ifndef OLD_FFMPEG_INCLUDES
+
+static GLogLevelFlags
+level_ffmpeg_to_glib(int level)
+{
+	if (level <= AV_LOG_FATAL)
+		return G_LOG_LEVEL_CRITICAL;
+
+	if (level <= AV_LOG_ERROR)
+		return G_LOG_LEVEL_WARNING;
+
+	if (level <= AV_LOG_INFO)
+		return G_LOG_LEVEL_MESSAGE;
+
+	return G_LOG_LEVEL_DEBUG;
+}
+
+static void
+mpd_ffmpeg_log_callback(G_GNUC_UNUSED void *ptr, int level,
+			const char *fmt, va_list vl)
+{
+	const AVClass * cls = NULL;
+
+	if (ptr != NULL)
+		cls = *(const AVClass *const*)ptr;
+
+	if (cls != NULL) {
+		char *domain = g_strconcat(G_LOG_DOMAIN, "/", cls->item_name(ptr), NULL);
+		g_logv(domain, level_ffmpeg_to_glib(level), fmt, vl);
+		g_free(domain);
+	}
+}
+
+#endif /* !OLD_FFMPEG_INCLUDES */
 
 struct mpd_ffmpeg_stream {
 	struct decoder *decoder;
@@ -79,7 +107,7 @@ mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence)
 	if (whence == AVSEEK_SIZE)
 		return stream->input->size;
 
-	ret = input_stream_seek(stream->input, pos, whence);
+	ret = input_stream_seek(stream->input, pos, whence, NULL);
 	if (!ret)
 		return -1;
 
@@ -115,6 +143,10 @@ mpd_ffmpeg_stream_close(struct mpd_ffmpeg_stream *stream)
 static bool
 ffmpeg_init(G_GNUC_UNUSED const struct config_param *param)
 {
+#ifndef OLD_FFMPEG_INCLUDES
+	av_log_set_callback(mpd_ffmpeg_log_callback);
+#endif
+
 	av_register_all();
 	return true;
 }
@@ -130,9 +162,95 @@ ffmpeg_find_audio_stream(const AVFormatContext *format_context)
 	return -1;
 }
 
+/**
+ * On some platforms, libavcodec wants the output buffer aligned to 16
+ * bytes (because it uses SSE/Altivec internally).  This function
+ * returns the aligned version of the specified buffer, and corrects
+ * the buffer size.
+ */
+static void *
+align16(void *p, size_t *length_p)
+{
+	unsigned add = 16 - (size_t)p % 16;
+
+	*length_p -= add;
+	return (char *)p + add;
+}
+
+static enum decoder_command
+ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
+		   const AVPacket *packet,
+		   AVCodecContext *codec_context,
+		   const AVRational *time_base)
+{
+	enum decoder_command cmd = DECODE_COMMAND_NONE;
+	uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
+	int16_t *aligned_buffer;
+	size_t buffer_size;
+	int len, audio_size;
+	uint8_t *packet_data;
+	int packet_size;
+
+	if (packet->pts != (int64_t)AV_NOPTS_VALUE)
+		decoder_timestamp(decoder,
+				  av_rescale_q(packet->pts, *time_base,
+					       (AVRational){1, 1}));
+
+	packet_data = packet->data;
+	packet_size = packet->size;
+
+	buffer_size = sizeof(audio_buf);
+	aligned_buffer = align16(audio_buf, &buffer_size);
+
+	while ((packet_size > 0) && (cmd == DECODE_COMMAND_NONE)) {
+		audio_size = buffer_size;
+		len = avcodec_decode_audio2(codec_context,
+					    aligned_buffer, &audio_size,
+					    packet_data, packet_size);
+
+		if (len < 0) {
+			/* if error, we skip the frame */
+			g_message("decoding failed\n");
+			break;
+		}
+
+		packet_data += len;
+		packet_size -= len;
+
+		if (audio_size <= 0)
+			continue;
+
+		cmd = decoder_data(decoder, is,
+				   aligned_buffer, audio_size,
+				   codec_context->bit_rate / 1000);
+	}
+	return cmd;
+}
+
+static enum sample_format
+ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
+{
+#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
+	switch (codec_context->sample_fmt) {
+	case SAMPLE_FMT_S16:
+		return SAMPLE_FORMAT_S16;
+
+	case SAMPLE_FMT_S32:
+		return SAMPLE_FORMAT_S32;
+
+	default:
+		g_warning("Unsupported libavcodec SampleFormat value: %d",
+			  codec_context->sample_fmt);
+		return SAMPLE_FORMAT_UNDEFINED;
+	}
+#else
+	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
+	return SAMPLE_FORMAT_S16;
+#endif
+}
+
 static AVInputFormat *
-ffmpeg_probe(struct decoder *decoder, struct input_stream *is,
-	     const char *uri)
+ffmpeg_probe(struct decoder *decoder, struct input_stream *is)
 {
 	enum {
 		BUFFER_SIZE = 16384,
@@ -141,7 +259,7 @@ ffmpeg_probe(struct decoder *decoder, struct input_stream *is,
 
 	unsigned char *buffer = g_malloc(BUFFER_SIZE);
 	size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE);
-	if (nbytes <= PADDING || !input_stream_seek(is, 0, SEEK_SET)) {
+	if (nbytes <= PADDING || !input_stream_seek(is, 0, SEEK_SET, NULL)) {
 		g_free(buffer);
 		return NULL;
 	}
@@ -155,7 +273,7 @@ ffmpeg_probe(struct decoder *decoder, struct input_stream *is,
 	AVProbeData avpd = {
 		.buf = buffer,
 		.buf_size = nbytes,
-		.filename = uri,
+		.filename = is->uri,
 	};
 
 	AVInputFormat *format = av_probe_input_format(&avpd, true);
@@ -164,15 +282,12 @@ ffmpeg_probe(struct decoder *decoder, struct input_stream *is,
 	return format;
 }
 
-static bool
-ffmpeg_helper(const char *uri,
-	      struct decoder *decoder, struct input_stream *input,
-	      bool (*callback)(struct ffmpeg_context *ctx),
-	      struct ffmpeg_context *ctx)
+static void
+ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 {
-	AVInputFormat *input_format = ffmpeg_probe(decoder, input, uri);
+	AVInputFormat *input_format = ffmpeg_probe(decoder, input);
 	if (input_format == NULL)
-		return false;
+		return;
 
 	g_debug("detected input format '%s' (%s)",
 		input_format->name, input_format->long_name);
@@ -181,28 +296,27 @@ ffmpeg_helper(const char *uri,
 		mpd_ffmpeg_stream_open(decoder, input);
 	if (stream == NULL) {
 		g_warning("Failed to open stream");
-		return false;
+		return;
 	}
 
 	AVFormatContext *format_context;
 	AVCodecContext *codec_context;
 	AVCodec *codec;
 	int audio_stream;
-	bool ret;
 
 	//ffmpeg works with ours "fileops" helper
-	if (av_open_input_stream(&format_context, stream->io, uri,
+	if (av_open_input_stream(&format_context, stream->io, input->uri,
 				 input_format, NULL) != 0) {
 		g_warning("Open failed\n");
 		mpd_ffmpeg_stream_close(stream);
-		return false;
+		return;
 	}
 
 	if (av_find_stream_info(format_context)<0) {
 		g_warning("Couldn't find stream info\n");
 		av_close_input_stream(format_context);
 		mpd_ffmpeg_stream_close(stream);
-		return false;
+		return;
 	}
 
 	audio_stream = ffmpeg_find_audio_stream(format_context);
@@ -210,7 +324,7 @@ ffmpeg_helper(const char *uri,
 		g_warning("No audio stream inside\n");
 		av_close_input_stream(format_context);
 		mpd_ffmpeg_stream_close(stream);
-		return false;
+		return;
 	}
 
 	codec_context = format_context->streams[audio_stream]->codec;
@@ -223,145 +337,48 @@ ffmpeg_helper(const char *uri,
 		g_warning("Unsupported audio codec\n");
 		av_close_input_stream(format_context);
 		mpd_ffmpeg_stream_close(stream);
-		return false;
+		return;
 	}
 
 	if (avcodec_open(codec_context, codec)<0) {
 		g_warning("Could not open codec\n");
 		av_close_input_stream(format_context);
 		mpd_ffmpeg_stream_close(stream);
-		return false;
-	}
-
-	if (callback) {
-		ctx->audio_stream = audio_stream;
-		ctx->format_context = format_context;
-		ctx->codec_context = codec_context;
-
-		ret = callback(ctx);
-	} else
-		ret = true;
-
-	avcodec_close(codec_context);
-	av_close_input_stream(format_context);
-	mpd_ffmpeg_stream_close(stream);
-
-	return ret;
-}
-
-/**
- * On some platforms, libavcodec wants the output buffer aligned to 16
- * bytes (because it uses SSE/Altivec internally).  This function
- * returns the aligned version of the specified buffer, and corrects
- * the buffer size.
- */
-static void *
-align16(void *p, size_t *length_p)
-{
-	unsigned add = 16 - (size_t)p % 16;
-
-	*length_p -= add;
-	return (char *)p + add;
-}
-
-static enum decoder_command
-ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
-		   const AVPacket *packet,
-		   AVCodecContext *codec_context,
-		   const AVRational *time_base)
-{
-	enum decoder_command cmd = DECODE_COMMAND_NONE;
-	int position;
-	uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
-	int16_t *aligned_buffer;
-	size_t buffer_size;
-	int len, audio_size;
-	uint8_t *packet_data;
-	int packet_size;
-
-	packet_data = packet->data;
-	packet_size = packet->size;
-
-	buffer_size = sizeof(audio_buf);
-	aligned_buffer = align16(audio_buf, &buffer_size);
-
-	while ((packet_size > 0) && (cmd == DECODE_COMMAND_NONE)) {
-		audio_size = buffer_size;
-		len = avcodec_decode_audio2(codec_context,
-					    aligned_buffer, &audio_size,
-					    packet_data, packet_size);
-
-		if (len < 0) {
-			/* if error, we skip the frame */
-			g_message("decoding failed\n");
-			break;
-		}
-
-		packet_data += len;
-		packet_size -= len;
-
-		if (audio_size <= 0)
-			continue;
-
-		position = packet->pts != (int64_t)AV_NOPTS_VALUE
-			? av_rescale_q(packet->pts, *time_base,
-				       (AVRational){1, 1})
-			: 0;
-
-		cmd = decoder_data(decoder, is,
-				   aligned_buffer, audio_size,
-				   position,
-				   codec_context->bit_rate / 1000, NULL);
+		return;
 	}
-	return cmd;
-}
 
-static bool
-ffmpeg_decode_internal(struct ffmpeg_context *ctx)
-{
-	struct decoder *decoder = ctx->decoder;
-	AVCodecContext *codec_context = ctx->codec_context;
-	AVFormatContext *format_context = ctx->format_context;
-	AVPacket packet;
+	GError *error = NULL;
 	struct audio_format audio_format;
-	enum decoder_command cmd;
-	int total_time;
-
-	total_time = 0;
-
-#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
-	audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
-#else
-	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
-	audio_format.bits = (uint8_t) 16;
-#endif
-	audio_format.sample_rate = (unsigned int)codec_context->sample_rate;
-	audio_format.channels = codec_context->channels;
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate, audio_format.bits,
-			  audio_format.channels);
-		return false;
+	if (!audio_format_init_checked(&audio_format,
+				       codec_context->sample_rate,
+				       ffmpeg_sample_format(codec_context),
+				       codec_context->channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		avcodec_close(codec_context);
+		av_close_input_stream(format_context);
+		mpd_ffmpeg_stream_close(stream);
+		return;
 	}
 
-	//there is some problem with this on some demux (mp3 at least)
-	if (format_context->duration != (int64_t)AV_NOPTS_VALUE) {
-		total_time = format_context->duration / AV_TIME_BASE;
-	}
+	int total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE
+		? format_context->duration / AV_TIME_BASE
+		: 0;
 
 	decoder_initialized(decoder, &audio_format,
-			    ctx->input->seekable, total_time);
+			    input->seekable, total_time);
 
+	enum decoder_command cmd;
 	do {
+		AVPacket packet;
 		if (av_read_frame(format_context, &packet) < 0)
 			/* end of file */
 			break;
 
-		if (packet.stream_index == ctx->audio_stream)
-			cmd = ffmpeg_send_packet(decoder, ctx->input,
+		if (packet.stream_index == audio_stream)
+			cmd = ffmpeg_send_packet(decoder, input,
 						 &packet, codec_context,
-						 &format_context->streams[ctx->audio_stream]->time_base);
+						 &format_context->streams[audio_stream]->time_base);
 		else
 			cmd = decoder_get_command(decoder);
 
@@ -378,115 +395,121 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 		}
 	} while (cmd != DECODE_COMMAND_STOP);
 
-	return true;
+	avcodec_close(codec_context);
+	av_close_input_stream(format_context);
+	mpd_ffmpeg_stream_close(stream);
 }
 
-static void
-ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
-{
-	struct ffmpeg_context ctx;
-
-	ctx.input = input;
-	ctx.decoder = decoder;
+#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
+typedef struct ffmpeg_tag_map {
+	enum tag_type type;
+	const char *name;
+} ffmpeg_tag_map;
 
-	char *uri = decoder_get_uri(decoder);
-	ffmpeg_helper(uri, decoder, input,
-		      ffmpeg_decode_internal, &ctx);
-	g_free(uri);
-}
+static const ffmpeg_tag_map ffmpeg_tag_maps[] = {
+	{ TAG_TITLE,             "title" },
+#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(50<<8))
+	{ TAG_ARTIST,            "artist" },
+	{ TAG_DATE,              "date" },
+#else
+	{ TAG_ARTIST,            "author" },
+	{ TAG_DATE,              "year" },
+#endif
+	{ TAG_ALBUM,             "album" },
+	{ TAG_COMMENT,           "comment" },
+	{ TAG_GENRE,             "genre" },
+	{ TAG_TRACK,             "track" },
+	{ TAG_ARTIST_SORT,       "author-sort" },
+	{ TAG_ALBUM_ARTIST,      "album_artist" },
+	{ TAG_ALBUM_ARTIST_SORT, "album_artist-sort" },
+	{ TAG_COMPOSER,          "composer" },
+	{ TAG_PERFORMER,         "performer" },
+	{ TAG_DISC,              "disc" },
+};
 
-#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
 static bool
 ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
-		     enum tag_type type, const char *name)
+		     const ffmpeg_tag_map tag_map)
 {
-	AVMetadataTag *mt = av_metadata_get(m, name, NULL, 0);
-	if (mt != NULL)
-		tag_add_item(tag, type, mt->value);
+	AVMetadataTag *mt = NULL;
+
+	while ((mt = av_metadata_get(m, tag_map.name, mt, 0)) != NULL)
+		tag_add_item(tag, tag_map.type, mt->value);
 	return mt != NULL;
 }
+
 #endif
 
-static bool ffmpeg_tag_internal(struct ffmpeg_context *ctx)
+//no tag reading in ffmpeg, check if playable
+static struct tag *
+ffmpeg_stream_tag(struct input_stream *is)
 {
-	struct tag *tag = (struct tag *) ctx->tag;
-	AVFormatContext *f = ctx->format_context;
+	AVInputFormat *input_format = ffmpeg_probe(NULL, is);
+	if (input_format == NULL)
+		return NULL;
 
-	tag->time = 0;
-	if (f->duration != (int64_t)AV_NOPTS_VALUE)
-		tag->time = f->duration / AV_TIME_BASE;
+	struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(NULL, is);
+	if (stream == NULL)
+		return NULL;
+
+	AVFormatContext *f;
+	if (av_open_input_stream(&f, stream->io, is->uri,
+				 input_format, NULL) != 0) {
+		mpd_ffmpeg_stream_close(stream);
+		return NULL;
+	}
+
+	if (av_find_stream_info(f) < 0) {
+		av_close_input_stream(f);
+		mpd_ffmpeg_stream_close(stream);
+		return NULL;
+	}
+
+	struct tag *tag = tag_new();
+
+	tag->time = f->duration != (int64_t)AV_NOPTS_VALUE
+		? f->duration / AV_TIME_BASE
+		: 0;
 
 #if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
 	av_metadata_conv(f, NULL, f->iformat->metadata_conv);
 
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TITLE, "title");
-#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(50<<8))
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "artist");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DATE, "date");
-#else
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "author");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DATE, "year");
-#endif
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM, "album");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_COMMENT, "comment");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_GENRE, "genre");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TRACK, "track");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM_ARTIST, "album_artist");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_COMPOSER, "composer");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_PERFORMER, "performer");
-	ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DISC, "disc");
+	for (unsigned i = 0; i < sizeof(ffmpeg_tag_maps)/sizeof(ffmpeg_tag_map); i++) {
+		int idx = ffmpeg_find_audio_stream(f);
+		ffmpeg_copy_metadata(tag, f->metadata, ffmpeg_tag_maps[i]);
+		if (idx >= 0)
+			ffmpeg_copy_metadata(tag, f->streams[idx]->metadata, ffmpeg_tag_maps[i]);
+	}
 #else
 	if (f->author[0])
-		tag_add_item(tag, TAG_ITEM_ARTIST, f->author);
+		tag_add_item(tag, TAG_ARTIST, f->author);
 	if (f->title[0])
-		tag_add_item(tag, TAG_ITEM_TITLE, f->title);
+		tag_add_item(tag, TAG_TITLE, f->title);
 	if (f->album[0])
-		tag_add_item(tag, TAG_ITEM_ALBUM, f->album);
+		tag_add_item(tag, TAG_ALBUM, f->album);
 
 	if (f->track > 0) {
 		char buffer[16];
 		snprintf(buffer, sizeof(buffer), "%d", f->track);
-		tag_add_item(tag, TAG_ITEM_TRACK, buffer);
+		tag_add_item(tag, TAG_TRACK, buffer);
 	}
 
 	if (f->comment[0])
-		tag_add_item(tag, TAG_ITEM_COMMENT, f->comment);
+		tag_add_item(tag, TAG_COMMENT, f->comment);
 	if (f->genre[0])
-		tag_add_item(tag, TAG_ITEM_GENRE, f->genre);
+		tag_add_item(tag, TAG_GENRE, f->genre);
 	if (f->year > 0) {
 		char buffer[16];
 		snprintf(buffer, sizeof(buffer), "%d", f->year);
-		tag_add_item(tag, TAG_ITEM_DATE, buffer);
+		tag_add_item(tag, TAG_DATE, buffer);
 	}
 
 #endif
-	return true;
-}
-
-//no tag reading in ffmpeg, check if playable
-static struct tag *ffmpeg_tag(const char *file)
-{
-	struct input_stream input;
-	struct ffmpeg_context ctx;
-	bool ret;
-
-	if (!input_stream_open(&input, file)) {
-		g_warning("failed to open %s\n", file);
-		return NULL;
-	}
-
-	ctx.decoder = NULL;
-	ctx.tag = tag_new();
-
-	ret = ffmpeg_helper(file, NULL, &input, ffmpeg_tag_internal, &ctx);
-	if (!ret) {
-		tag_free(ctx.tag);
-		ctx.tag = NULL;
-	}
 
-	input_stream_close(&input);
+	av_close_input_stream(f);
+	mpd_ffmpeg_stream_close(stream);
 
-	return ctx.tag;
+	return tag;
 }
 
 /**
@@ -592,6 +615,12 @@ static const char *const ffmpeg_mime_types[] = {
 	"video/x-vid",
 	"video/x-wmv",
 	"video/x-xvid",
+
+	/* special value for the "ffmpeg" input plugin: all streams by
+	   the "ffmpeg" input plugin shall be decoded by this
+	   plugin */
+	"audio/x-mpd-ffmpeg",
+
 	NULL
 };
 
@@ -599,7 +628,7 @@ const struct decoder_plugin ffmpeg_decoder_plugin = {
 	.name = "ffmpeg",
 	.init = ffmpeg_init,
 	.stream_decode = ffmpeg_decode,
-	.tag_dup = ffmpeg_tag,
+	.stream_tag = ffmpeg_stream_tag,
 	.suffixes = ffmpeg_suffixes,
 	.mime_types = ffmpeg_mime_types
 };
diff --git a/src/decoder/flac_compat.h b/src/decoder/flac_compat.h
new file mode 100644
index 000000000..d597690a0
--- /dev/null
+++ b/src/decoder/flac_compat.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+/*
+ * Common data structures and functions used by FLAC and OggFLAC
+ */
+
+#ifndef MPD_FLAC_COMPAT_H
+#define MPD_FLAC_COMPAT_H
+
+#include <FLAC/export.h>
+#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+#  include <FLAC/seekable_stream_decoder.h>
+
+/* starting with libFLAC 1.1.3, the SeekableStreamDecoder has been
+   merged into the StreamDecoder.  The following macros try to emulate
+   the new API for libFLAC 1.1.2 by mapping MPD's StreamDecoder calls
+   to the old SeekableStreamDecoder API. */
+
+#define FLAC__StreamDecoder FLAC__SeekableStreamDecoder
+#define FLAC__stream_decoder_new FLAC__seekable_stream_decoder_new
+#define FLAC__stream_decoder_get_decode_position FLAC__seekable_stream_decoder_get_decode_position
+#define FLAC__stream_decoder_get_state FLAC__seekable_stream_decoder_get_state
+#define FLAC__stream_decoder_process_single FLAC__seekable_stream_decoder_process_single
+#define FLAC__stream_decoder_process_until_end_of_metadata FLAC__seekable_stream_decoder_process_until_end_of_metadata
+#define FLAC__stream_decoder_seek_absolute FLAC__seekable_stream_decoder_seek_absolute
+#define FLAC__stream_decoder_finish FLAC__seekable_stream_decoder_finish
+#define FLAC__stream_decoder_delete FLAC__seekable_stream_decoder_delete
+
+#define FLAC__STREAM_DECODER_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM
+
+typedef unsigned flac_read_status_size_t;
+
+#define FLAC__StreamDecoderReadStatus FLAC__SeekableStreamDecoderReadStatus
+#define FLAC__STREAM_DECODER_READ_STATUS_CONTINUE FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK
+#define FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK
+#define FLAC__STREAM_DECODER_READ_STATUS_ABORT FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR
+
+#define FLAC__StreamDecoderSeekStatus FLAC__SeekableStreamDecoderSeekStatus
+#define FLAC__STREAM_DECODER_SEEK_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK
+#define FLAC__STREAM_DECODER_SEEK_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR
+#define FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR
+
+#define FLAC__StreamDecoderTellStatus FLAC__SeekableStreamDecoderTellStatus
+#define FLAC__STREAM_DECODER_TELL_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK
+#define FLAC__STREAM_DECODER_TELL_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
+#define FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
+
+#define FLAC__StreamDecoderLengthStatus FLAC__SeekableStreamDecoderLengthStatus
+#define FLAC__STREAM_DECODER_LENGTH_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK
+#define FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
+#define FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
+
+typedef enum {
+	FLAC__STREAM_DECODER_INIT_STATUS_OK,
+	FLAC__STREAM_DECODER_INIT_STATUS_ERROR,
+} FLAC__StreamDecoderInitStatus;
+
+static inline FLAC__StreamDecoderInitStatus
+FLAC__stream_decoder_init_stream(FLAC__SeekableStreamDecoder *decoder,
+				 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)
+{
+	return FLAC__seekable_stream_decoder_set_read_callback(decoder, read_cb) &&
+		FLAC__seekable_stream_decoder_set_seek_callback(decoder, seek_cb) &&
+		FLAC__seekable_stream_decoder_set_tell_callback(decoder, tell_cb) &&
+		FLAC__seekable_stream_decoder_set_length_callback(decoder, length_cb) &&
+		FLAC__seekable_stream_decoder_set_eof_callback(decoder, eof_cb) &&
+		FLAC__seekable_stream_decoder_set_write_callback(decoder, write_cb) &&
+		FLAC__seekable_stream_decoder_set_metadata_callback(decoder, metadata_cb) &&
+		FLAC__seekable_stream_decoder_set_metadata_respond(decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT) &&
+		FLAC__seekable_stream_decoder_set_error_callback(decoder, error_cb) &&
+		FLAC__seekable_stream_decoder_set_client_data(decoder, data) &&
+		FLAC__seekable_stream_decoder_init(decoder) == FLAC__SEEKABLE_STREAM_DECODER_OK
+		? FLAC__STREAM_DECODER_INIT_STATUS_OK
+		: FLAC__STREAM_DECODER_INIT_STATUS_ERROR;
+}
+
+#else /* FLAC_API_VERSION_CURRENT > 7 */
+
+#  include <FLAC/stream_decoder.h>
+
+#  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)
+
+typedef size_t flac_read_status_size_t;
+
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+#endif /* _FLAC_COMMON_H */
diff --git a/src/decoder/flac_decoder_plugin.c b/src/decoder/flac_decoder_plugin.c
new file mode 100644
index 000000000..e89e2ea11
--- /dev/null
+++ b/src/decoder/flac_decoder_plugin.c
@@ -0,0 +1,497 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h" /* must be first for large file support */
+#include "_flac_common.h"
+#include "flac_compat.h"
+#include "flac_metadata.h"
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+#include "_ogg_common.h"
+#endif
+
+#include <glib.h>
+
+#include <assert.h>
+#include <unistd.h>
+
+#include <sys/stat.h>
+#include <sys/types.h>
+
+/* this code was based on flac123, from flac-tools */
+
+static FLAC__StreamDecoderReadStatus
+flac_read_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd,
+	     FLAC__byte buf[], flac_read_status_size_t *bytes,
+	     void *fdata)
+{
+	struct flac_data *data = fdata;
+	size_t r;
+
+	r = decoder_read(data->decoder, data->input_stream,
+			 (void *)buf, *bytes);
+	*bytes = r;
+
+	if (r == 0) {
+		if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE ||
+		    input_stream_eof(data->input_stream))
+			return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
+		else
+			return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
+	}
+
+	return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+}
+
+static FLAC__StreamDecoderSeekStatus
+flac_seek_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd,
+	     FLAC__uint64 offset, void *fdata)
+{
+	struct flac_data *data = (struct flac_data *) fdata;
+
+	if (!data->input_stream->seekable)
+		return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED;
+
+	if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL))
+		return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR;
+
+	return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
+}
+
+static FLAC__StreamDecoderTellStatus
+flac_tell_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd,
+	     FLAC__uint64 * offset, void *fdata)
+{
+	struct flac_data *data = (struct flac_data *) fdata;
+
+	if (!data->input_stream->seekable)
+		return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED;
+
+	*offset = (long)(data->input_stream->offset);
+
+	return FLAC__STREAM_DECODER_TELL_STATUS_OK;
+}
+
+static FLAC__StreamDecoderLengthStatus
+flac_length_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd,
+	       FLAC__uint64 * length, void *fdata)
+{
+	struct flac_data *data = (struct flac_data *) fdata;
+
+	if (data->input_stream->size < 0)
+		return FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED;
+
+	*length = (size_t) (data->input_stream->size);
+
+	return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
+}
+
+static FLAC__bool
+flac_eof_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, void *fdata)
+{
+	struct flac_data *data = (struct flac_data *) fdata;
+
+	return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE &&
+		decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) ||
+		input_stream_eof(data->input_stream);
+}
+
+static void
+flac_error_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd,
+	      FLAC__StreamDecoderErrorStatus status, void *fdata)
+{
+	flac_error_common_cb("flac", status, (struct flac_data *) 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";
+	}
+
+	g_warning("%s\n", str);
+}
+#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";
+	}
+
+	g_warning("%s\n", str);
+}
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+static void flacMetadata(G_GNUC_UNUSED const FLAC__StreamDecoder * dec,
+			 const FLAC__StreamMetadata * block, void *vdata)
+{
+	flac_metadata_common_cb(block, (struct flac_data *) vdata);
+}
+
+static FLAC__StreamDecoderWriteStatus
+flac_write_cb(const FLAC__StreamDecoder *dec, const FLAC__Frame *frame,
+	      const FLAC__int32 *const buf[], void *vdata)
+{
+	struct flac_data *data = (struct flac_data *) vdata;
+	FLAC__uint64 nbytes = 0;
+
+	if (FLAC__stream_decoder_get_decode_position(dec, &nbytes)) {
+		if (data->position > 0 && nbytes > data->position) {
+			nbytes -= data->position;
+			data->position += nbytes;
+		} else {
+			data->position = nbytes;
+			nbytes = 0;
+		}
+	} else
+		nbytes = 0;
+
+	return flac_common_write(data, frame, buf, nbytes);
+}
+
+static struct tag *
+flac_tag_dup(const char *file)
+{
+	return flac_tag_load(file, NULL);
+}
+
+/**
+ * Some glue code around FLAC__stream_decoder_new().
+ */
+static FLAC__StreamDecoder *
+flac_decoder_new(void)
+{
+	FLAC__StreamDecoder *sd = FLAC__stream_decoder_new();
+	if (sd == NULL) {
+		g_warning("FLAC__stream_decoder_new() failed");
+		return NULL;
+	}
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+	if(!FLAC__stream_decoder_set_metadata_respond(sd, FLAC__METADATA_TYPE_VORBIS_COMMENT))
+		g_debug("FLAC__stream_decoder_set_metadata_respond() has failed");
+#endif
+
+	return sd;
+}
+
+static bool
+flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd,
+			FLAC__uint64 duration)
+{
+	data->total_frames = duration;
+
+	if (!FLAC__stream_decoder_process_until_end_of_metadata(sd)) {
+		g_warning("problem reading metadata");
+		return false;
+	}
+
+	if (data->initialized) {
+		/* done */
+		decoder_initialized(data->decoder, &data->audio_format,
+				    data->input_stream->seekable,
+				    (float)data->total_frames /
+				    (float)data->audio_format.sample_rate);
+		return true;
+	}
+
+	if (data->input_stream->seekable)
+		/* allow the workaround below only for nonseekable
+		   streams*/
+		return false;
+
+	/* no stream_info packet found; try to initialize the decoder
+	   from the first frame header */
+	FLAC__stream_decoder_process_single(sd);
+	return data->initialized;
+}
+
+static void
+flac_decoder_loop(struct flac_data *data, FLAC__StreamDecoder *flac_dec,
+		  FLAC__uint64 t_start, FLAC__uint64 t_end)
+{
+	struct decoder *decoder = data->decoder;
+	enum decoder_command cmd;
+
+	data->first_frame = t_start;
+
+	while (true) {
+		if (data->tag != NULL && !tag_is_empty(data->tag)) {
+			cmd = decoder_tag(data->decoder, data->input_stream,
+					  data->tag);
+			tag_free(data->tag);
+			data->tag = tag_new();
+		} else
+			cmd = decoder_get_command(decoder);
+
+		if (cmd == DECODE_COMMAND_SEEK) {
+			FLAC__uint64 seek_sample = t_start +
+				decoder_seek_where(decoder) *
+				data->audio_format.sample_rate;
+			if (seek_sample >= t_start &&
+			    (t_end == 0 || seek_sample <= t_end) &&
+			    FLAC__stream_decoder_seek_absolute(flac_dec, seek_sample)) {
+				data->next_frame = seek_sample;
+				data->position = 0;
+				decoder_command_finished(decoder);
+			} else
+				decoder_seek_error(decoder);
+		} else if (cmd == DECODE_COMMAND_STOP ||
+			   FLAC__stream_decoder_get_state(flac_dec) == FLAC__STREAM_DECODER_END_OF_STREAM)
+			break;
+
+		if (t_end != 0 && data->next_frame >= t_end)
+			/* end of this sub track */
+			break;
+
+		if (!FLAC__stream_decoder_process_single(flac_dec)) {
+			cmd = decoder_get_command(decoder);
+			if (cmd != DECODE_COMMAND_SEEK)
+				break;
+		}
+	}
+
+	if (cmd != DECODE_COMMAND_STOP) {
+		flacPrintErroredState(FLAC__stream_decoder_get_state(flac_dec));
+		FLAC__stream_decoder_finish(flac_dec);
+	}
+}
+
+static void
+flac_decode_internal(struct decoder * decoder,
+		     struct input_stream *input_stream,
+		     bool is_ogg)
+{
+	FLAC__StreamDecoder *flac_dec;
+	struct flac_data data;
+	const char *err = NULL;
+
+	flac_dec = flac_decoder_new();
+	if (flac_dec == NULL)
+		return;
+
+	flac_data_init(&data, decoder, input_stream);
+	data.tag = tag_new();
+
+	if (is_ogg) {
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+		FLAC__StreamDecoderInitStatus status =
+			FLAC__stream_decoder_init_ogg_stream(flac_dec,
+							     flac_read_cb,
+							     flac_seek_cb,
+							     flac_tell_cb,
+							     flac_length_cb,
+							     flac_eof_cb,
+							     flac_write_cb,
+							     flacMetadata,
+							     flac_error_cb,
+							     (void *)&data);
+		if (status != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
+			err = "doing Ogg init()";
+			goto fail;
+		}
+#else
+		goto fail;
+#endif
+	} else {
+		FLAC__StreamDecoderInitStatus status =
+			FLAC__stream_decoder_init_stream(flac_dec,
+							 flac_read_cb,
+							 flac_seek_cb,
+							 flac_tell_cb,
+							 flac_length_cb,
+							 flac_eof_cb,
+							 flac_write_cb,
+							 flacMetadata,
+							 flac_error_cb,
+							 (void *)&data);
+		if (status != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
+			err = "doing init()";
+			goto fail;
+		}
+	}
+
+	if (!flac_decoder_initialize(&data, flac_dec, 0)) {
+		flac_data_deinit(&data);
+		FLAC__stream_decoder_delete(flac_dec);
+		return;
+	}
+
+	flac_decoder_loop(&data, flac_dec, 0, 0);
+
+fail:
+	flac_data_deinit(&data);
+	FLAC__stream_decoder_delete(flac_dec);
+
+	if (err)
+		g_warning("%s\n", err);
+}
+
+static void
+flac_decode(struct decoder * decoder, struct input_stream *input_stream)
+{
+	flac_decode_internal(decoder, input_stream, false);
+}
+
+#ifndef HAVE_OGGFLAC
+
+static bool
+oggflac_init(G_GNUC_UNUSED const struct config_param *param)
+{
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+	return !!FLAC_API_SUPPORTS_OGG_FLAC;
+#else
+	/* disable oggflac when libflac is too old */
+	return false;
+#endif
+}
+
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+
+static struct tag *
+oggflac_tag_dup(const 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);
+
+	ret = tag_new();
+	do {
+		if (!(block = FLAC__metadata_iterator_get_block(it)))
+			break;
+
+		flac_tag_apply_metadata(ret, NULL, block);
+	} while (FLAC__metadata_iterator_next(it));
+	FLAC__metadata_iterator_delete(it);
+
+	if (!tag_is_defined(ret)) {
+		tag_free(ret);
+		ret = NULL;
+	}
+
+out:
+	FLAC__metadata_chain_delete(chain);
+	return ret;
+}
+
+static void
+oggflac_decode(struct decoder *decoder, struct input_stream *input_stream)
+{
+	if (ogg_stream_type_detect(input_stream) != FLAC)
+		return;
+
+	/* rewind the stream, because ogg_stream_type_detect() has
+	   moved it */
+	input_stream_seek(input_stream, 0, SEEK_SET, NULL);
+
+	flac_decode_internal(decoder, input_stream, true);
+}
+
+static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL };
+static const char *const oggflac_mime_types[] = {
+	"application/ogg",
+	"application/x-ogg",
+	"audio/ogg",
+	"audio/x-flac+ogg",
+	"audio/x-ogg",
+	NULL
+};
+
+#endif /* FLAC_API_VERSION_CURRENT >= 7 */
+
+const struct decoder_plugin oggflac_decoder_plugin = {
+	.name = "oggflac",
+	.init = oggflac_init,
+#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
+	.stream_decode = oggflac_decode,
+	.tag_dup = oggflac_tag_dup,
+	.suffixes = oggflac_suffixes,
+	.mime_types = oggflac_mime_types
+#endif
+};
+
+#endif /* HAVE_OGGFLAC */
+
+static const char *const flac_suffixes[] = { "flac", NULL };
+static const char *const flac_mime_types[] = {
+	"application/flac",
+	"application/x-flac",
+	"audio/flac",
+	"audio/x-flac",
+	NULL
+};
+
+const struct decoder_plugin flac_decoder_plugin = {
+	.name = "flac",
+	.stream_decode = flac_decode,
+	.tag_dup = flac_tag_dup,
+	.suffixes = flac_suffixes,
+	.mime_types = flac_mime_types,
+};
diff --git a/src/decoder/flac_metadata.c b/src/decoder/flac_metadata.c
new file mode 100644
index 000000000..f2f2f954d
--- /dev/null
+++ b/src/decoder/flac_metadata.c
@@ -0,0 +1,286 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "flac_metadata.h"
+#include "replay_gain_info.h"
+#include "tag.h"
+
+#include <glib.h>
+
+#include <assert.h>
+#include <stdbool.h>
+#include <stdlib.h>
+
+static bool
+flac_find_float_comment(const FLAC__StreamMetadata *block,
+			const char *cmnt, float *fl)
+{
+	int offset;
+	size_t pos;
+	int len;
+	unsigned char tmp, *p;
+
+	offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0,
+								     cmnt);
+	if (offset < 0)
+		return false;
+
+	pos = strlen(cmnt) + 1; /* 1 is for '=' */
+	len = block->data.vorbis_comment.comments[offset].length - pos;
+	if (len <= 0)
+		return false;
+
+	p = &block->data.vorbis_comment.comments[offset].entry[pos];
+	tmp = p[len];
+	p[len] = '\0';
+	*fl = (float)atof((char *)p);
+	p[len] = tmp;
+
+	return true;
+}
+
+bool
+flac_parse_replay_gain(struct replay_gain_info *rgi,
+		       const FLAC__StreamMetadata *block)
+{
+	bool found = false;
+
+	replay_gain_info_init(rgi);
+
+	if (flac_find_float_comment(block, "replaygain_album_gain",
+				    &rgi->tuples[REPLAY_GAIN_ALBUM].gain))
+		found = true;
+	if (flac_find_float_comment(block, "replaygain_album_peak",
+				    &rgi->tuples[REPLAY_GAIN_ALBUM].peak))
+		found = true;
+	if (flac_find_float_comment(block, "replaygain_track_gain",
+				    &rgi->tuples[REPLAY_GAIN_TRACK].gain))
+		found = true;
+	if (flac_find_float_comment(block, "replaygain_track_peak",
+				    &rgi->tuples[REPLAY_GAIN_TRACK].peak))
+		found = true;
+
+	return found;
+}
+
+static bool
+flac_find_string_comment(const FLAC__StreamMetadata *block,
+			 const char *cmnt, char **str)
+{
+	int offset;
+	size_t pos;
+	int len;
+	const unsigned char *p;
+
+	*str = NULL;
+	offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0,
+								     cmnt);
+	if (offset < 0)
+		return false;
+
+	pos = strlen(cmnt) + 1; /* 1 is for '=' */
+	len = block->data.vorbis_comment.comments[offset].length - pos;
+	if (len <= 0)
+		return false;
+
+	p = &block->data.vorbis_comment.comments[offset].entry[pos];
+	*str = g_strndup((const char *)p, len);
+
+	return true;
+}
+
+bool
+flac_parse_mixramp(char **mixramp_start, char **mixramp_end,
+		   const FLAC__StreamMetadata *block)
+{
+	bool found = false;
+
+	if (flac_find_string_comment(block, "mixramp_start", mixramp_start))
+		found = true;
+	if (flac_find_string_comment(block, "mixramp_end", mixramp_end))
+		found = true;
+
+	return found;
+}
+
+/**
+ * Checks if the specified name matches the entry's name, and if yes,
+ * returns the comment value (not null-temrinated).
+ */
+static const char *
+flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry,
+		   const char *name, const char *char_tnum, size_t *length_r)
+{
+	size_t name_length = strlen(name);
+	size_t char_tnum_length = 0;
+	const char *comment = (const char*)entry->entry;
+
+	if (entry->length <= name_length ||
+	    g_ascii_strncasecmp(comment, name, name_length) != 0)
+		return NULL;
+
+	if (char_tnum != NULL) {
+		char_tnum_length = strlen(char_tnum);
+		if (entry->length > name_length + char_tnum_length + 2 &&
+		    comment[name_length] == '[' &&
+		    g_ascii_strncasecmp(comment + name_length + 1,
+					char_tnum, char_tnum_length) == 0 &&
+		    comment[name_length + char_tnum_length + 1] == ']')
+			name_length = name_length + char_tnum_length + 2;
+		else if (entry->length > name_length + char_tnum_length &&
+			 g_ascii_strncasecmp(comment + name_length,
+					     char_tnum, char_tnum_length) == 0)
+			name_length = name_length + char_tnum_length;
+	}
+
+	if (comment[name_length] == '=') {
+		*length_r = entry->length - name_length - 1;
+		return comment + name_length + 1;
+	}
+
+	return NULL;
+}
+
+/**
+ * Check if the comment's name equals the passed name, and if so, copy
+ * the comment value into the tag.
+ */
+static bool
+flac_copy_comment(struct tag *tag,
+		  const FLAC__StreamMetadata_VorbisComment_Entry *entry,
+		  const char *name, enum tag_type tag_type,
+		  const char *char_tnum)
+{
+	const char *value;
+	size_t value_length;
+
+	value = flac_comment_value(entry, name, char_tnum, &value_length);
+	if (value != NULL) {
+		tag_add_item_n(tag, tag_type, value, value_length);
+		return true;
+	}
+
+	return false;
+}
+
+/* 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 void
+flac_parse_comment(struct tag *tag, const char *char_tnum,
+		   const FLAC__StreamMetadata_VorbisComment_Entry *entry)
+{
+	assert(tag != NULL);
+
+	if (flac_copy_comment(tag, entry, VORBIS_COMMENT_TRACK_KEY,
+			      TAG_TRACK, char_tnum) ||
+	    flac_copy_comment(tag, entry, VORBIS_COMMENT_DISC_KEY,
+			      TAG_DISC, char_tnum) ||
+	    flac_copy_comment(tag, entry, "album artist",
+			      TAG_ALBUM_ARTIST, char_tnum))
+		return;
+
+	for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i)
+		if (flac_copy_comment(tag, entry,
+				      tag_item_names[i], i, char_tnum))
+			return;
+}
+
+void
+flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
+			    const FLAC__StreamMetadata_VorbisComment *comment)
+{
+	for (unsigned i = 0; i < comment->num_comments; ++i)
+		flac_parse_comment(tag, char_tnum, &comment->comments[i]);
+}
+
+void
+flac_tag_apply_metadata(struct tag *tag, const char *track,
+			const FLAC__StreamMetadata *block)
+{
+	switch (block->type) {
+	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+		flac_vorbis_comments_to_tag(tag, track,
+					    &block->data.vorbis_comment);
+		break;
+
+	case FLAC__METADATA_TYPE_STREAMINFO:
+		tag->time = flac_duration(&block->data.stream_info);
+		break;
+
+	default:
+		break;
+	}
+}
+
+struct tag *
+flac_tag_load(const char *file, const char *char_tnum)
+{
+	struct tag *tag;
+	FLAC__Metadata_SimpleIterator *it;
+	FLAC__StreamMetadata *block = NULL;
+
+	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];
+		}
+		g_debug("Reading '%s' metadata gave the following error: %s\n",
+			file, err);
+		FLAC__metadata_simple_iterator_delete(it);
+		return NULL;
+	}
+
+	tag = tag_new();
+	do {
+		block = FLAC__metadata_simple_iterator_get_block(it);
+		if (!block)
+			break;
+
+		flac_tag_apply_metadata(tag, char_tnum, block);
+		FLAC__metadata_object_delete(block);
+	} while (FLAC__metadata_simple_iterator_next(it));
+
+	FLAC__metadata_simple_iterator_delete(it);
+
+	if (!tag_is_defined(tag)) {
+		tag_free(tag);
+		tag = NULL;
+	}
+
+	return tag;
+}
diff --git a/src/decoder/flac_metadata.h b/src/decoder/flac_metadata.h
new file mode 100644
index 000000000..06e691d1d
--- /dev/null
+++ b/src/decoder/flac_metadata.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_FLAC_METADATA_H
+#define MPD_FLAC_METADATA_H
+
+#include <stdbool.h>
+#include <FLAC/metadata.h>
+
+struct tag;
+struct replay_gain_info;
+
+static inline unsigned
+flac_duration(const FLAC__StreamMetadata_StreamInfo *stream_info)
+{
+	return (stream_info->total_samples + stream_info->sample_rate - 1) /
+		stream_info->sample_rate;
+}
+
+bool
+flac_parse_replay_gain(struct replay_gain_info *rgi,
+		       const FLAC__StreamMetadata *block);
+
+bool
+flac_parse_mixramp(char **mixramp_start, char **mixramp_end,
+		   const FLAC__StreamMetadata *block);
+
+void
+flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum,
+			    const FLAC__StreamMetadata_VorbisComment *comment);
+
+void
+flac_tag_apply_metadata(struct tag *tag, const char *track,
+			const FLAC__StreamMetadata *block);
+
+struct tag *
+flac_tag_load(const char *file, const char *char_tnum);
+
+#endif
diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c
new file mode 100644
index 000000000..bf6e2612c
--- /dev/null
+++ b/src/decoder/flac_pcm.c
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "flac_pcm.h"
+
+#include <assert.h>
+
+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];
+}
+
+void
+flac_convert(void *dest,
+	     unsigned int num_channels, enum sample_format sample_format,
+	     const FLAC__int32 *const buf[],
+	     unsigned int position, unsigned int end)
+{
+	switch (sample_format) {
+	case SAMPLE_FORMAT_S16:
+		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 SAMPLE_FORMAT_S24_P32:
+	case SAMPLE_FORMAT_S32:
+		flac_convert_32((int32_t*)dest, num_channels, buf,
+				position, end);
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		flac_convert_8((int8_t*)dest, num_channels, buf,
+			       position, end);
+		break;
+
+	case SAMPLE_FORMAT_S24:
+	case SAMPLE_FORMAT_UNDEFINED:
+		/* unreachable */
+		assert(false);
+	}
+}
diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h
new file mode 100644
index 000000000..bccfc645c
--- /dev/null
+++ b/src/decoder/flac_pcm.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_FLAC_PCM_H
+#define MPD_FLAC_PCM_H
+
+#include "audio_format.h"
+
+#include <FLAC/ordinals.h>
+
+void
+flac_convert(void *dest,
+	     unsigned int num_channels, enum sample_format sample_format,
+	     const FLAC__int32 *const buf[],
+	     unsigned int position, unsigned int end);
+
+#endif
diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c
deleted file mode 100644
index 1e568f70d..000000000
--- a/src/decoder/flac_plugin.c
+++ /dev/null
@@ -1,918 +0,0 @@
-/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
- * 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.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include "_flac_common.h"
-
-#include <glib.h>
-
-#include <assert.h>
-#include <unistd.h>
-
-#include <sys/stat.h>
-#include <sys/types.h>
-
-#ifdef HAVE_CUE /* libcue */
-#include "../cue/cue_tag.h"
-#endif
-
-/* this code was based on flac123, from flac-tools */
-
-static flac_read_status
-flac_read_cb(G_GNUC_UNUSED const flac_decoder *fd,
-	     FLAC__byte buf[], flac_read_status_size_t *bytes,
-	     void *fdata)
-{
-	struct flac_data *data = fdata;
-	size_t r;
-
-	r = decoder_read(data->decoder, data->input_stream,
-			 (void *)buf, *bytes);
-	*bytes = r;
-
-	if (r == 0) {
-		if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE ||
-		    input_stream_eof(data->input_stream))
-			return flac_read_status_eof;
-		else
-			return flac_read_status_abort;
-	}
-
-	return flac_read_status_continue;
-}
-
-static flac_seek_status
-flac_seek_cb(G_GNUC_UNUSED const flac_decoder *fd,
-	     FLAC__uint64 offset, void *fdata)
-{
-	struct flac_data *data = (struct flac_data *) fdata;
-
-	if (!input_stream_seek(data->input_stream, offset, SEEK_SET))
-		return flac_seek_status_error;
-
-	return flac_seek_status_ok;
-}
-
-static flac_tell_status
-flac_tell_cb(G_GNUC_UNUSED const flac_decoder *fd,
-	     FLAC__uint64 * offset, void *fdata)
-{
-	struct flac_data *data = (struct flac_data *) fdata;
-
-	*offset = (long)(data->input_stream->offset);
-
-	return flac_tell_status_ok;
-}
-
-static flac_length_status
-flac_length_cb(G_GNUC_UNUSED const flac_decoder *fd,
-	       FLAC__uint64 * length, void *fdata)
-{
-	struct flac_data *data = (struct flac_data *) fdata;
-
-	if (data->input_stream->size < 0)
-		return flac_length_status_unsupported;
-
-	*length = (size_t) (data->input_stream->size);
-
-	return flac_length_status_ok;
-}
-
-static FLAC__bool
-flac_eof_cb(G_GNUC_UNUSED const flac_decoder *fd, void *fdata)
-{
-	struct flac_data *data = (struct flac_data *) fdata;
-
-	return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE &&
-		decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) ||
-		input_stream_eof(data->input_stream);
-}
-
-static void
-flac_error_cb(G_GNUC_UNUSED const flac_decoder *fd,
-	      FLAC__StreamDecoderErrorStatus status, void *fdata)
-{
-	flac_error_common_cb("flac", status, (struct flac_data *) 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";
-	}
-
-	g_warning("%s\n", str);
-}
-
-static bool
-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)
-{
-	return FLAC__seekable_stream_decoder_set_read_callback(dec, read_cb) &&
-		FLAC__seekable_stream_decoder_set_seek_callback(dec, seek_cb) &&
-		FLAC__seekable_stream_decoder_set_tell_callback(dec, tell_cb) &&
-		FLAC__seekable_stream_decoder_set_length_callback(dec, length_cb) &&
-		FLAC__seekable_stream_decoder_set_eof_callback(dec, eof_cb) &&
-		FLAC__seekable_stream_decoder_set_write_callback(dec, write_cb) &&
-		FLAC__seekable_stream_decoder_set_metadata_callback(dec, metadata_cb) &&
-		FLAC__seekable_stream_decoder_set_metadata_respond(dec, FLAC__METADATA_TYPE_VORBIS_COMMENT) &&
-		FLAC__seekable_stream_decoder_set_error_callback(dec, error_cb) &&
-		FLAC__seekable_stream_decoder_set_client_data(dec, data) &&
-		FLAC__seekable_stream_decoder_init(dec) == FLAC__SEEKABLE_STREAM_DECODER_OK;
-}
-#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";
-	}
-
-	g_warning("%s\n", str);
-}
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-
-static void flacMetadata(G_GNUC_UNUSED const flac_decoder * dec,
-			 const FLAC__StreamMetadata * block, void *vdata)
-{
-	flac_metadata_common_cb(block, (struct flac_data *) vdata);
-}
-
-static FLAC__StreamDecoderWriteStatus
-flac_write_cb(const flac_decoder *dec, const FLAC__Frame *frame,
-	      const FLAC__int32 *const buf[], void *vdata)
-{
-	FLAC__uint32 samples = frame->header.blocksize;
-	struct flac_data *data = (struct flac_data *) 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->bit_rate =
-		    ((newPosition - data->position) * 8.0 / timeChange)
-		    / 1000 + 0.5;
-	}
-	data->position = newPosition;
-
-	return flac_common_write(data, frame, buf);
-}
-
-static struct tag *
-flac_tag_load(const char *file, const char *char_tnum)
-{
-	struct tag *tag;
-	FLAC__Metadata_SimpleIterator *it;
-	FLAC__StreamMetadata *block = NULL;
-
-	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];
-		}
-		g_debug("Reading '%s' metadata gave the following error: %s\n",
-			file, err);
-		FLAC__metadata_simple_iterator_delete(it);
-		return NULL;
-	}
-
-	tag = tag_new();
-	do {
-		block = FLAC__metadata_simple_iterator_get_block(it);
-		if (!block)
-			break;
-		if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
-			flac_vorbis_comments_to_tag(tag, char_tnum, block);
-		} else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
-			tag->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);
-
-	if (!tag_is_defined(tag)) {
-		tag_free(tag);
-		tag = NULL;
-	}
-
-	return tag;
-}
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-
-static struct tag *
-flac_cue_tag_load(const char *file)
-{
-	struct tag* tag = NULL;
-	char* char_tnum = NULL;
-	char* ptr = NULL;
-	unsigned int tnum = 0;
-	unsigned int sample_rate = 0;
-	FLAC__uint64 track_time = 0;
-#ifdef HAVE_CUE /* libcue */
-	FLAC__StreamMetadata* vc;
-#endif /* libcue */
-	FLAC__StreamMetadata* si = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO);
-	FLAC__StreamMetadata* cs;
-
-	tnum = flac_vtrack_tnum(file);
-	char_tnum = g_strdup_printf("%u", tnum);
-
-	ptr = strrchr(file, '/');
-	*ptr = '\0';
-
-#ifdef HAVE_CUE /* libcue */
-	if (FLAC__metadata_get_tags(file, &vc))
-	{
-		for (unsigned i = 0; i < vc->data.vorbis_comment.num_comments;
-		     i++)
-		{
-			if ((ptr = (char*)vc->data.vorbis_comment.comments[i].entry) != NULL)
-			{
-				if (g_ascii_strncasecmp(ptr, "cuesheet", 8) == 0)
-				{
-					while (*(++ptr) != '=');
-					tag = cue_tag_string(   ++ptr,
-								tnum);
-				}
-			}
-		}
-
-		FLAC__metadata_object_delete(vc);
-	}
-#endif /* libcue */
-
-	if (tag == NULL)
-		tag = flac_tag_load(file, char_tnum);
-
-	if (char_tnum != NULL)
-	{
-		tag_add_item(	tag,
-				TAG_ITEM_TRACK,
-				char_tnum);
-		g_free(char_tnum);
-	}
-
-	if (FLAC__metadata_get_streaminfo(file, si))
-	{
-		sample_rate = si->data.stream_info.sample_rate;
-		FLAC__metadata_object_delete(si);
-	}
-
-	if (FLAC__metadata_get_cuesheet(file, &cs))
-	{
-		if (cs->data.cue_sheet.tracks != NULL
-				&& (tnum <= cs->data.cue_sheet.num_tracks - 1))
-		{
-			track_time = cs->data.cue_sheet.tracks[tnum].offset
-				- cs->data.cue_sheet.tracks[tnum - 1].offset;
-		}
-		FLAC__metadata_object_delete(cs);
-	}
-
-	if (sample_rate != 0)
-	{
-		tag->time = (unsigned int)(track_time/sample_rate);
-	}
-
-	return tag;
-}
-
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-
-static struct tag *
-flac_tag_dup(const char *file)
-{
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-	struct stat st;
-
-	if (stat(file, &st) < 0)
-		return flac_cue_tag_load(file);
-	else
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-		return flac_tag_load(file, NULL);
-}
-
-static void
-flac_decode_internal(struct decoder * decoder,
-		     struct input_stream *input_stream,
-		     bool is_ogg)
-{
-	flac_decoder *flac_dec;
-	struct flac_data data;
-	enum decoder_command cmd;
-	const char *err = NULL;
-
-	if (!(flac_dec = flac_new()))
-		return;
-	flac_data_init(&data, decoder, input_stream);
-	data.tag = tag_new();
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-        if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
-        {
-                g_debug("Failed to set metadata respond\n");
-        }
-#endif
-
-	if (is_ogg) {
-		if (!flac_ogg_init(flac_dec, flac_read_cb,
-				   flac_seek_cb, flac_tell_cb,
-				   flac_length_cb, flac_eof_cb,
-				   flac_write_cb, flacMetadata,
-				   flac_error_cb, (void *)&data)) {
-			err = "doing Ogg init()";
-			goto fail;
-		}
-	} else {
-		if (!flac_init(flac_dec, flac_read_cb,
-			       flac_seek_cb, flac_tell_cb,
-			       flac_length_cb, flac_eof_cb,
-			       flac_write_cb, flacMetadata,
-			       flac_error_cb, (void *)&data)) {
-			err = "doing init()";
-			goto fail;
-		}
-	}
-
-	if (!flac_process_metadata(flac_dec)) {
-		err = "problem reading metadata";
-		goto fail;
-	}
-
-	if (!audio_format_valid(&data.audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  data.audio_format.sample_rate,
-			  data.audio_format.bits,
-			  data.audio_format.channels);
-		goto fail;
-	}
-
-	decoder_initialized(decoder, &data.audio_format,
-			    input_stream->seekable, data.total_time);
-
-	while (true) {
-		if (!tag_is_empty(data.tag)) {
-			cmd = decoder_tag(decoder, input_stream, data.tag);
-			tag_free(data.tag);
-			data.tag = tag_new();
-		} else
-			cmd = decoder_get_command(decoder);
-
-		if (cmd == DECODE_COMMAND_SEEK) {
-			FLAC__uint64 seek_sample = decoder_seek_where(decoder) *
-			    data.audio_format.sample_rate + 0.5;
-			if (flac_seek_absolute(flac_dec, seek_sample)) {
-				data.time = ((float)seek_sample) /
-				    data.audio_format.sample_rate;
-				data.position = 0;
-				decoder_command_finished(decoder);
-			} else
-				decoder_seek_error(decoder);
-		} else if (cmd == DECODE_COMMAND_STOP ||
-			   flac_get_state(flac_dec) == flac_decoder_eof)
-			break;
-
-		if (!flac_process_single(flac_dec)) {
-			cmd = decoder_get_command(decoder);
-			if (cmd != DECODE_COMMAND_SEEK)
-				break;
-		}
-	}
-	if (cmd != DECODE_COMMAND_STOP) {
-		flacPrintErroredState(flac_get_state(flac_dec));
-		flac_finish(flac_dec);
-	}
-
-fail:
-	if (data.replay_gain_info)
-		replay_gain_info_free(data.replay_gain_info);
-
-	tag_free(data.tag);
-
-	if (flac_dec)
-		flac_delete(flac_dec);
-
-	if (err)
-		g_warning("%s\n", err);
-}
-
-static void
-flac_decode(struct decoder * decoder, struct input_stream *input_stream)
-{
-	flac_decode_internal(decoder, input_stream, false);
-}
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-
-/**
- * @brief Decode a flac file with embedded cue sheets
- * @param const char* fname filename on fs
- */
-static void
-flac_container_decode(struct decoder* decoder,
-		     const char* fname,
-		     bool is_ogg)
-{
-	unsigned int tnum = 0;
-	FLAC__uint64 t_start = 0;
-	FLAC__uint64 t_end = 0;
-	FLAC__uint64 track_time = 0;
-	FLAC__StreamMetadata* cs = NULL;
-
-	flac_decoder *flac_dec;
-	struct flac_data data;
-	const char *err = NULL;
-
-	char* pathname = g_strdup(fname);
-	char* slash = strrchr(pathname, '/');
-	*slash = '\0';
-
-	tnum = flac_vtrack_tnum(fname);
-
-	cs = FLAC__metadata_object_new(FLAC__METADATA_TYPE_CUESHEET);
-
-	FLAC__metadata_get_cuesheet(pathname, &cs);
-
-	if (cs != NULL)
-	{
-		if (cs->data.cue_sheet.tracks != NULL
-				&& (tnum <= cs->data.cue_sheet.num_tracks - 1))
-		{
-			t_start = cs->data.cue_sheet.tracks[tnum - 1].offset;
-			t_end = cs->data.cue_sheet.tracks[tnum].offset;
-			track_time = cs->data.cue_sheet.tracks[tnum].offset
-				- cs->data.cue_sheet.tracks[tnum - 1].offset;
-		}
-
-		FLAC__metadata_object_delete(cs);
-	}
-	else
-	{
-		g_free(pathname);
-		return;
-	}
-
-	if (!(flac_dec = flac_new()))
-	{
-		g_free(pathname);
-		return;
-	}
-
-	flac_data_init(&data, decoder, NULL);
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-        if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
-        {
-                g_debug("Failed to set metadata respond\n");
-        }
-#endif
-
-
-	if (is_ogg)
-	{
-		if (FLAC__stream_decoder_init_ogg_file(	flac_dec,
-							pathname,
-							flac_write_cb,
-							flacMetadata,
-							flac_error_cb,
-							(void*) &data	)
-			!= FLAC__STREAM_DECODER_INIT_STATUS_OK		)
-		{
-			err = "doing Ogg init()";
-			goto fail;
-		}
-	}
-	else
-	{
-		if (FLAC__stream_decoder_init_file(	flac_dec,
-							pathname,
-							flac_write_cb,
-							flacMetadata,
-							flac_error_cb,
-							(void*) &data	)
-			!= FLAC__STREAM_DECODER_INIT_STATUS_OK		)
-		{
-			err = "doing init()";
-			goto fail;
-		}
-	}
-
-	if (!flac_process_metadata(flac_dec))
-	{
-		err = "problem reading metadata";
-		goto fail;
-	}
-
-	if (!audio_format_valid(&data.audio_format))
-	{
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  data.audio_format.sample_rate,
-			  data.audio_format.bits,
-			  data.audio_format.channels);
-		goto fail;
-	}
-
-	// set track time (order is important: after stream init)
-	data.total_time = ((float)track_time / (float)data.audio_format.sample_rate);
-	data.position = 0;
-
-	decoder_initialized(decoder, &data.audio_format,
-			    true, data.total_time);
-
-	// seek to song start (order is important: after decoder init)
-	flac_seek_absolute(flac_dec, (FLAC__uint64)t_start);
-
-	while (true)
-	{
-		if (!flac_process_single(flac_dec))
-			break;
-
-		// we only need to break at the end of track if we are in "cue mode"
-		if (data.time >= data.total_time)
-		{
-			flacPrintErroredState(flac_get_state(flac_dec));
-			flac_finish(flac_dec);
-		}
-
-		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK)
-		{
-			FLAC__uint64 seek_sample = t_start +
-				(decoder_seek_where(decoder) * data.audio_format.sample_rate);
-
-			if (seek_sample >= t_start && seek_sample <= t_end &&
-			    flac_seek_absolute(flac_dec, (FLAC__uint64)seek_sample)) {
-				data.time = (float)(seek_sample - t_start) /
-					data.audio_format.sample_rate;
-				data.position = 0;
-
-				decoder_command_finished(decoder);
-			} else
-				decoder_seek_error(decoder);
-		}
-		else if (flac_get_state(flac_dec) == flac_decoder_eof)
-			break;
-	}
-
-	if (decoder_get_command(decoder) != DECODE_COMMAND_STOP)
-	{
-		flacPrintErroredState(flac_get_state(flac_dec));
-		flac_finish(flac_dec);
-	}
-
-fail:
-	if (pathname)
-		g_free(pathname);
-
-	if (data.replay_gain_info)
-		replay_gain_info_free(data.replay_gain_info);
-
-	if (flac_dec)
-		flac_delete(flac_dec);
-
-	if (err)
-		g_warning("%s\n", err);
-}
-
-/**
- * @brief Open a flac file for decoding
- * @param const char* fname filename on fs
- */
-static void
-flac_filedecode_internal(struct decoder* decoder,
-		     const char* fname,
-		     bool is_ogg)
-{
-	flac_decoder *flac_dec;
-	struct flac_data data;
-	const char *err = NULL;
-	unsigned int flac_err_state = 0;
-
-	if (!(flac_dec = flac_new()))
-		return;
-
-	flac_data_init(&data, decoder, NULL);
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-        if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
-        {
-                g_debug("Failed to set metadata respond\n");
-        }
-#endif
-
-
-	if (is_ogg)
-	{
-		if (	(flac_err_state = FLAC__stream_decoder_init_ogg_file(	flac_dec,
-										fname,
-										flac_write_cb,
-										flacMetadata,
-										flac_error_cb,
-										(void*) &data	))
-			== FLAC__STREAM_DECODER_INIT_STATUS_ERROR_OPENING_FILE)
-		{
-			flac_container_decode(decoder, fname, is_ogg);
-		}
-		else if (flac_err_state != FLAC__STREAM_DECODER_INIT_STATUS_OK)
-		{
-			err = "doing Ogg init()";
-			goto fail;
-		}
-	}
-	else
-	{
-		if (	(flac_err_state = FLAC__stream_decoder_init_file(	flac_dec,
-										fname,
-										flac_write_cb,
-										flacMetadata,
-										flac_error_cb,
-										(void*) &data	))
-			== FLAC__STREAM_DECODER_INIT_STATUS_ERROR_OPENING_FILE)
-		{
-			flac_container_decode(decoder, fname, is_ogg);
-		}
-		else if (flac_err_state != FLAC__STREAM_DECODER_INIT_STATUS_OK)
-		{
-			err = "doing init()";
-			goto fail;
-		}
-	}
-
-	if (!flac_process_metadata(flac_dec))
-	{
-		err = "problem reading metadata";
-		goto fail;
-	}
-
-	if (!audio_format_valid(&data.audio_format))
-	{
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  data.audio_format.sample_rate,
-			  data.audio_format.bits,
-			  data.audio_format.channels);
-		goto fail;
-	}
-
-	decoder_initialized(decoder, &data.audio_format,
-			    true, data.total_time);
-
-	while (true)
-	{
-		if (!flac_process_single(flac_dec))
-			break;
-
-		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK)
-		{
-			FLAC__uint64 seek_sample = decoder_seek_where(decoder) *
-			    data.audio_format.sample_rate + 0.5;
-			if (flac_seek_absolute(flac_dec, seek_sample))
-			{
-				data.time = ((float)seek_sample) /
-				    data.audio_format.sample_rate;
-				data.position = 0;
-				decoder_command_finished(decoder);
-			}
-			else
-				decoder_seek_error(decoder);
-
-		}
-		else if (flac_get_state(flac_dec) == flac_decoder_eof)
-			break;
-	}
-
-	if (decoder_get_command(decoder) != DECODE_COMMAND_STOP)
-	{
-		flacPrintErroredState(flac_get_state(flac_dec));
-		flac_finish(flac_dec);
-	}
-
-fail:
-	if (data.replay_gain_info)
-		replay_gain_info_free(data.replay_gain_info);
-
-	if (flac_dec)
-		flac_delete(flac_dec);
-
-	if (err)
-		g_warning("%s\n", err);
-}
-
-/**
- * @brief	wrapper function for
- * 		flac_filedecode_internal method
- * 		for decoding without ogg
- */
-static void
-flac_filedecode(struct decoder *decoder, const char *fname)
-{
-	struct stat st;
-
-	if (stat(fname, &st) < 0) {
-		flac_container_decode(decoder, fname, false);
-	} else 
-		flac_filedecode_internal(decoder, fname, false);
-}
-
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-
-#ifndef HAVE_OGGFLAC
-
-static bool
-oggflac_init(G_GNUC_UNUSED const struct config_param *param)
-{
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-	return !!FLAC_API_SUPPORTS_OGG_FLAC;
-#else
-	/* disable oggflac when libflac is too old */
-	return false;
-#endif
-}
-
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-
-static struct tag *
-oggflac_tag_dup(const 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);
-
-	ret = tag_new();
-	do {
-		if (!(block = FLAC__metadata_iterator_get_block(it)))
-			break;
-		if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
-			flac_vorbis_comments_to_tag(ret, NULL, block);
-		} else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) {
-			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);
-
-	if (!tag_is_defined(ret)) {
-		tag_free(ret);
-		ret = NULL;
-	}
-
-out:
-	FLAC__metadata_chain_delete(chain);
-	return ret;
-}
-
-static void
-oggflac_decode(struct decoder *decoder, struct input_stream *input_stream)
-{
-	if (ogg_stream_type_detect(input_stream) != FLAC)
-		return;
-
-	/* rewind the stream, because ogg_stream_type_detect() has
-	   moved it */
-	input_stream_seek(input_stream, 0, SEEK_SET);
-
-	flac_decode_internal(decoder, input_stream, true);
-}
-
-static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL };
-static const char *const oggflac_mime_types[] = {
-	"application/ogg",
-	"application/x-ogg",
-	"audio/ogg",
-	"audio/x-flac+ogg",
-	"audio/x-ogg",
-	NULL
-};
-
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-
-const struct decoder_plugin oggflac_decoder_plugin = {
-	.name = "oggflac",
-	.init = oggflac_init,
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-	.stream_decode = oggflac_decode,
-	.tag_dup = oggflac_tag_dup,
-	.suffixes = oggflac_suffixes,
-	.mime_types = oggflac_mime_types
-#endif
-};
-
-#endif /* HAVE_OGGFLAC */
-
-static const char *const flac_suffixes[] = { "flac", NULL };
-static const char *const flac_mime_types[] = {
-	"application/flac",
-	"application/x-flac",
-	"audio/flac",
-	"audio/x-flac",
-	NULL
-};
-
-const struct decoder_plugin flac_decoder_plugin = {
-	.name = "flac",
-	.stream_decode = flac_decode,
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-	.file_decode = flac_filedecode,
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-	.tag_dup = flac_tag_dup,
-	.suffixes = flac_suffixes,
-	.mime_types = flac_mime_types,
-#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
-	.container_scan = flac_cue_track,
-#endif /* FLAC_API_VERSION_CURRENT >= 7 */
-};
diff --git a/src/decoder/fluidsynth_plugin.c b/src/decoder/fluidsynth_decoder_plugin.c
index 99c874c09..b9a2d0d99 100644
--- a/src/decoder/fluidsynth_plugin.c
+++ b/src/decoder/fluidsynth_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -26,9 +26,10 @@
  *
  */
 
-#include "../decoder_api.h"
-#include "../timer.h"
-#include "../conf.h"
+#include "config.h"
+#include "decoder_api.h"
+#include "timer.h"
+#include "conf.h"
 
 #include <glib.h>
 
@@ -87,7 +88,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
 {
 	static const struct audio_format audio_format = {
 		.sample_rate = 48000,
-		.bits = 16,
+		.format = SAMPLE_FORMAT_S16,
 		.channels = 2,
 	};
 	char setting_sample_rate[] = "synth.sample-rate";
@@ -203,7 +204,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
 			break;
 
 		cmd = decoder_data(decoder, NULL, buffer, sizeof(buffer),
-				   0, 0, NULL);
+				   0);
 	} while (cmd == DECODE_COMMAND_NONE);
 
 	/* clean up */
diff --git a/src/decoder/gme_decoder_plugin.c b/src/decoder/gme_decoder_plugin.c
new file mode 100644
index 000000000..4a5220a3f
--- /dev/null
+++ b/src/decoder/gme_decoder_plugin.c
@@ -0,0 +1,244 @@
+#include "config.h"
+#include "../decoder_api.h"
+#include "audio_check.h"
+#include "uri.h"
+
+#include <glib.h>
+#include <assert.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <gme/gme.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "gme"
+
+#define SUBTUNE_PREFIX "tune_"
+
+enum {
+	GME_SAMPLE_RATE = 44100,
+	GME_CHANNELS = 2,
+	GME_BUFFER_FRAMES = 2048,
+	GME_BUFFER_SAMPLES = GME_BUFFER_FRAMES * GME_CHANNELS,
+};
+
+/**
+ * returns the file path stripped of any /tune_xxx.* subtune
+ * suffix
+ */
+static char *
+get_container_name(const char *path_fs)
+{
+	const char *subtune_suffix = uri_get_suffix(path_fs);
+	char *path_container = g_strdup(path_fs);
+	char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", subtune_suffix, NULL);
+	GPatternSpec *path_with_subtune = g_pattern_spec_new(pat);
+	g_free(pat);
+	if (!g_pattern_match(path_with_subtune,
+			     strlen(path_container), path_container, NULL)) {
+		g_pattern_spec_free(path_with_subtune);
+		return path_container;
+	}
+
+	char *ptr = g_strrstr(path_container, "/" SUBTUNE_PREFIX);
+	if (ptr != NULL)
+		*ptr='\0';
+
+	g_pattern_spec_free(path_with_subtune);
+	return path_container;
+}
+
+/**
+ * returns tune number from file.nsf/tune_xxx.* style path or 0 if no subtune
+ * is appended.
+ */
+static int
+get_song_num(const char *path_fs)
+{
+	const char *subtune_suffix = uri_get_suffix(path_fs);
+	char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", subtune_suffix, NULL);
+	GPatternSpec *path_with_subtune = g_pattern_spec_new(pat);
+	g_free(pat);
+
+	if (g_pattern_match(path_with_subtune,
+			    strlen(path_fs), path_fs, NULL)) {
+		char *sub = g_strrstr(path_fs, "/" SUBTUNE_PREFIX);
+		g_pattern_spec_free(path_with_subtune);
+		if(!sub)
+			return 0;
+
+		sub += strlen("/" SUBTUNE_PREFIX);
+		int song_num = strtol(sub, NULL, 10);
+
+		return song_num - 1;
+	} else {
+		g_pattern_spec_free(path_with_subtune);
+		return 0;
+	}
+}
+
+static char *
+gme_container_scan(const char *path_fs, const unsigned int tnum)
+{
+	Music_Emu *emu;
+	const char* gme_err;
+	unsigned int num_songs;
+
+	gme_err = gme_open_file(path_fs, &emu, GME_SAMPLE_RATE);
+	if (gme_err != NULL) {
+		g_warning("%s", gme_err);
+		return NULL;
+	}
+
+	num_songs = gme_track_count(emu);
+	/* if it only contains a single tune, don't treat as container */
+	if (num_songs < 2)
+		return NULL;
+
+	const char *subtune_suffix = uri_get_suffix(path_fs);
+	if (tnum <= num_songs){
+		char *subtune = g_strdup_printf(
+			SUBTUNE_PREFIX "%03u.%s", tnum, subtune_suffix);
+		return subtune;
+	} else
+		return NULL;
+}
+
+static void
+gme_file_decode(struct decoder *decoder, const char *path_fs)
+{
+	float song_len;
+	Music_Emu *emu;
+	gme_info_t *ti;
+	struct audio_format audio_format;
+	enum decoder_command cmd;
+	short buf[GME_BUFFER_SAMPLES];
+	const char* gme_err;
+	char *path_container = get_container_name(path_fs);
+	int song_num = get_song_num(path_fs);
+
+	gme_err = gme_open_file(path_container, &emu, GME_SAMPLE_RATE);
+	g_free(path_container);
+	if (gme_err != NULL) {
+		g_warning("%s", gme_err);
+		return;
+	}
+
+	if((gme_err = gme_track_info(emu, &ti, song_num)) != NULL){
+		g_warning("%s", gme_err);
+		gme_delete(emu);
+		return;
+	}
+
+	if(ti->length > 0)
+		song_len = ti->length / 1000.0;
+	else song_len = -1;
+
+	/* initialize the MPD decoder */
+
+	GError *error = NULL;
+	if (!audio_format_init_checked(&audio_format, GME_SAMPLE_RATE,
+				       SAMPLE_FORMAT_S16, GME_CHANNELS,
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		gme_free_info(ti);
+		gme_delete(emu);
+		return;
+	}
+
+	decoder_initialized(decoder, &audio_format, true, song_len);
+
+	if((gme_err = gme_start_track(emu, song_num)) != NULL)
+		g_warning("%s", gme_err);
+
+	/* play */
+	do {
+		gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf);
+		if (gme_err != NULL) {
+			g_warning("%s", gme_err);
+			return;
+		}
+		cmd = decoder_data(decoder, NULL, buf, sizeof(buf), 0);
+
+		if(cmd == DECODE_COMMAND_SEEK) {
+			float where = decoder_seek_where(decoder);
+			if((gme_err = gme_seek(emu, (int)where*1000)) != NULL)
+				g_warning("%s", gme_err);
+			decoder_command_finished(decoder);
+		}
+
+		if(gme_track_ended(emu))
+			break;
+	} while(cmd != DECODE_COMMAND_STOP);
+
+	gme_free_info(ti);
+	gme_delete(emu);
+}
+
+static struct tag *
+gme_tag_dup(const char *path_fs)
+{
+	Music_Emu *emu;
+	gme_info_t *ti;
+	const char* gme_err;
+	char *path_container=get_container_name(path_fs);
+	int song_num;
+	song_num=get_song_num(path_fs);
+
+	gme_err = gme_open_file(path_container, &emu, GME_SAMPLE_RATE);
+	g_free(path_container);
+	if (gme_err != NULL) {
+		g_warning("%s", gme_err);
+		return NULL;
+	}
+	if((gme_err = gme_track_info(emu, &ti, song_num)) != NULL){
+		g_warning("%s", gme_err);
+		gme_delete(emu);
+		return NULL;
+	}
+
+	struct tag *tag = tag_new();
+	if(ti != NULL){
+		if(ti->length > 0)
+			tag->time = ti->length / 1000;
+		if(ti->song != NULL){
+			if(gme_track_count(emu) > 1){
+				/* start numbering subtunes from 1 */
+				char *tag_title=g_strdup_printf("%s (%d/%d)",
+					ti->song, song_num+1, gme_track_count(emu));
+				tag_add_item(tag, TAG_TITLE, tag_title);
+				g_free(tag_title);
+			}else
+				tag_add_item(tag, TAG_TITLE, ti->song);
+		}
+		if(ti->author != NULL)
+			tag_add_item(tag, TAG_ARTIST, ti->author);
+		if(ti->game != NULL)
+			tag_add_item(tag, TAG_ALBUM, ti->game);
+		if(ti->comment != NULL)
+			tag_add_item(tag, TAG_COMMENT, ti->comment);
+		if(ti->copyright != NULL)
+			tag_add_item(tag, TAG_DATE, ti->copyright);
+	}
+
+	gme_free_info(ti);
+	gme_delete(emu);
+	return tag;
+}
+
+static const char *const gme_suffixes[] = {
+	"ay", "gbs", "gym", "hes", "kss", "nsf",
+	"nsfe", "sap", "spc", "vgm", "vgz",
+	NULL
+};
+
+extern const struct decoder_plugin gme_decoder_plugin;
+const struct decoder_plugin gme_decoder_plugin = {
+	.name = "gme",
+	.file_decode = gme_file_decode,
+	.tag_dup = gme_tag_dup,
+	.suffixes = gme_suffixes,
+	.container_scan = gme_container_scan,
+};
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_decoder_plugin.c
index 9b3259485..a11d1b020 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,10 +17,12 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
-#include "../conf.h"
 #include "config.h"
+#include "decoder_api.h"
+#include "conf.h"
 #include "tag_id3.h"
+#include "tag_rva2.h"
+#include "audio_check.h"
 
 #include <assert.h>
 #include <unistd.h>
@@ -125,6 +127,7 @@ struct mp3_data {
 	unsigned int drop_end_frames;
 	unsigned int drop_start_samples;
 	unsigned int drop_end_samples;
+	bool found_replay_gain;
 	bool found_xing;
 	bool found_first_frame;
 	bool decoded_first_frame;
@@ -148,6 +151,7 @@ mp3_data_init(struct mp3_data *data, struct decoder *decoder,
 	data->drop_end_frames = 0;
 	data->drop_start_samples = 0;
 	data->drop_end_samples = 0;
+	data->found_replay_gain = false;
 	data->found_xing = false;
 	data->found_first_frame = false;
 	data->decoded_first_frame = false;
@@ -164,7 +168,7 @@ mp3_data_init(struct mp3_data *data, struct decoder *decoder,
 
 static bool mp3_seek(struct mp3_data *data, long offset)
 {
-	if (!input_stream_seek(data->input_stream, offset, SEEK_SET))
+	if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL))
 		return false;
 
 	mad_stream_buffer(&data->stream, data->input_buffer, 0);
@@ -208,105 +212,17 @@ mp3_fill_buffer(struct mp3_data *data)
 }
 
 #ifdef HAVE_ID3TAG
-/* Parse mp3 RVA2 frame. Shamelessly stolen from madplay. */
 static bool
-parse_rva2(struct id3_tag *tag, struct replay_gain_info *replay_gain_info)
-{
-	struct id3_frame const * frame;
-
-	id3_latin1_t const *id;
-	id3_byte_t const *data;
-	id3_length_t length;
-
-	enum {
-		CHANNEL_OTHER         = 0x00,
-		CHANNEL_MASTER_VOLUME = 0x01,
-		CHANNEL_FRONT_RIGHT   = 0x02,
-		CHANNEL_FRONT_LEFT    = 0x03,
-		CHANNEL_BACK_RIGHT    = 0x04,
-		CHANNEL_BACK_LEFT     = 0x05,
-		CHANNEL_FRONT_CENTRE  = 0x06,
-		CHANNEL_BACK_CENTRE   = 0x07,
-		CHANNEL_SUBWOOFER     = 0x08
-	};
-
-	/* relative volume adjustment information */
-
-	frame = id3_tag_findframe(tag, "RVA2", 0);
-	if (frame == NULL)
-		return false;
-
-	id   = id3_field_getlatin1(id3_frame_field(frame, 0));
-	data = id3_field_getbinarydata(id3_frame_field(frame, 1),
-					&length);
-
-	if (id == NULL || data == NULL)
-		return false;
-
-	/*
-	 * "The 'identification' string is used to identify the
-	 * situation and/or device where this adjustment should apply.
-	 * The following is then repeated for every channel
-	 *
-	 *   Type of channel         $xx
-	 *   Volume adjustment       $xx xx
-	 *   Bits representing peak  $xx
-	 *   Peak volume             $xx (xx ...)"
-	 */
-
-	while (length >= 4) {
-		unsigned int peak_bytes;
-
-		peak_bytes = (data[3] + 7) / 8;
-		if (4 + peak_bytes > length)
-			break;
-
-		if (data[0] == CHANNEL_MASTER_VOLUME) {
-			signed int voladj_fixed;
-			double voladj_float;
-
-			/*
-			 * "The volume adjustment is encoded as a fixed
-			 * point decibel value, 16 bit signed integer
-			 * representing (adjustment*512), giving +/- 64
-			 * dB with a precision of 0.001953125 dB."
-			 */
-
-			voladj_fixed  = (data[1] << 8) | (data[2] << 0);
-			voladj_fixed |= -(voladj_fixed & 0x8000);
-
-			voladj_float  = (double) voladj_fixed / 512;
-
-			replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = voladj_float;
-			replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = voladj_float;
-
-			g_debug("parseRVA2: Relative Volume "
-				"%+.1f dB adjustment (%s)\n",
-				voladj_float, id);
-
-			return true;
-		}
-
-		data   += 4 + peak_bytes;
-		length -= 4 + peak_bytes;
-	}
-
-	return false;
-}
-#endif
-
-#ifdef HAVE_ID3TAG
-static struct replay_gain_info *
-parse_id3_replay_gain_info(struct id3_tag *tag)
+parse_id3_replay_gain_info(struct replay_gain_info *replay_gain_info,
+			   struct id3_tag *tag)
 {
 	int i;
 	char *key;
 	char *value;
 	struct id3_frame *frame;
 	bool found = false;
-	struct replay_gain_info *replay_gain_info;
 
-	replay_gain_info = replay_gain_info_new();
+	replay_gain_info_init(replay_gain_info);
 
 	for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) {
 		if (frame->nfields < 3)
@@ -337,21 +253,55 @@ parse_id3_replay_gain_info(struct id3_tag *tag)
 		free(value);
 	}
 
-	if (!found) {
+	return found ||
 		/* fall back on RVA2 if no replaygain tags found */
-		found = parse_rva2(tag, replay_gain_info);
+		tag_rva2_parse(tag, replay_gain_info);
+}
+#endif
+
+#ifdef HAVE_ID3TAG
+static bool
+parse_id3_mixramp(char **mixramp_start, char **mixramp_end,
+		  struct id3_tag *tag)
+{
+	int i;
+	char *key;
+	char *value;
+	struct id3_frame *frame;
+	bool found = false;
+
+	*mixramp_start = NULL;
+	*mixramp_end = NULL;
+
+	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 (g_ascii_strcasecmp(key, "mixramp_start") == 0) {
+			*mixramp_start = g_strdup(value);
+			found = true;
+		} else if (g_ascii_strcasecmp(key, "mixramp_end") == 0) {
+			*mixramp_end = g_strdup(value);
+			found = true;
+		}
+
+		free(key);
+		free(value);
 	}
 
-	if (found)
-		return replay_gain_info;
-	replay_gain_info_free(replay_gain_info);
-	return NULL;
+	return found;
 }
 #endif
 
 static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
-			  struct tag **mpd_tag,
-			  struct replay_gain_info **replay_gain_info_r)
+			  struct tag **mpd_tag)
 {
 #ifdef HAVE_ID3TAG
 	struct id3_tag *id3_tag = NULL;
@@ -404,13 +354,20 @@ static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
 		}
 	}
 
-	if (replay_gain_info_r) {
-		struct replay_gain_info *tmp_rgi =
-			parse_id3_replay_gain_info(id3_tag);
-		if (tmp_rgi != NULL) {
-			if (*replay_gain_info_r)
-				replay_gain_info_free(*replay_gain_info_r);
-			*replay_gain_info_r = tmp_rgi;
+	if (data->decoder != NULL) {
+		struct replay_gain_info rgi;
+		char *mixramp_start;
+		char *mixramp_end;
+		float replay_gain_db = 0;
+
+		if (parse_id3_replay_gain_info(&rgi, id3_tag)) {
+			replay_gain_db = decoder_replay_gain(data->decoder, &rgi);
+			data->found_replay_gain = true;
+		}
+		if (parse_id3_mixramp(&mixramp_start, &mixramp_end, id3_tag)) {
+			g_debug("setting mixramp_tags");
+			decoder_mixramp(data->decoder, replay_gain_db,
+					mixramp_start, mixramp_end);
 		}
 	}
 
@@ -419,7 +376,6 @@ static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
 	g_free(allocated);
 #else /* !HAVE_ID3TAG */
 	(void)mpd_tag;
-	(void)replay_gain_info_r;
 
 	/* This code is enabled when libid3tag is disabled.  Instead
 	   of parsing the ID3 frame, it just skips it. */
@@ -467,8 +423,7 @@ id3_tag_query(const void *p0, size_t length)
 #endif /* !HAVE_ID3TAG */
 
 static enum mp3_action
-decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag,
-			 G_GNUC_UNUSED struct replay_gain_info **replay_gain_info_r)
+decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag)
 {
 	enum mad_layer layer;
 
@@ -490,7 +445,7 @@ decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag,
 			if (tagsize > 0) {
 				if (tag && !(*tag)) {
 					mp3_parse_id3(data, (size_t)tagsize,
-						      tag, replay_gain_info_r);
+						      tag);
 				} else {
 					mad_stream_skip(&(data->stream),
 							tagsize);
@@ -798,10 +753,10 @@ mp3_frame_duration(const struct mad_frame *frame)
 			       MAD_UNITS_MILLISECONDS) / 1000.0;
 }
 
-static off_t
+static goffset
 mp3_this_frame_offset(const struct mp3_data *data)
 {
-	off_t offset = data->input_stream->offset;
+	goffset offset = data->input_stream->offset;
 
 	if (data->stream.this_frame != NULL)
 		offset -= data->stream.bufend - data->stream.this_frame;
@@ -811,7 +766,7 @@ mp3_this_frame_offset(const struct mp3_data *data)
 	return offset;
 }
 
-static off_t
+static goffset
 mp3_rest_including_this_frame(const struct mp3_data *data)
 {
 	return data->input_stream->size - mp3_this_frame_offset(data);
@@ -823,7 +778,7 @@ mp3_rest_including_this_frame(const struct mp3_data *data)
 static void
 mp3_filesize_to_song_length(struct mp3_data *data)
 {
-	off_t rest = mp3_rest_including_this_frame(data);
+	goffset rest = mp3_rest_including_this_frame(data);
 
 	if (rest > 0) {
 		float frame_duration = mp3_frame_duration(&data->frame);
@@ -838,8 +793,7 @@ mp3_filesize_to_song_length(struct mp3_data *data)
 }
 
 static bool
-mp3_decode_first_frame(struct mp3_data *data, struct tag **tag,
-		       struct replay_gain_info **replay_gain_info_r)
+mp3_decode_first_frame(struct mp3_data *data, struct tag **tag)
 {
 	struct xing xing;
 	struct lame lame;
@@ -853,8 +807,7 @@ mp3_decode_first_frame(struct mp3_data *data, struct tag **tag,
 
 	while (true) {
 		do {
-			ret = decode_next_frame_header(data, tag,
-						       replay_gain_info_r);
+			ret = decode_next_frame_header(data, tag);
 		} while (ret == DECODE_CONT);
 		if (ret == DECODE_BREAK)
 			return false;
@@ -897,14 +850,17 @@ mp3_decode_first_frame(struct mp3_data *data, struct tag **tag,
 
 			/* Album gain isn't currently used.  See comment in
 			 * parse_lame() for details. -- jat */
-			if (replay_gain_info_r && !*replay_gain_info_r &&
+			if (data->decoder != NULL &&
+			    !data->found_replay_gain &&
 			    lame.track_gain) {
-				*replay_gain_info_r = replay_gain_info_new();
-				(*replay_gain_info_r)->tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain;
-				(*replay_gain_info_r)->tuples[REPLAY_GAIN_TRACK].peak = lame.peak;
+				struct replay_gain_info rgi;
+				replay_gain_info_init(&rgi);
+				rgi.tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain;
+				rgi.tuples[REPLAY_GAIN_TRACK].peak = lame.peak;
+				decoder_replay_gain(data->decoder, &rgi);
 			}
 		}
-	} 
+	}
 
 	if (!data->max_frames)
 		return false;
@@ -932,33 +888,29 @@ static void mp3_data_finish(struct mp3_data *data)
 }
 
 /* this is primarily used for getting total time for tags */
-static int mp3_total_file_time(const char *file)
+static int
+mad_decoder_total_file_time(struct input_stream *is)
 {
-	struct input_stream input_stream;
 	struct mp3_data data;
 	int ret;
 
-	if (!input_stream_open(&input_stream, file))
-		return -1;
-	mp3_data_init(&data, NULL, &input_stream);
-	if (!mp3_decode_first_frame(&data, NULL, NULL))
+	mp3_data_init(&data, NULL, is);
+	if (!mp3_decode_first_frame(&data, NULL))
 		ret = -1;
 	else
 		ret = data.total_time + 0.5;
 	mp3_data_finish(&data);
-	input_stream_close(&input_stream);
 
 	return ret;
 }
 
 static bool
 mp3_open(struct input_stream *is, struct mp3_data *data,
-	 struct decoder *decoder, struct tag **tag,
-	 struct replay_gain_info **replay_gain_info_r)
+	 struct decoder *decoder, struct tag **tag)
 {
 	mp3_data_init(data, decoder, is);
 	*tag = NULL;
-	if (!mp3_decode_first_frame(data, tag, replay_gain_info_r)) {
+	if (!mp3_decode_first_frame(data, tag)) {
 		mp3_data_finish(data);
 		if (tag && *tag)
 			tag_free(*tag);
@@ -1017,8 +969,7 @@ mp3_update_timer_next_frame(struct mp3_data *data)
  * Sends the synthesized current frame via decoder_data().
  */
 static enum decoder_command
-mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length,
-	     struct replay_gain_info *replay_gain_info)
+mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length)
 {
 	unsigned max_samples;
 
@@ -1043,9 +994,7 @@ mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length,
 		cmd = decoder_data(data->decoder, data->input_stream,
 				   data->output_buffer,
 				   sizeof(data->output_buffer[0]) * num_samples,
-				   data->elapsed_time,
-				   data->bit_rate / 1000,
-				   replay_gain_info);
+				   data->bit_rate / 1000);
 		if (cmd != DECODE_COMMAND_NONE)
 			return cmd;
 	}
@@ -1057,8 +1006,7 @@ mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length,
  * Synthesize the current frame and send it via decoder_data().
  */
 static enum decoder_command
-mp3_synth_and_send(struct mp3_data *data,
-		   struct replay_gain_info *replay_gain_info)
+mp3_synth_and_send(struct mp3_data *data)
 {
 	unsigned i, pcm_length;
 	enum decoder_command cmd;
@@ -1099,7 +1047,7 @@ mp3_synth_and_send(struct mp3_data *data,
 			pcm_length -= data->drop_end_samples;
 	}
 
-	cmd = mp3_send_pcm(data, i, pcm_length, replay_gain_info);
+	cmd = mp3_send_pcm(data, i, pcm_length);
 	if (cmd != DECODE_COMMAND_NONE)
 		return cmd;
 
@@ -1113,7 +1061,7 @@ mp3_synth_and_send(struct mp3_data *data,
 }
 
 static bool
-mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
+mp3_read(struct mp3_data *data)
 {
 	struct decoder *decoder = data->decoder;
 	enum mp3_action ret;
@@ -1130,9 +1078,7 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
 			data->mute_frame = MUTEFRAME_NONE;
 		break;
 	case MUTEFRAME_NONE:
-		cmd = mp3_synth_and_send(data,
-					 replay_gain_info_r != NULL
-					 ? *replay_gain_info_r : NULL);
+		cmd = mp3_synth_and_send(data);
 		if (cmd == DECODE_COMMAND_SEEK) {
 			unsigned long j;
 
@@ -1161,8 +1107,7 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
 		do {
 			struct tag *tag = NULL;
 
-			ret = decode_next_frame_header(data, &tag,
-						       replay_gain_info_r);
+			ret = decode_next_frame_header(data, &tag);
 
 			if (tag != NULL) {
 				decoder_tag(decoder, data->input_stream, tag);
@@ -1189,29 +1134,34 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
 	return ret != DECODE_BREAK;
 }
 
-static void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
-{
-	af->bits = 24;
-	af->sample_rate = (data->frame).header.samplerate;
-	af->channels = MAD_NCHANNELS(&(data->frame).header);
-}
-
 static void
 mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 {
 	struct mp3_data data;
+	GError *error = NULL;
 	struct tag *tag = NULL;
-	struct replay_gain_info *replay_gain_info = NULL;
 	struct audio_format audio_format;
 
-	if (!mp3_open(input_stream, &data, decoder, &tag, &replay_gain_info)) {
+	if (!mp3_open(input_stream, &data, decoder, &tag)) {
 		if (decoder_get_command(decoder) == DECODE_COMMAND_NONE)
 			g_warning
 			    ("Input does not appear to be a mp3 bit stream.\n");
 		return;
 	}
 
-	mp3_audio_format(&data, &audio_format);
+	if (!audio_format_init_checked(&audio_format,
+				       data.frame.header.samplerate,
+				       SAMPLE_FORMAT_S24_P32,
+				       MAD_NCHANNELS(&data.frame.header),
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+
+		if (tag != NULL)
+			tag_free(tag);
+		mp3_data_finish(&data);
+		return;
+	}
 
 	decoder_initialized(decoder, &audio_format,
 			    data.input_stream->seekable, data.total_time);
@@ -1221,24 +1171,20 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 		tag_free(tag);
 	}
 
-	while (mp3_read(&data, &replay_gain_info)) ;
-
-	if (replay_gain_info)
-		replay_gain_info_free(replay_gain_info);
+	while (mp3_read(&data)) ;
 
 	mp3_data_finish(&data);
 }
 
-static struct tag *mp3_tag_dup(const char *file)
+static struct tag *
+mad_decoder_stream_tag(struct input_stream *is)
 {
 	struct tag *tag;
 	int total_time;
 
-	total_time = mp3_total_file_time(file);
-	if (total_time < 0) {
-		g_debug("Failed to get total song time from: %s", file);
+	total_time = mad_decoder_total_file_time(is);
+	if (total_time < 0)
 		return NULL;
-	}
 
 	tag = tag_new();
 	tag->time = total_time;
@@ -1252,7 +1198,7 @@ const struct decoder_plugin mad_decoder_plugin = {
 	.name = "mad",
 	.init = mp3_plugin_init,
 	.stream_decode = mp3_decode,
-	.tag_dup = mp3_tag_dup,
+	.stream_tag = mad_decoder_stream_tag,
 	.suffixes = mp3_suffixes,
 	.mime_types = mp3_mime_types
 };
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_decoder_plugin.c
index f60dcbc61..91478e86f 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,10 +17,13 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
+#include "config.h"
+#include "decoder_api.h"
+#include "mpd_error.h"
 
 #include <glib.h>
 #include <mikmod.h>
+#include <assert.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "mikmod"
@@ -29,30 +32,34 @@
 
 #define MIKMOD_FRAME_SIZE	4096
 
-static BOOL mod_mpd_Init(void)
+static BOOL
+mikmod_mpd_init(void)
 {
 	return VC_Init();
 }
 
-static void mod_mpd_Exit(void)
+static void
+mikmod_mpd_exit(void)
 {
 	VC_Exit();
 }
 
-static void mod_mpd_Update(void)
+static void
+mikmod_mpd_update(void)
 {
 }
 
-static BOOL mod_mpd_IsThere(void)
+static BOOL
+mikmod_mpd_is_present(void)
 {
-	return 1;
+	return true;
 }
 
-static char drv_name[] = "MPD";
-static char drv_version[] = "MPD Output Driver v0.1";
+static char drv_name[] = PACKAGE_NAME;
+static char drv_version[] = VERSION;
 
 #if (LIBMIKMOD_VERSION > 0x030106)
-static char drv_alias[] = "mpd";
+static char drv_alias[] = PACKAGE;
 #endif
 
 static MDRIVER drv_mpd = {
@@ -68,18 +75,18 @@ static MDRIVER drv_mpd = {
 #endif
 	NULL,  /* CommandLine */
 #endif
-	mod_mpd_IsThere,
+	mikmod_mpd_is_present,
 	VC_SampleLoad,
 	VC_SampleUnload,
 	VC_SampleSpace,
 	VC_SampleLength,
-	mod_mpd_Init,
-	mod_mpd_Exit,
+	mikmod_mpd_init,
+	mikmod_mpd_exit,
 	NULL,
 	VC_SetNumVoices,
 	VC_PlayStart,
 	VC_PlayStop,
-	mod_mpd_Update,
+	mikmod_mpd_update,
 	NULL,
 	VC_VoiceSetVolume,
 	VC_VoiceGetVolume,
@@ -94,11 +101,19 @@ static MDRIVER drv_mpd = {
 	VC_VoiceRealVolume
 };
 
+static unsigned mikmod_sample_rate;
+
 static bool
-mod_initMikMod(G_GNUC_UNUSED const struct config_param *param)
+mikmod_decoder_init(const struct config_param *param)
 {
 	static char params[] = "";
 
+	mikmod_sample_rate = config_get_block_unsigned(param, "sample_rate",
+						       44100);
+	if (!audio_valid_sample_rate(mikmod_sample_rate))
+		MPD_ERROR("Invalid sample rate in line %d: %u",
+			  param->line, mikmod_sample_rate);
+
 	md_device = 0;
 	md_reverb = 0;
 
@@ -106,7 +121,7 @@ mod_initMikMod(G_GNUC_UNUSED const struct config_param *param)
 	MikMod_RegisterAllLoaders();
 
 	md_pansep = 64;
-	md_mixfreq = 44100;
+	md_mixfreq = mikmod_sample_rate;
 	md_mode = (DMODE_SOFT_MUSIC | DMODE_INTERP | DMODE_STEREO |
 		   DMODE_16BITS);
 
@@ -119,117 +134,80 @@ mod_initMikMod(G_GNUC_UNUSED const struct config_param *param)
 	return true;
 }
 
-static void mod_finishMikMod(void)
+static void
+mikmod_decoder_finish(void)
 {
 	MikMod_Exit();
 }
 
-typedef struct _mod_Data {
-	MODULE *moduleHandle;
-	SBYTE audio_buffer[MIKMOD_FRAME_SIZE];
-} mod_Data;
-
-static mod_Data *mod_open(const char *path)
-{
-	char *path2;
-	MODULE *moduleHandle;
-	mod_Data *data;
-
-	path2 = g_strdup(path);
-	moduleHandle = Player_Load(path2, 128, 0);
-	g_free(path2);
-
-	if (moduleHandle == NULL)
-		return NULL;
-
-	/* Prevent module from looping forever */
-	moduleHandle->loop = 0;
-
-	data = g_new(mod_Data, 1);
-	data->moduleHandle = moduleHandle;
-
-	Player_Start(data->moduleHandle);
-
-	return data;
-}
-
-static void mod_close(mod_Data * data)
-{
-	Player_Stop();
-	Player_Free(data->moduleHandle);
-	g_free(data);
-}
-
 static void
-mod_decode(struct decoder *decoder, const char *path)
+mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs)
 {
-	mod_Data *data;
+	char *path2;
+	MODULE *handle;
 	struct audio_format audio_format;
-	float total_time = 0.0;
 	int ret;
-	float secPerByte;
+	SBYTE buffer[MIKMOD_FRAME_SIZE];
 	enum decoder_command cmd = DECODE_COMMAND_NONE;
 
-	if (!(data = mod_open(path))) {
-		g_warning("failed to open mod: %s\n", path);
+	path2 = g_strdup(path_fs);
+	handle = Player_Load(path2, 128, 0);
+	g_free(path2);
+
+	if (handle == NULL) {
+		g_warning("failed to open mod: %s", path_fs);
 		return;
 	}
 
-	audio_format.bits = 16;
-	audio_format.sample_rate = 44100;
-	audio_format.channels = 2;
+	/* Prevent module from looping forever */
+	handle->loop = 0;
 
-	secPerByte =
-	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
-		   (float)audio_format.sample_rate);
+	audio_format_init(&audio_format, mikmod_sample_rate, SAMPLE_FORMAT_S16, 2);
+	assert(audio_format_valid(&audio_format));
 
 	decoder_initialized(decoder, &audio_format, false, 0);
 
+	Player_Start(handle);
 	while (cmd == DECODE_COMMAND_NONE && Player_Active()) {
-		ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE);
-		total_time += ret * secPerByte;
-		cmd = decoder_data(decoder, NULL,
-				   data->audio_buffer, ret,
-				   total_time, 0, NULL);
+		ret = VC_WriteBytes(buffer, sizeof(buffer));
+		cmd = decoder_data(decoder, NULL, buffer, ret, 0);
 	}
 
-	mod_close(data);
+	Player_Stop();
+	Player_Free(handle);
 }
 
-static struct tag *modTagDup(const char *file)
+static struct tag *
+mikmod_decoder_tag_dup(const char *path_fs)
 {
-	char *path2;
-	struct tag *ret = NULL;
-	MODULE *moduleHandle;
-	char *title;
+	char *path2 = g_strdup(path_fs);
+	MODULE *handle = Player_Load(path2, 128, 0);
 
-	path2 = g_strdup(file);
-	moduleHandle = Player_Load(path2, 128, 0);
-	g_free(path2);
-
-	if (moduleHandle == NULL) {
-		g_debug("Failed to open file: %s", file);
+	if (handle == NULL) {
+		g_free(path2);
+		g_debug("Failed to open file: %s", path_fs);
 		return NULL;
 
 	}
-	Player_Free(moduleHandle);
 
-	ret = tag_new();
+	Player_Free(handle);
 
-	ret->time = 0;
+	struct tag *tag = tag_new();
 
-	path2 = g_strdup(file);
-	title = Player_LoadTitle(path2);
+	tag->time = 0;
+
+	char *title = Player_LoadTitle(path2);
 	g_free(path2);
-	if (title) {
-		tag_add_item(ret, TAG_ITEM_TITLE, title);
+
+	if (title != NULL) {
+		tag_add_item(tag, TAG_TITLE, title);
 		free(title);
 	}
 
-	return ret;
+	return tag;
 }
 
-static const char *const modSuffixes[] = {
+static const char *const mikmod_decoder_suffixes[] = {
 	"amf",
 	"dsm",
 	"far",
@@ -250,9 +228,9 @@ static const char *const modSuffixes[] = {
 
 const struct decoder_plugin mikmod_decoder_plugin = {
 	.name = "mikmod",
-	.init = mod_initMikMod,
-	.finish = mod_finishMikMod,
-	.file_decode = mod_decode,
-	.tag_dup = modTagDup,
-	.suffixes = modSuffixes,
+	.init = mikmod_decoder_init,
+	.finish = mikmod_decoder_finish,
+	.file_decode = mikmod_decoder_file_decode,
+	.tag_dup = mikmod_decoder_tag_dup,
+	.suffixes = mikmod_decoder_suffixes,
 };
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_decoder_plugin.c
index f636f2fa6..037c2fd74 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,10 +17,12 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
+#include "config.h"
+#include "decoder_api.h"
 
 #include <glib.h>
 #include <modplug.h>
+#include <assert.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "modplug"
@@ -92,10 +94,8 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 	ModPlug_Settings settings;
 	GByteArray *bdatas;
 	struct audio_format audio_format;
-	float total_time = 0.0;
-	int ret, current;
+	int ret;
 	char audio_buffer[MODPLUG_FRAME_SIZE];
-	float sec_perbyte;
 	enum decoder_command cmd = DECODE_COMMAND_NONE;
 
 	bdatas = mod_loadfile(decoder, is);
@@ -121,37 +121,26 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	audio_format.bits = 16;
-	audio_format.sample_rate = 44100;
-	audio_format.channels = 2;
-
-	sec_perbyte =
-	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
-		   (float)audio_format.sample_rate);
-
-	total_time = ModPlug_GetLength(f) / 1000;
+	audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
+	assert(audio_format_valid(&audio_format));
 
 	decoder_initialized(decoder, &audio_format,
-			    is->seekable, total_time);
-
-	total_time = 0;
+			    is->seekable, ModPlug_GetLength(f) / 1000.0);
 
 	do {
 		ret = ModPlug_Read(f, audio_buffer, MODPLUG_FRAME_SIZE);
-
-		if (ret == 0) {
+		if (ret <= 0)
 			break;
-		}
 
-		total_time += ret * sec_perbyte;
 		cmd = decoder_data(decoder, NULL,
 				   audio_buffer, ret,
-				   total_time, 0, NULL);
+				   0);
 
 		if (cmd == DECODE_COMMAND_SEEK) {
-			total_time = decoder_seek_where(decoder);
-			current = total_time * 1000;
-			ModPlug_Seek(f, current);
+			float where = decoder_seek_where(decoder);
+
+			ModPlug_Seek(f, (int)(where * 1000.0));
+
 			decoder_command_finished(decoder);
 		}
 
@@ -160,43 +149,33 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 	ModPlug_Unload(f);
 }
 
-static struct tag *mod_tagdup(const char *file)
+static struct tag *
+modplug_stream_tag(struct input_stream *is)
 {
 	ModPlugFile *f;
 	struct tag *ret = NULL;
 	GByteArray *bdatas;
 	char *title;
-	struct input_stream is;
 
-	if (!input_stream_open(&is, file)) {
-		g_warning("cant open file %s\n", file);
+	bdatas = mod_loadfile(NULL, is);
+	if (!bdatas)
 		return NULL;
-	}
-
-	bdatas = mod_loadfile(NULL, &is);
-	if (!bdatas) {
-		g_warning("cant load file %s\n", file);
-		return NULL;
-	}
 
 	f = ModPlug_Load(bdatas->data, bdatas->len);
 	g_byte_array_free(bdatas, TRUE);
-	if (!f) {
-		g_warning("could not decode file %s\n", file);
+	if (f == NULL)
 		return NULL;
-        }
+
 	ret = tag_new();
-	ret->time = 0;
+	ret->time = ModPlug_GetLength(f) / 1000;
 
 	title = g_strdup(ModPlug_GetName(f));
 	if (title)
-		tag_add_item(ret, TAG_ITEM_TITLE, title);
+		tag_add_item(ret, TAG_TITLE, title);
 	g_free(title);
 
 	ModPlug_Unload(f);
 
-	input_stream_close(&is);
-
 	return ret;
 }
 
@@ -210,6 +189,6 @@ static const char *const mod_suffixes[] = {
 const struct decoder_plugin modplug_decoder_plugin = {
 	.name = "modplug",
 	.stream_decode = mod_decode,
-	.tag_dup = mod_tagdup,
+	.stream_tag = modplug_stream_tag,
 	.suffixes = mod_suffixes,
 };
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_decoder_plugin.c
index 4d4d47c6c..861b08194 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,9 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
 #include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
 #include "tag_table.h"
 
 #include <glib.h>
@@ -35,7 +36,9 @@
 
 /* all code here is either based on or copied from FAAD2's frontend code */
 
-struct mp4_context {
+struct mp4ff_input_stream {
+	mp4ff_callback_t callback;
+
 	struct decoder *decoder;
 	struct input_stream *input_stream;
 };
@@ -89,20 +92,38 @@ mp4_get_aac_track(mp4ff_t * infile, faacDecHandle decoder,
 static uint32_t
 mp4_read(void *user_data, void *buffer, uint32_t length)
 {
-	struct mp4_context *ctx = user_data;
+	struct mp4ff_input_stream *mis = user_data;
 
-	return decoder_read(ctx->decoder, ctx->input_stream, buffer, length);
+	return decoder_read(mis->decoder, mis->input_stream, buffer, length);
 }
 
 static uint32_t
 mp4_seek(void *user_data, uint64_t position)
 {
-	struct mp4_context *ctx = user_data;
+	struct mp4ff_input_stream *mis = user_data;
 
-	return input_stream_seek(ctx->input_stream, position, SEEK_SET)
+	return input_stream_seek(mis->input_stream, position, SEEK_SET, NULL)
 		? 0 : -1;
 }
 
+static const mp4ff_callback_t mpd_mp4ff_callback = {
+	.read = mp4_read,
+	.seek = mp4_seek,
+};
+
+static mp4ff_t *
+mp4ff_input_stream_open(struct mp4ff_input_stream *mis,
+			struct decoder *decoder,
+			struct input_stream *input_stream)
+{
+	mis->callback = mpd_mp4ff_callback;
+	mis->callback.user_data = mis;
+	mis->decoder = decoder;
+	mis->input_stream = input_stream;
+
+	return mp4ff_open_read(&mis->callback);
+}
+
 static faacDecHandle
 mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 {
@@ -111,6 +132,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 	int track;
 	uint32_t sample_rate;
 	unsigned char channels;
+	GError *error = NULL;
 
 	decoder = faacDecOpen();
 
@@ -131,37 +153,24 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 		return NULL;
 	}
 
-	*track_r = track;
-	*audio_format = (struct audio_format){
-		.bits = 16,
-		.channels = channels,
-		.sample_rate = sample_rate,
-	};
-
-	if (!audio_format_valid(audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format->sample_rate,
-			  audio_format->bits,
-			  audio_format->channels);
+	if (!audio_format_init_checked(audio_format, sample_rate,
+				       SAMPLE_FORMAT_S16, channels,
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		faacDecClose(decoder);
 		return NULL;
 	}
 
+	*track_r = track;
+
 	return decoder;
 }
 
 static void
 mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
 {
-	struct mp4_context ctx = {
-		.decoder = mpd_decoder,
-		.input_stream = input_stream,
-	};
-	mp4ff_callback_t callback = {
-		.read = mp4_read,
-		.seek = mp4_seek,
-		.user_data = &ctx,
-	};
+	struct mp4ff_input_stream mis;
 	mp4ff_t *mp4fh;
 	int32_t track;
 	float file_time, total_time;
@@ -187,7 +196,7 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
 	double seek_where = 0;
 	enum decoder_command cmd = DECODE_COMMAND_NONE;
 
-	mp4fh = mp4ff_open_read(&callback);
+	mp4fh = mp4ff_input_stream_open(&mis, mpd_decoder, input_stream);
 	if (!mp4fh) {
 		g_warning("Input does not appear to be a mp4 stream.\n");
 		return;
@@ -266,7 +275,7 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
 			dur -= offset;
 		file_time += ((float)dur) / scale;
 
-		if (seeking && file_time > seek_where)
+		if (seeking && file_time >= seek_where)
 			seek_position_found = true;
 
 		if (seeking && seek_position_found) {
@@ -332,7 +341,7 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
 
 		cmd = decoder_data(mpd_decoder, input_stream,
 				   sample_buffer, sample_buffer_length,
-				   file_time, bit_rate, NULL);
+				   bit_rate);
 	}
 
 	g_free(seek_table);
@@ -341,9 +350,9 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
 }
 
 static const char *const mp4ff_tag_names[TAG_NUM_OF_ITEM_TYPES] = {
-	[TAG_ITEM_ALBUM_ARTIST] = "album artist",
-	[TAG_ITEM_COMPOSER] = "writer",
-	[TAG_ITEM_PERFORMER] = "band",
+	[TAG_ALBUM_ARTIST] = "album artist",
+	[TAG_COMPOSER] = "writer",
+	[TAG_PERFORMER] = "band",
 };
 
 static enum tag_type
@@ -353,57 +362,41 @@ mp4ff_tag_name_parse(const char *name)
 	if (type == TAG_NUM_OF_ITEM_TYPES)
 		type = tag_name_parse_i(name);
 
+	if (g_ascii_strcasecmp(name, "albumartist") == 0 ||
+	    g_ascii_strcasecmp(name, "album_artist") == 0)
+		return TAG_ALBUM_ARTIST;
+
 	return type;
 }
 
 static struct tag *
-mp4_tag_dup(const char *file)
+mp4_stream_tag(struct input_stream *is)
 {
-	struct tag *ret = NULL;
-	struct input_stream input_stream;
-	struct mp4_context ctx = {
-		.decoder = NULL,
-		.input_stream = &input_stream,
-	};
-	mp4ff_callback_t callback = {
-		.read = mp4_read,
-		.seek = mp4_seek,
-		.user_data = &ctx,
-	};
-	mp4ff_t *mp4fh;
+	struct mp4ff_input_stream mis;
 	int32_t track;
 	int32_t file_time;
 	int32_t scale;
 	int i;
 
-	if (!input_stream_open(&input_stream, file)) {
-		g_warning("Failed to open file: %s", file);
+	mp4ff_t *mp4fh = mp4ff_input_stream_open(&mis, NULL, is);
+	if (mp4fh == NULL)
 		return NULL;
-	}
-
-	mp4fh = mp4ff_open_read(&callback);
-	if (!mp4fh) {
-		input_stream_close(&input_stream);
-		return NULL;
-	}
 
 	track = mp4_get_aac_track(mp4fh, NULL, NULL, NULL);
 	if (track < 0) {
 		mp4ff_close(mp4fh);
-		input_stream_close(&input_stream);
 		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);
-		input_stream_close(&input_stream);
-		tag_free(ret);
 		return NULL;
 	}
-	ret->time = ((float)file_time) / scale + 0.5;
+
+	struct tag *tag = tag_new();
+	tag->time = ((float)file_time) / scale + 0.5;
 
 	for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
 		char *item;
@@ -413,16 +406,15 @@ mp4_tag_dup(const char *file)
 
 		enum tag_type type = mp4ff_tag_name_parse(item);
 		if (type != TAG_NUM_OF_ITEM_TYPES)
-			tag_add_item(ret, type, value);
+			tag_add_item(tag, type, value);
 
 		free(item);
 		free(value);
 	}
 
 	mp4ff_close(mp4fh);
-	input_stream_close(&input_stream);
 
-	return ret;
+	return tag;
 }
 
 static const char *const mp4_suffixes[] = {
@@ -435,9 +427,9 @@ static const char *const mp4_suffixes[] = {
 static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL };
 
 const struct decoder_plugin mp4ff_decoder_plugin = {
-	.name = "mp4",
+	.name = "mp4ff",
 	.stream_decode = mp4_decode,
-	.tag_dup = mp4_tag_dup,
+	.stream_tag = mp4_stream_tag,
 	.suffixes = mp4_suffixes,
 	.mime_types = mp4_mime_types,
 };
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_decoder_plugin.c
index 72a516f22..4df8dd218 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,9 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
 #include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
 
 #ifdef MPC_IS_OLD_API
 #include <mpcdec/mpcdec.h>
@@ -28,6 +29,7 @@
 #endif
 
 #include <glib.h>
+#include <assert.h>
 #include <unistd.h>
 
 #undef G_LOG_DOMAIN
@@ -59,7 +61,7 @@ mpc_seek_cb(cb_first_arg, mpc_int32_t offset)
 {
 	struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data;
 
-	return input_stream_seek(data->is, offset, SEEK_SET);
+	return input_stream_seek(data->is, offset, SEEK_SET, NULL);
 }
 
 static mpc_int32_t
@@ -141,6 +143,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 #endif
 	mpc_reader reader;
 	mpc_streaminfo info;
+	GError *error = NULL;
 	struct audio_format audio_format;
 
 	struct mpc_decoder_data data;
@@ -150,11 +153,8 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_uint32_t ret;
 	int32_t chunk[G_N_ELEMENTS(sample_buffer)];
 	long bit_rate = 0;
-	unsigned long sample_pos = 0;
 	mpc_uint32_t vbr_update_acc;
 	mpc_uint32_t vbr_update_bits;
-	float total_time;
-	struct replay_gain_info *replay_gain_info = NULL;
 	enum decoder_command cmd = DECODE_COMMAND_NONE;
 
 	data.is = is;
@@ -194,50 +194,47 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_demux_get_info(demux, &info);
 #endif
 
-	audio_format.bits = 24;
-	audio_format.channels = info.channels;
-	audio_format.sample_rate = info.sample_freq;
-
-	if (!audio_format_valid(&audio_format)) {
+	if (!audio_format_init_checked(&audio_format, info.sample_freq,
+				       SAMPLE_FORMAT_S24_P32,
+				       info.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 #ifndef MPC_IS_OLD_API
 		mpc_demux_exit(demux);
 #endif
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate,
-			  audio_format.bits,
-			  audio_format.channels);
 		return;
 	}
 
-	replay_gain_info = replay_gain_info_new();
+	struct replay_gain_info replay_gain_info;
+	replay_gain_info_init(&replay_gain_info);
 #ifdef MPC_IS_OLD_API
-	replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = info.gain_album * 0.01;
-	replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = info.peak_album / 32767.0;
-	replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = info.gain_title * 0.01;
-	replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = info.peak_title / 32767.0;
+	replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = info.gain_album * 0.01;
+	replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = info.peak_album / 32767.0;
+	replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = info.gain_title * 0.01;
+	replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = info.peak_title / 32767.0;
 #else
-	replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = MPC_OLD_GAIN_REF  - (info.gain_album  / 256.);
-	replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = pow(10, info.peak_album / 256. / 20) / 32767;
-	replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = MPC_OLD_GAIN_REF  - (info.gain_title  / 256.);
-	replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = pow(10, info.peak_title / 256. / 20) / 32767;
+	replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = MPC_OLD_GAIN_REF  - (info.gain_album  / 256.);
+	replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = pow(10, info.peak_album / 256. / 20) / 32767;
+	replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = MPC_OLD_GAIN_REF  - (info.gain_title  / 256.);
+	replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = pow(10, info.peak_title / 256. / 20) / 32767;
 #endif
 
+	decoder_replay_gain(mpd_decoder, &replay_gain_info);
+
 	decoder_initialized(mpd_decoder, &audio_format,
 			    is->seekable,
 			    mpc_streaminfo_get_length(&info));
 
 	do {
 		if (cmd == DECODE_COMMAND_SEEK) {
-			bool success;
-
-			sample_pos = decoder_seek_where(mpd_decoder) *
+			mpc_int64_t where = decoder_seek_where(mpd_decoder) *
 				audio_format.sample_rate;
+			bool success;
 
 #ifdef MPC_IS_OLD_API
-			success = mpc_decoder_seek_sample(&decoder,
-							  sample_pos);
+			success = mpc_decoder_seek_sample(&decoder, where);
 #else
-			success = mpc_demux_seek_sample(demux, sample_pos)
+			success = mpc_demux_seek_sample(demux, where)
 				== MPC_STATUS_OK;
 #endif
 			if (success)
@@ -268,33 +265,26 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 		ret = frame.samples;
 #endif
 
-		sample_pos += ret;
-
 		ret *= info.channels;
 
 		mpc_to_mpd_buffer(chunk, sample_buffer, ret);
 
-		total_time = ((float)sample_pos) / audio_format.sample_rate;
 		bit_rate = vbr_update_bits * audio_format.sample_rate
 			/ 1152 / 1000;
 
 		cmd = decoder_data(mpd_decoder, is,
 				   chunk, ret * sizeof(chunk[0]),
-				   total_time,
-				   bit_rate, replay_gain_info);
+				   bit_rate);
 	} while (cmd != DECODE_COMMAND_STOP);
 
-	replay_gain_info_free(replay_gain_info);
-
 #ifndef MPC_IS_OLD_API
 	mpc_demux_exit(demux);
 #endif
 }
 
 static float
-mpcdec_get_file_duration(const char *file)
+mpcdec_get_file_duration(struct input_stream *is)
 {
-	struct input_stream is;
 	float total_time = -1;
 
 	mpc_reader reader;
@@ -304,10 +294,7 @@ mpcdec_get_file_duration(const char *file)
 	mpc_streaminfo info;
 	struct mpc_decoder_data data;
 
-	if (!input_stream_open(&is, file))
-		return -1;
-
-	data.is = &is;
+	data.is = is;
 	data.decoder = NULL;
 
 	reader.read = mpc_read_cb;
@@ -320,16 +307,12 @@ mpcdec_get_file_duration(const char *file)
 #ifdef MPC_IS_OLD_API
 	mpc_streaminfo_init(&info);
 
-	if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) {
-		input_stream_close(&is);
+	if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK)
 		return -1;
-	}
 #else
 	demux = mpc_demux_init(&reader);
-	if (demux == NULL) {
-		input_stream_close(&is);
+	if (demux == NULL)
 		return -1;
-	}
 
 	mpc_demux_get_info(demux, &info);
 	mpc_demux_exit(demux);
@@ -337,21 +320,17 @@ mpcdec_get_file_duration(const char *file)
 
 	total_time = mpc_streaminfo_get_length(&info);
 
-	input_stream_close(&is);
-
 	return total_time;
 }
 
 static struct tag *
-mpcdec_tag_dup(const char *file)
+mpcdec_stream_tag(struct input_stream *is)
 {
-	float total_time = mpcdec_get_file_duration(file);
+	float total_time = mpcdec_get_file_duration(is);
 	struct tag *tag;
 
-	if (total_time < 0) {
-		g_debug("Failed to get duration of file: %s", file);
+	if (total_time < 0)
 		return NULL;
-	}
 
 	tag = tag_new();
 	tag->time = total_time;
@@ -363,6 +342,6 @@ static const char *const mpcdec_suffixes[] = { "mpc", NULL };
 const struct decoder_plugin mpcdec_decoder_plugin = {
 	.name = "mpcdec",
 	.stream_decode = mpcdec_decode,
-	.tag_dup = mpcdec_tag_dup,
+	.stream_tag = mpcdec_stream_tag,
 	.suffixes = mpcdec_suffixes,
 };
diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c
new file mode 100644
index 000000000..7b48ebfaf
--- /dev/null
+++ b/src/decoder/mpg123_decoder_plugin.c
@@ -0,0 +1,209 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h" /* must be first for large file support */
+#include "decoder_api.h"
+#include "audio_check.h"
+
+#include <glib.h>
+
+#include <mpg123.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "mpg123"
+
+static bool
+mpd_mpg123_init(G_GNUC_UNUSED const struct config_param *param)
+{
+	mpg123_init();
+
+	return true;
+}
+
+static void
+mpd_mpg123_finish(void)
+{
+	mpg123_exit();
+}
+
+/**
+ * Opens a file with an existing #mpg123_handle.
+ *
+ * @param handle a handle which was created before; on error, this
+ * function will not free it
+ * @param audio_format this parameter is filled after successful
+ * return
+ * @return true on success
+ */
+static bool
+mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
+		struct audio_format *audio_format)
+{
+	GError *gerror = NULL;
+	char *path_dup;
+	int error;
+	int channels, encoding;
+	long rate;
+
+	/* mpg123_open() wants a writable string :-( */
+	path_dup = g_strdup(path_fs);
+
+	error = mpg123_open(handle, path_dup);
+	g_free(path_dup);
+	if (error != MPG123_OK) {
+		g_warning("libmpg123 failed to open %s: %s",
+			  path_fs, mpg123_plain_strerror(error));
+		return false;
+	}
+
+	/* obtain the audio format */
+
+	error = mpg123_getformat(handle, &rate, &channels, &encoding);
+	if (error != MPG123_OK) {
+		g_warning("mpg123_getformat() failed: %s",
+			  mpg123_plain_strerror(error));
+		return false;
+	}
+
+	if (encoding != MPG123_ENC_SIGNED_16) {
+		/* other formats not yet implemented */
+		g_warning("expected MPG123_ENC_SIGNED_16, got %d", encoding);
+		return false;
+	}
+
+	if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16,
+				       channels, &gerror)) {
+		g_warning("%s", gerror->message);
+		g_error_free(gerror);
+		return false;
+	}
+
+	return true;
+}
+
+static void
+mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs)
+{
+	struct audio_format audio_format;
+	mpg123_handle *handle;
+	int error;
+	off_t num_samples;
+	enum decoder_command cmd;
+
+	/* open the file */
+
+	handle = mpg123_new(NULL, &error);
+	if (handle == NULL) {
+		g_warning("mpg123_new() failed: %s",
+			  mpg123_plain_strerror(error));
+		return;
+	}
+
+	if (!mpd_mpg123_open(handle, path_fs, &audio_format)) {
+		mpg123_delete(handle);
+		return;
+	}
+
+	num_samples = mpg123_length(handle);
+
+	/* tell MPD core we're ready */
+
+	decoder_initialized(decoder, &audio_format, false,
+			    (float)num_samples /
+			    (float)audio_format.sample_rate);
+
+	/* the decoder main loop */
+
+	do {
+		unsigned char buffer[8192];
+		size_t nbytes;
+
+		/* decode */
+
+		error = mpg123_read(handle, buffer, sizeof(buffer), &nbytes);
+		if (error != MPG123_OK) {
+			if (error != MPG123_DONE)
+				g_warning("mpg123_read() failed: %s",
+					  mpg123_plain_strerror(error));
+			break;
+		}
+
+		/* send to MPD */
+
+		cmd = decoder_data(decoder, NULL, buffer, nbytes, 0);
+
+		/* seeking not yet implemented */
+	} while (cmd == DECODE_COMMAND_NONE);
+
+	/* cleanup */
+
+	mpg123_delete(handle);
+}
+
+static struct tag *
+mpd_mpg123_tag_dup(const char *path_fs)
+{
+	struct audio_format audio_format;
+	mpg123_handle *handle;
+	int error;
+	off_t num_samples;
+	struct tag *tag;
+
+	handle = mpg123_new(NULL, &error);
+	if (handle == NULL) {
+		g_warning("mpg123_new() failed: %s",
+			  mpg123_plain_strerror(error));
+		return NULL;
+	}
+
+	if (!mpd_mpg123_open(handle, path_fs, &audio_format)) {
+		mpg123_delete(handle);
+		return NULL;
+	}
+
+	num_samples = mpg123_length(handle);
+	if (num_samples <= 0) {
+		mpg123_delete(handle);
+		return NULL;
+	}
+
+	tag = tag_new();
+
+	tag->time = num_samples / audio_format.sample_rate;
+
+	/* ID3 tag support not yet implemented */
+
+	mpg123_delete(handle);
+	return tag;
+}
+
+static const char *const mpg123_suffixes[] = {
+	"mp3",
+	NULL
+};
+
+const struct decoder_plugin mpg123_decoder_plugin = {
+	.name = "mpg123",
+	.init = mpd_mpg123_init,
+	.finish = mpd_mpg123_finish,
+	.file_decode = mpd_mpg123_file_decode,
+	/* streaming not yet implemented */
+	.tag_dup = mpd_mpg123_tag_dup,
+	.suffixes = mpg123_suffixes,
+};
diff --git a/src/decoder/oggflac_plugin.c b/src/decoder/oggflac_decoder_plugin.c
index bdd589ccb..7e5f48318 100644
--- a/src/decoder/oggflac_plugin.c
+++ b/src/decoder/oggflac_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -21,19 +21,18 @@
  * OggFLAC support (half-stolen from flac_plugin.c :))
  */
 
+#include "config.h" /* must be first for large file support */
 #include "_flac_common.h"
 #include "_ogg_common.h"
+#include "flac_metadata.h"
 
 #include <glib.h>
 #include <OggFLAC/seekable_stream_decoder.h>
 #include <assert.h>
 #include <unistd.h>
 
-static void oggflac_cleanup(struct flac_data *data,
-			    OggFLAC__SeekableStreamDecoder * decoder)
+static void oggflac_cleanup(OggFLAC__SeekableStreamDecoder * decoder)
 {
-	if (data->replay_gain_info)
-		replay_gain_info_free(data->replay_gain_info);
 	if (decoder)
 		OggFLAC__seekable_stream_decoder_delete(decoder);
 }
@@ -67,7 +66,7 @@ static OggFLAC__SeekableStreamDecoderSeekStatus of_seek_cb(G_GNUC_UNUSED const
 {
 	struct flac_data *data = (struct flac_data *) fdata;
 
-	if (!input_stream_seek(data->input_stream, offset, SEEK_SET))
+	if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL))
 		return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR;
 
 	return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK;
@@ -156,13 +155,8 @@ oggflac_write_cb(G_GNUC_UNUSED const OggFLAC__SeekableStreamDecoder *decoder,
 		 void *vdata)
 {
 	struct flac_data *data = (struct flac_data *) vdata;
-	FLAC__uint32 samples = frame->header.blocksize;
-	float time_change;
 
-	time_change = ((float)samples) / frame->header.sample_rate;
-	data->time += time_change;
-
-	return flac_common_write(data, frame, buf);
+	return flac_common_write(data, frame, buf, 0);
 }
 
 /* used by TagDup */
@@ -173,17 +167,7 @@ static void of_metadata_dup_cb(G_GNUC_UNUSED const OggFLAC__SeekableStreamDecode
 
 	assert(data->tag != NULL);
 
-	switch (block->type) {
-	case FLAC__METADATA_TYPE_STREAMINFO:
-		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:
-		flac_vorbis_comments_to_tag(data->tag, NULL, block);
-	default:
-		break;
-	}
+	flac_tag_apply_metadata(data->tag, NULL, block);
 }
 
 /* used by decode */
@@ -259,24 +243,20 @@ fail:
 
 /* public functions: */
 static struct tag *
-oggflac_tag_dup(const char *file)
+oggflac_stream_tag(struct input_stream *is)
 {
-	struct input_stream input_stream;
 	OggFLAC__SeekableStreamDecoder *decoder;
 	struct flac_data data;
+	struct tag *tag;
 
-	if (!input_stream_open(&input_stream, file))
-		return NULL;
-	if (ogg_stream_type_detect(&input_stream) != FLAC) {
-		input_stream_close(&input_stream);
+	if (ogg_stream_type_detect(is) != FLAC)
 		return NULL;
-	}
 
 	/* rewind the stream, because ogg_stream_type_detect() has
 	   moved it */
-	input_stream_seek(&input_stream, 0, SEEK_SET);
+	input_stream_seek(is, 0, SEEK_SET, NULL);
 
-	flac_data_init(&data, NULL, &input_stream);
+	flac_data_init(&data, NULL, is);
 
 	data.tag = tag_new();
 
@@ -284,15 +264,17 @@ oggflac_tag_dup(const char *file)
 	 * 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);
-	input_stream_close(&input_stream);
+	oggflac_cleanup(decoder);
 
-	if (!tag_is_defined(data.tag)) {
-		tag_free(data.tag);
+	if (tag_is_defined(data.tag)) {
+		tag = data.tag;
 		data.tag = NULL;
-	}
+	} else
+		tag = NULL;
 
-	return data.tag;
+	flac_data_deinit(&data);
+
+	return tag;
 }
 
 static void
@@ -300,13 +282,14 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream)
 {
 	OggFLAC__SeekableStreamDecoder *decoder = NULL;
 	struct flac_data data;
+	struct audio_format audio_format;
 
 	if (ogg_stream_type_detect(input_stream) != FLAC)
 		return;
 
 	/* rewind the stream, because ogg_stream_type_detect() has
 	   moved it */
-	input_stream_seek(input_stream, 0, SEEK_SET);
+	input_stream_seek(input_stream, 0, SEEK_SET, NULL);
 
 	flac_data_init(&data, mpd_decoder, input_stream);
 
@@ -314,16 +297,13 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream)
 		goto fail;
 	}
 
-	if (!audio_format_valid(&data.audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  data.audio_format.sample_rate,
-			  data.audio_format.bits,
-			  data.audio_format.channels);
+	if (!data.initialized)
 		goto fail;
-	}
 
-	decoder_initialized(mpd_decoder, &data.audio_format,
-			    input_stream->seekable, data.total_time);
+	decoder_initialized(mpd_decoder, &audio_format,
+			    input_stream->seekable,
+			    (float)data.total_frames /
+			    (float)data.audio_format.sample_rate);
 
 	while (true) {
 		OggFLAC__seekable_stream_decoder_process_single(decoder);
@@ -333,11 +313,10 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream)
 		}
 		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
 			FLAC__uint64 seek_sample = decoder_seek_where(mpd_decoder) *
-			    data.audio_format.sample_rate + 0.5;
+			    data.audio_format.sample_rate;
 			if (OggFLAC__seekable_stream_decoder_seek_absolute
 			    (decoder, seek_sample)) {
-				data.time = ((float)seek_sample) /
-				    data.audio_format.sample_rate;
+				data.next_frame = seek_sample;
 				data.position = 0;
 				decoder_command_finished(mpd_decoder);
 			} else
@@ -352,7 +331,8 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream)
 	}
 
 fail:
-	oggflac_cleanup(&data, decoder);
+	oggflac_cleanup(decoder);
+	flac_data_deinit(&data);
 }
 
 static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL };
@@ -368,7 +348,7 @@ static const char *const oggflac_mime_types[] = {
 const struct decoder_plugin oggflac_decoder_plugin = {
 	.name = "oggflac",
 	.stream_decode = oggflac_decode,
-	.tag_dup = oggflac_tag_dup,
+	.stream_tag = oggflac_stream_tag,
 	.suffixes = oggflac_suffixes,
 	.mime_types = oggflac_mime_types
 };
diff --git a/src/decoder/sidplay_decoder_plugin.cxx b/src/decoder/sidplay_decoder_plugin.cxx
new file mode 100644
index 000000000..6fceeb30f
--- /dev/null
+++ b/src/decoder/sidplay_decoder_plugin.cxx
@@ -0,0 +1,429 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+
+extern "C" {
+#include "../decoder_api.h"
+}
+
+#include <errno.h>
+#include <stdlib.h>
+#include <glib.h>
+
+#include <sidplay/sidplay2.h>
+#include <sidplay/builders/resid.h>
+#include <sidplay/utils/SidTuneMod.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "sidplay"
+
+#define SUBTUNE_PREFIX "tune_"
+
+static GPatternSpec *path_with_subtune;
+static const char *songlength_file;
+static GKeyFile *songlength_database;
+
+static bool all_files_are_containers;
+static unsigned default_songlength;
+
+static bool filter_setting;
+
+static GKeyFile *
+sidplay_load_songlength_db(const char *path)
+{
+	GError *error = NULL;
+	gchar *data;
+	gsize size;
+
+	if (!g_file_get_contents(path, &data, &size, &error)) {
+		g_warning("unable to read songlengths file %s: %s",
+			  path, error->message);
+		g_error_free(error);
+		return NULL;
+	}
+
+	/* replace any ; comment characters with # */
+	for (gsize i = 0; i < size; i++)
+		if (data[i] == ';')
+			data[i] = '#';
+
+	GKeyFile *db = g_key_file_new();
+	bool success = g_key_file_load_from_data(db, data, size,
+						 G_KEY_FILE_NONE, &error);
+	g_free(data);
+	if (!success) {
+		g_warning("unable to parse songlengths file %s: %s",
+			  path, error->message);
+		g_error_free(error);
+		g_key_file_free(db);
+		return NULL;
+	}
+
+	g_key_file_set_list_separator(db, ' ');
+	return db;
+}
+
+static bool
+sidplay_init(const struct config_param *param)
+{
+	/* read the songlengths database file */
+	songlength_file=config_get_block_string(param,
+		"songlength_database", NULL);
+	if (songlength_file != NULL)
+		songlength_database = sidplay_load_songlength_db(songlength_file);
+
+	default_songlength=config_get_block_unsigned(param,
+		"default_songlength", 0);
+
+	all_files_are_containers=config_get_block_bool(param,
+		"all_files_are_containers", true);
+
+	path_with_subtune=g_pattern_spec_new(
+			"*/" SUBTUNE_PREFIX "???.sid");
+
+	filter_setting=config_get_block_bool(param, "filter", true);
+
+	return true;
+}
+
+void
+sidplay_finish()
+{
+	g_pattern_spec_free(path_with_subtune);
+
+	if(songlength_database)
+		g_key_file_free(songlength_database);
+}
+
+/**
+ * returns the file path stripped of any /tune_xxx.sid subtune
+ * suffix
+ */
+static char *
+get_container_name(const char *path_fs)
+{
+	char *path_container=g_strdup(path_fs);
+
+	if(!g_pattern_match(path_with_subtune,
+		strlen(path_container), path_container, NULL))
+		return path_container;
+
+	char *ptr=g_strrstr(path_container, "/" SUBTUNE_PREFIX);
+	if(ptr) *ptr='\0';
+
+	return path_container;
+}
+
+/**
+ * returns tune number from file.sid/tune_xxx.sid style path or 1 if
+ * no subtune is appended
+ */
+static int
+get_song_num(const char *path_fs)
+{
+	if(g_pattern_match(path_with_subtune,
+		strlen(path_fs), path_fs, NULL)) {
+		char *sub=g_strrstr(path_fs, "/" SUBTUNE_PREFIX);
+		if(!sub) return 1;
+
+		sub+=strlen("/" SUBTUNE_PREFIX);
+		int song_num=strtol(sub, NULL, 10);
+
+		if (errno == EINVAL)
+			return 1;
+		else
+			return song_num;
+	} else
+		return 1;
+}
+
+/* get the song length in seconds */
+static int
+get_song_length(const char *path_fs)
+{
+	if (songlength_database == NULL)
+		return -1;
+
+	gchar *sid_file=get_container_name(path_fs);
+	SidTuneMod tune(sid_file);
+	g_free(sid_file);
+	if(!tune) {
+		g_warning("failed to load file for calculating md5 sum");
+		return -1;
+	}
+	char md5sum[SIDTUNE_MD5_LENGTH+1];
+	tune.createMD5(md5sum);
+
+	int song_num=get_song_num(path_fs);
+
+	gsize num_items;
+	gchar **values=g_key_file_get_string_list(songlength_database,
+		"Database", md5sum, &num_items, NULL);
+	if(!values || song_num>num_items) {
+		g_strfreev(values);
+		return -1;
+	}
+
+	int minutes=strtol(values[song_num-1], NULL, 10);
+	if(errno==EINVAL) minutes=0;
+
+	int seconds;
+	char *ptr=strchr(values[song_num-1], ':');
+	if(ptr) {
+		seconds=strtol(ptr+1, NULL, 10);
+		if(errno==EINVAL) seconds=0;
+	} else
+		seconds=0;
+
+	g_strfreev(values);
+
+	return (minutes*60)+seconds;
+}
+
+static void
+sidplay_file_decode(struct decoder *decoder, const char *path_fs)
+{
+	int ret;
+	int channels;
+
+	/* load the tune */
+
+	char *path_container=get_container_name(path_fs);
+	SidTune tune(path_container, NULL, true);
+	g_free(path_container);
+	if (!tune) {
+		g_warning("failed to load file");
+		return;
+	}
+
+	int song_num=get_song_num(path_fs);
+	tune.selectSong(song_num);
+
+	int song_len=get_song_length(path_fs);
+	if(song_len==-1) song_len=default_songlength;
+
+	/* initialize the player */
+
+	sidplay2 player;
+	int iret = player.load(&tune);
+	if (iret != 0) {
+		g_warning("sidplay2.load() failed: %s", player.error());
+		return;
+	}
+
+	/* initialize the builder */
+
+	ReSIDBuilder builder("ReSID");
+	if (!builder) {
+		g_warning("failed to initialize ReSIDBuilder");
+		return;
+	}
+
+	builder.create(player.info().maxsids);
+	if (!builder) {
+		g_warning("ReSIDBuilder.create() failed");
+		return;
+	}
+
+	builder.filter(filter_setting);
+	if (!builder) {
+		g_warning("ReSIDBuilder.filter() failed");
+		return;
+	}
+
+	/* configure the player */
+
+	sid2_config_t config = player.config();
+
+	config.clockDefault = SID2_CLOCK_PAL;
+	config.clockForced = true;
+	config.clockSpeed = SID2_CLOCK_CORRECT;
+	config.frequency = 48000;
+	config.optimisation = SID2_DEFAULT_OPTIMISATION;
+
+	config.precision = 16;
+	config.sidDefault = SID2_MOS6581;
+	config.sidEmulation = &builder;
+	config.sidModel = SID2_MODEL_CORRECT;
+	config.sidSamples = true;
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+	config.sampleFormat = SID2_LITTLE_SIGNED;
+#else
+	config.sampleFormat = SID2_BIG_SIGNED;
+#endif
+	if (tune.isStereo()) {
+		config.playback = sid2_stereo;
+		channels = 2;
+	} else {
+		config.playback = sid2_mono;
+		channels = 1;
+	}
+
+	iret = player.config(config);
+	if (iret != 0) {
+		g_warning("sidplay2.config() failed: %s", player.error());
+		return;
+	}
+
+	/* initialize the MPD decoder */
+
+	struct audio_format audio_format;
+	audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, channels);
+	assert(audio_format_valid(&audio_format));
+
+	decoder_initialized(decoder, &audio_format, true, (float)song_len);
+
+	/* .. and play */
+
+	const unsigned timebase = player.timebase();
+	song_len *= timebase;
+
+	enum decoder_command cmd;
+	do {
+		char buffer[4096];
+		size_t nbytes;
+
+		nbytes = player.play(buffer, sizeof(buffer));
+		if (nbytes == 0)
+			break;
+
+		decoder_timestamp(decoder, (double)player.time() / timebase);
+
+		cmd = decoder_data(decoder, NULL, buffer, nbytes, 0);
+
+		if(cmd==DECODE_COMMAND_SEEK) {
+			unsigned data_time = player.time();
+			unsigned target_time = (unsigned)
+				(decoder_seek_where(decoder) * timebase);
+
+			/* can't rewind so return to zero and seek forward */
+			if(target_time<data_time) {
+				player.stop();
+				data_time=0;
+			}
+
+			/* ignore data until target time is reached */
+			while(data_time<target_time) {
+				nbytes=player.play(buffer, sizeof(buffer));
+				if(nbytes==0)
+					break;
+				data_time = player.time();
+			}
+
+			decoder_command_finished(decoder);
+		}
+
+		if (song_len > 0 && player.time() >= song_len)
+			break;
+
+	} while (cmd != DECODE_COMMAND_STOP);
+}
+
+static struct tag *
+sidplay_tag_dup(const char *path_fs)
+{
+	int song_num=get_song_num(path_fs);
+	char *path_container=get_container_name(path_fs);
+
+	SidTune tune(path_container, NULL, true);
+	g_free(path_container);
+	if (!tune)
+		return NULL;
+
+	const SidTuneInfo &info = tune.getInfo();
+	struct tag *tag = tag_new();
+
+	/* title */
+	const char *title;
+	if (info.numberOfInfoStrings > 0 && info.infoString[0] != NULL)
+		title=info.infoString[0];
+	else
+		title="";
+
+	if(info.songs>1) {
+		char *tag_title=g_strdup_printf("%s (%d/%d)",
+			title, song_num, info.songs);
+		tag_add_item(tag, TAG_TITLE, tag_title);
+		g_free(tag_title);
+	} else
+		tag_add_item(tag, TAG_TITLE, title);
+
+	/* artist */
+	if (info.numberOfInfoStrings > 1 && info.infoString[1] != NULL)
+		tag_add_item(tag, TAG_ARTIST, info.infoString[1]);
+
+	/* track */
+	char *track=g_strdup_printf("%d", song_num);
+	tag_add_item(tag, TAG_TRACK, track);
+	g_free(track);
+
+	/* time */
+	int song_len=get_song_length(path_fs);
+	if(song_len!=-1) tag->time=song_len;
+
+	return tag;
+}
+
+static char *
+sidplay_container_scan(const char *path_fs, const unsigned int tnum)
+{
+	SidTune tune(path_fs, NULL, true);
+	if (!tune)
+		return NULL;
+
+	const SidTuneInfo &info=tune.getInfo();
+
+	/* Don't treat sids containing a single tune
+		as containers */
+	if(!all_files_are_containers && info.songs<2)
+		return NULL;
+
+	/* Construct container/tune path names, eg.
+		Delta.sid/tune_001.sid */
+	if(tnum<=info.songs) {
+		char *subtune= g_strdup_printf(
+			SUBTUNE_PREFIX "%03u.sid", tnum);
+		return subtune;
+	} else
+		return NULL;
+}
+
+static const char *const sidplay_suffixes[] = {
+	"sid",
+	"mus",
+	"str",
+	"prg",
+	"P00",
+	NULL
+};
+
+extern const struct decoder_plugin sidplay_decoder_plugin;
+const struct decoder_plugin sidplay_decoder_plugin = {
+	"sidplay",
+	sidplay_init,
+	sidplay_finish,
+	NULL, /* stream_decode() */
+	sidplay_file_decode,
+	sidplay_tag_dup,
+	NULL, /* stream_tag() */
+	sidplay_container_scan,
+	sidplay_suffixes,
+	NULL, /* mime_types */
+};
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
deleted file mode 100644
index c62e6b4b6..000000000
--- a/src/decoder/sidplay_plugin.cxx
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
- * 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.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-extern "C" {
-#include "../decoder_api.h"
-}
-
-#include <glib.h>
-
-#include <sidplay/sidplay2.h>
-#include <sidplay/builders/resid.h>
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "sidplay"
-
-static void
-sidplay_file_decode(struct decoder *decoder, const char *path_fs)
-{
-	int ret;
-
-	/* load the tune */
-
-	SidTune tune(path_fs, NULL, true);
-	if (!tune) {
-		g_warning("failed to load file");
-		return;
-	}
-
-	tune.selectSong(1);
-
-	/* initialize the player */
-
-	sidplay2 player;
-	int iret = player.load(&tune);
-	if (iret != 0) {
-		g_warning("sidplay2.load() failed: %s", player.error());
-		return;
-	}
-
-	/* initialize the builder */
-
-	ReSIDBuilder builder("ReSID");
-	if (!builder) {
-		g_warning("failed to initialize ReSIDBuilder");
-		return;
-	}
-
-	builder.create(player.info().maxsids);
-	if (!builder) {
-		g_warning("ReSIDBuilder.create() failed");
-		return;
-	}
-
-	builder.filter(false);
-	if (!builder) {
-		g_warning("ReSIDBuilder.filter() failed");
-		return;
-	}
-
-	/* configure the player */
-
-	sid2_config_t config = player.config();
-
-	config.clockDefault = SID2_CLOCK_PAL;
-	config.clockForced = true;
-	config.clockSpeed = SID2_CLOCK_CORRECT;
-	config.frequency = 48000;
-	config.optimisation = SID2_DEFAULT_OPTIMISATION;
-	config.playback = sid2_stereo;
-	config.precision = 16;
-	config.sidDefault = SID2_MOS6581;
-	config.sidEmulation = &builder;
-	config.sidModel = SID2_MODEL_CORRECT;
-	config.sidSamples = true;
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
-	config.sampleFormat = SID2_LITTLE_SIGNED;
-#else
-	config.sampleFormat = SID2_BIG_SIGNED;
-#endif
-
-	iret = player.config(config);
-	if (iret != 0) {
-		g_warning("sidplay2.config() failed: %s", player.error());
-		return;
-	}
-
-	/* initialize the MPD decoder */
-
-	struct audio_format audio_format;
-	audio_format.sample_rate = 48000;
-	audio_format.bits = 16;
-	audio_format.channels = 2;
-
-	decoder_initialized(decoder, &audio_format, false, -1);
-
-	/* .. and play */
-
-	enum decoder_command cmd;
-	do {
-		char buffer[4096];
-		size_t nbytes;
-
-		nbytes = player.play(buffer, sizeof(buffer));
-		if (nbytes == 0)
-			break;
-
-		cmd = decoder_data(decoder, NULL, buffer, nbytes,
-				   0, 0, NULL);
-	} while (cmd == DECODE_COMMAND_NONE);
-}
-
-static struct tag *
-sidplay_tag_dup(const char *path_fs)
-{
-	SidTune tune(path_fs, NULL, true);
-	if (!tune)
-		return NULL;
-
-	const SidTuneInfo &info = tune.getInfo();
-	struct tag *tag = tag_new();
-
-	if (info.numberOfInfoStrings > 0 && info.infoString[0] != NULL)
-		tag_add_item(tag, TAG_ITEM_TITLE, info.infoString[0]);
-
-	if (info.numberOfInfoStrings > 1 && info.infoString[1] != NULL)
-		tag_add_item(tag, TAG_ITEM_ARTIST, info.infoString[1]);
-
-	return tag;
-}
-
-static const char *const sidplay_suffixes[] = {
-	"sid",
-	NULL
-};
-
-extern const struct decoder_plugin sidplay_decoder_plugin;
-const struct decoder_plugin sidplay_decoder_plugin = {
-	"sidplay",
-	NULL, /* init() */
-	NULL, /* finish() */
-	NULL, /* stream_decode() */
-	sidplay_file_decode,
-	sidplay_tag_dup,
-	NULL, /* container_scan */
-	sidplay_suffixes,
-	NULL, /* mime_types */
-};
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
new file mode 100644
index 000000000..af68f117d
--- /dev/null
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -0,0 +1,251 @@
+/*
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
+
+#include <sndfile.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "sndfile"
+
+static sf_count_t
+sndfile_vio_get_filelen(void *user_data)
+{
+	const struct input_stream *is = user_data;
+
+	return is->size;
+}
+
+static sf_count_t
+sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
+{
+	struct input_stream *is = user_data;
+	bool success;
+
+	success = input_stream_seek(is, offset, whence, NULL);
+	if (!success)
+		return -1;
+
+	return is->offset;
+}
+
+static sf_count_t
+sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
+{
+	struct input_stream *is = user_data;
+	GError *error = NULL;
+	size_t nbytes;
+
+	nbytes = input_stream_read(is, ptr, count, &error);
+	if (nbytes == 0 && error != NULL) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return -1;
+	}
+
+	return nbytes;
+}
+
+static sf_count_t
+sndfile_vio_write(G_GNUC_UNUSED const void *ptr,
+		  G_GNUC_UNUSED sf_count_t count,
+		  G_GNUC_UNUSED void *user_data)
+{
+	/* no writing! */
+	return -1;
+}
+
+static sf_count_t
+sndfile_vio_tell(void *user_data)
+{
+	const struct input_stream *is = user_data;
+
+	return is->offset;
+}
+
+/**
+ * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a
+ * libsndfile stream.
+ */
+static SF_VIRTUAL_IO vio = {
+	.get_filelen = sndfile_vio_get_filelen,
+	.seek = sndfile_vio_seek,
+	.read = sndfile_vio_read,
+	.write = sndfile_vio_write,
+	.tell = sndfile_vio_tell,
+};
+
+/**
+ * Converts a frame number to a timestamp (in seconds).
+ */
+static float
+frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
+{
+	return (float)frame / (float)audio_format->sample_rate;
+}
+
+/**
+ * Converts a timestamp (in seconds) to a frame number.
+ */
+static sf_count_t
+time_to_frame(float t, const struct audio_format *audio_format)
+{
+	return (sf_count_t)(t * audio_format->sample_rate);
+}
+
+static void
+sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
+{
+	GError *error = NULL;
+	SNDFILE *sf;
+	SF_INFO info;
+	struct audio_format audio_format;
+	size_t frame_size;
+	sf_count_t read_frames, num_frames;
+	int buffer[4096];
+	enum decoder_command cmd;
+
+	info.format = 0;
+
+	sf = sf_open_virtual(&vio, SFM_READ, &info, is);
+	if (sf == NULL) {
+		g_warning("sf_open_virtual() failed");
+		return;
+	}
+
+	/* for now, always read 32 bit samples.  Later, we could lower
+	   MPD's CPU usage by reading 16 bit samples with
+	   sf_readf_short() on low-quality source files. */
+	if (!audio_format_init_checked(&audio_format, info.samplerate,
+				       SAMPLE_FORMAT_S32,
+				       info.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return;
+	}
+
+	decoder_initialized(decoder, &audio_format, info.seekable,
+			    frame_to_time(info.frames, &audio_format));
+
+	frame_size = audio_format_frame_size(&audio_format);
+	read_frames = sizeof(buffer) / frame_size;
+
+	do {
+		num_frames = sf_readf_int(sf, buffer, read_frames);
+		if (num_frames <= 0)
+			break;
+
+		cmd = decoder_data(decoder, is,
+				   buffer, num_frames * frame_size,
+				   0);
+		if (cmd == DECODE_COMMAND_SEEK) {
+			sf_count_t c =
+				time_to_frame(decoder_seek_where(decoder),
+					      &audio_format);
+			c = sf_seek(sf, c, SEEK_SET);
+			if (c < 0)
+				decoder_seek_error(decoder);
+			else
+				decoder_command_finished(decoder);
+			cmd = DECODE_COMMAND_NONE;
+		}
+	} while (cmd == DECODE_COMMAND_NONE);
+
+	sf_close(sf);
+}
+
+static struct tag *
+sndfile_tag_dup(const char *path_fs)
+{
+	SNDFILE *sf;
+	SF_INFO info;
+	struct tag *tag;
+	const char *p;
+
+	info.format = 0;
+
+	sf = sf_open(path_fs, SFM_READ, &info);
+	if (sf == NULL)
+		return NULL;
+
+	if (!audio_valid_sample_rate(info.samplerate)) {
+		sf_close(sf);
+		g_warning("Invalid sample rate in %s\n", path_fs);
+		return NULL;
+	}
+
+	tag = tag_new();
+	tag->time = info.frames / info.samplerate;
+
+	p = sf_get_string(sf, SF_STR_TITLE);
+	if (p != NULL)
+		tag_add_item(tag, TAG_TITLE, p);
+
+	p = sf_get_string(sf, SF_STR_ARTIST);
+	if (p != NULL)
+		tag_add_item(tag, TAG_ARTIST, p);
+
+	p = sf_get_string(sf, SF_STR_DATE);
+	if (p != NULL)
+		tag_add_item(tag, TAG_DATE, p);
+
+	sf_close(sf);
+
+	return tag;
+}
+
+static const char *const sndfile_suffixes[] = {
+	"wav", "aiff", "aif", /* Microsoft / SGI / Apple */
+	"au", "snd", /* Sun / DEC / NeXT */
+	"paf", /* Paris Audio File */
+	"iff", "svx", /* Commodore Amiga IFF / SVX */
+	"sf", /* IRCAM */
+	"voc", /* Creative */
+	"w64", /* Soundforge */
+	"pvf", /* Portable Voice Format */
+	"xi", /* Fasttracker */
+	"htk", /* HMM Tool Kit */
+	"caf", /* Apple */
+	"sd2", /* Sound Designer II */
+
+	/* libsndfile also supports FLAC and Ogg Vorbis, but only by
+	   linking with libFLAC and libvorbis - we can do better, we
+	   have native plugins for these libraries */
+
+	NULL
+};
+
+static const char *const sndfile_mime_types[] = {
+	"audio/x-wav",
+	"audio/x-aiff",
+
+	/* what are the MIME types of the other supported formats? */
+
+	NULL
+};
+
+const struct decoder_plugin sndfile_decoder_plugin = {
+	.name = "sndfile",
+	.stream_decode = sndfile_stream_decode,
+	.tag_dup = sndfile_tag_dup,
+	.suffixes = sndfile_suffixes,
+	.mime_types = sndfile_mime_types,
+};
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_decoder_plugin.c
index 7c782a779..0a3944ad6 100644
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,19 +17,19 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-/* TODO 'ogg' should probably be replaced with 'oggvorbis' in all instances */
-
-#include "_ogg_common.h"
 #include "config.h"
+#include "_ogg_common.h"
+#include "audio_check.h"
 #include "uri.h"
 
 #ifndef HAVE_TREMOR
+#define OV_EXCLUDE_STATIC_CALLBACKS
 #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. 
+   milliseconds where libogg uses double seconds.
 */
 #define ov_read(VF, BUFFER, LENGTH, BIGENDIANP, WORD, SGNED, BITSTREAM) \
         ov_read(VF, BUFFER, LENGTH, BITSTREAM)
@@ -55,46 +55,46 @@
 #define OGG_DECODE_USE_BIGENDIAN	0
 #endif
 
-typedef struct _OggCallbackData {
+struct vorbis_input_stream {
 	struct decoder *decoder;
 
 	struct input_stream *input_stream;
 	bool seekable;
-} OggCallbackData;
+};
 
-static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
+static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *data)
 {
+	struct vorbis_input_stream *vis = data;
 	size_t ret;
-	OggCallbackData *data = (OggCallbackData *) vdata;
 
-	ret = decoder_read(data->decoder, data->input_stream, ptr, size * nmemb);
+	ret = decoder_read(vis->decoder, vis->input_stream, ptr, size * nmemb);
 
 	errno = 0;
 
 	return ret / size;
 }
 
-static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence)
+static int ogg_seek_cb(void *data, ogg_int64_t offset, int whence)
 {
-	const OggCallbackData *data = (const OggCallbackData *) vdata;
+	struct vorbis_input_stream *vis = data;
 
-	return data->seekable &&
-		decoder_get_command(data->decoder) != DECODE_COMMAND_STOP &&
-		input_stream_seek(data->input_stream, offset, whence)
+	return vis->seekable &&
+		(!vis->decoder || decoder_get_command(vis->decoder) != DECODE_COMMAND_STOP) &&
+		input_stream_seek(vis->input_stream, offset, whence, NULL)
 		? 0 : -1;
 }
 
 /* TODO: check Ogg libraries API and see if we can just not have this func */
-static int ogg_close_cb(G_GNUC_UNUSED void *vdata)
+static int ogg_close_cb(G_GNUC_UNUSED void *data)
 {
 	return 0;
 }
 
-static long ogg_tell_cb(void *vdata)
+static long ogg_tell_cb(void *data)
 {
-	const OggCallbackData *data = (const OggCallbackData *) vdata;
+	const struct vorbis_input_stream *vis = data;
 
-	return (long)data->input_stream->offset;
+	return (long)vis->input_stream->offset;
 }
 
 static const ov_callbacks vorbis_is_callbacks = {
@@ -105,6 +105,52 @@ static const ov_callbacks vorbis_is_callbacks = {
 };
 
 static const char *
+vorbis_strerror(int code)
+{
+	switch (code) {
+	case OV_EREAD:
+		return "read error";
+
+	case OV_ENOTVORBIS:
+		return "not vorbis stream";
+
+	case OV_EVERSION:
+		return "vorbis version mismatch";
+
+	case OV_EBADHEADER:
+		return "invalid vorbis header";
+
+	case OV_EFAULT:
+		return "internal logic error";
+
+	default:
+		return "unknown error";
+	}
+}
+
+static bool
+vorbis_is_open(struct vorbis_input_stream *vis, OggVorbis_File *vf,
+	       struct decoder *decoder, struct input_stream *input_stream)
+{
+	vis->decoder = decoder;
+	vis->input_stream = input_stream;
+	vis->seekable = input_stream->seekable &&
+		(input_stream->uri == NULL ||
+		 !uri_has_scheme(input_stream->uri));
+
+	int ret = ov_open_callbacks(vis, vf, NULL, 0, vorbis_is_callbacks);
+	if (ret < 0) {
+		if (decoder == NULL ||
+		    decoder_get_command(decoder) == DECODE_COMMAND_NONE)
+			g_warning("Failed to open Ogg Vorbis stream: %s",
+				  vorbis_strerror(ret));
+		return false;
+	}
+
+	return true;
+}
+
+static const char *
 vorbis_comment_value(const char *comment, const char *needle)
 {
 	size_t len = strlen(needle);
@@ -116,14 +162,13 @@ vorbis_comment_value(const char *comment, const char *needle)
 	return NULL;
 }
 
-static struct replay_gain_info *
-vorbis_comments_to_replay_gain(char **comments)
+static bool
+vorbis_comments_to_replay_gain(struct replay_gain_info *rgi, char **comments)
 {
-	struct replay_gain_info *rgi;
 	const char *temp;
 	bool found = false;
 
-	rgi = replay_gain_info_new();
+	replay_gain_info_init(rgi);
 
 	while (*comments) {
 		if ((temp =
@@ -147,12 +192,7 @@ vorbis_comments_to_replay_gain(char **comments)
 		comments++;
 	}
 
-	if (!found) {
-		replay_gain_info_free(rgi);
-		rgi = NULL;
-	}
-
-	return rgi;
+	return found;
 }
 
 static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber";
@@ -183,11 +223,11 @@ vorbis_parse_comment(struct tag *tag, const char *comment)
 	assert(tag != NULL);
 
 	if (vorbis_copy_comment(tag, comment, VORBIS_COMMENT_TRACK_KEY,
-				TAG_ITEM_TRACK) ||
+				TAG_TRACK) ||
 	    vorbis_copy_comment(tag, comment, VORBIS_COMMENT_DISC_KEY,
-				TAG_ITEM_DISC) ||
+				TAG_DISC) ||
 	    vorbis_copy_comment(tag, comment, "album artist",
-				TAG_ITEM_ALBUM_ARTIST))
+				TAG_ALBUM_ARTIST))
 		return;
 
 	for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i)
@@ -226,42 +266,23 @@ vorbis_send_comments(struct decoder *decoder, struct input_stream *is,
 	tag_free(tag);
 }
 
-static bool
-oggvorbis_seekable(struct decoder *decoder)
-{
-	char *uri;
-	bool seekable;
-
-	uri = decoder_get_uri(decoder);
-	if (uri == NULL)
-		return false;
-
-	/* disable seeking on remote streams, because libvorbis seeks
-	   around like crazy, and due to being very expensive, this
-	   delays song playback my 10 or 20 seconds */
-	seekable = !uri_has_scheme(uri);
-	g_free(uri);
-
-	return seekable;
-}
-
 /* public */
 static void
 vorbis_stream_decode(struct decoder *decoder,
 		     struct input_stream *input_stream)
 {
+	GError *error = NULL;
 	OggVorbis_File vf;
-	OggCallbackData data;
+	struct vorbis_input_stream vis;
 	struct audio_format audio_format;
+	float total_time;
 	int current_section;
 	int prev_section = -1;
 	long ret;
 	char chunk[OGG_CHUNK_SIZE];
 	long bitRate = 0;
 	long test;
-	struct replay_gain_info *replay_gain_info = NULL;
-	char **comments;
-	bool initialized = false;
+	const vorbis_info *vi;
 	enum decoder_command cmd = DECODE_COMMAND_NONE;
 
 	if (ogg_stream_type_detect(input_stream) != VORBIS)
@@ -269,43 +290,30 @@ vorbis_stream_decode(struct decoder *decoder,
 
 	/* rewind the stream, because ogg_stream_type_detect() has
 	   moved it */
-	input_stream_seek(input_stream, 0, SEEK_SET);
-
-	data.decoder = decoder;
-	data.input_stream = input_stream;
-	data.seekable = input_stream->seekable && oggvorbis_seekable(decoder);
+	input_stream_seek(input_stream, 0, SEEK_SET, NULL);
 
-	if ((ret = ov_open_callbacks(&data, &vf, NULL, 0,
-				     vorbis_is_callbacks)) < 0) {
-		const char *error;
-		if (decoder_get_command(decoder) != DECODE_COMMAND_NONE)
-			return;
+	if (!vorbis_is_open(&vis, &vf, decoder, input_stream))
+		return;
 
-		switch (ret) {
-		case OV_EREAD:
-			error = "read error";
-			break;
-		case OV_ENOTVORBIS:
-			error = "not vorbis stream";
-			break;
-		case OV_EVERSION:
-			error = "vorbis version mismatch";
-			break;
-		case OV_EBADHEADER:
-			error = "invalid vorbis header";
-			break;
-		case OV_EFAULT:
-			error = "internal logic error";
-			break;
-		default:
-			error = "unknown error";
-			break;
-		}
+	vi = ov_info(&vf, -1);
+	if (vi == NULL) {
+		g_warning("ov_info() has failed");
+		return;
+	}
 
-		g_warning("Error decoding Ogg Vorbis stream: %s", error);
+	if (!audio_format_init_checked(&audio_format, vi->rate,
+				       SAMPLE_FORMAT_S16,
+				       vi->channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return;
 	}
-	audio_format.bits = 16;
+
+	total_time = ov_time_total(&vf, -1);
+	if (total_time < 0)
+		total_time = 0;
+
+	decoder_initialized(decoder, &audio_format, vis.seekable, total_time);
 
 	do {
 		if (cmd == DECODE_COMMAND_SEEK) {
@@ -325,83 +333,61 @@ vorbis_stream_decode(struct decoder *decoder,
 			break;
 
 		if (current_section != prev_section) {
-			/*printf("new song!\n"); */
-			vorbis_info *vi = ov_info(&vf, -1);
-			struct replay_gain_info *new_rgi;
-
-			audio_format.channels = vi->channels;
-			audio_format.sample_rate = vi->rate;
-
-			if (!audio_format_valid(&audio_format)) {
-				g_warning("Invalid audio format: %u:%u:%u\n",
-					  audio_format.sample_rate,
-					  audio_format.bits,
-					  audio_format.channels);
+			char **comments;
+
+			vi = ov_info(&vf, -1);
+			if (vi == NULL) {
+				g_warning("ov_info() has failed");
 				break;
 			}
 
-			if (!initialized) {
-				float total_time = ov_time_total(&vf, -1);
-				if (total_time < 0)
-					total_time = 0;
-				decoder_initialized(decoder, &audio_format,
-						    data.seekable,
-						    total_time);
-				initialized = true;
+			if (vi->rate != (long)audio_format.sample_rate ||
+			    vi->channels != (int)audio_format.channels) {
+				/* we don't support audio format
+				   change yet */
+				g_warning("audio format change, stopping here");
+				break;
 			}
+
 			comments = ov_comment(&vf, -1)->user_comments;
 			vorbis_send_comments(decoder, input_stream, comments);
-			new_rgi = vorbis_comments_to_replay_gain(comments);
-			if (new_rgi != NULL) {
-				if (replay_gain_info != NULL)
-					replay_gain_info_free(replay_gain_info);
-				replay_gain_info = new_rgi;
-			}
-		}
 
-		prev_section = current_section;
+			struct replay_gain_info rgi;
+			if (vorbis_comments_to_replay_gain(&rgi, comments))
+				decoder_replay_gain(decoder, &rgi);
+
+			prev_section = current_section;
+		}
 
 		if ((test = ov_bitrate_instant(&vf)) > 0)
 			bitRate = test / 1000;
 
 		cmd = decoder_data(decoder, input_stream,
 				   chunk, ret,
-				   ov_pcm_tell(&vf) / audio_format.sample_rate,
-				   bitRate, replay_gain_info);
+				   bitRate);
 	} while (cmd != DECODE_COMMAND_STOP);
 
-	if (replay_gain_info)
-		replay_gain_info_free(replay_gain_info);
-
 	ov_clear(&vf);
 }
 
 static struct tag *
-vorbis_tag_dup(const char *file)
+vorbis_stream_tag(struct input_stream *is)
 {
-	struct tag *ret;
-	FILE *fp;
+	struct vorbis_input_stream vis;
 	OggVorbis_File vf;
 
-	fp = fopen(file, "r");
-	if (!fp) {
+	if (!vorbis_is_open(&vis, &vf, NULL, is))
 		return NULL;
-	}
 
-	if (ov_open(fp, &vf, NULL, 0) < 0) {
-		fclose(fp);
-		return NULL;
-	}
-
-	ret = vorbis_comments_to_tag(ov_comment(&vf, -1)->user_comments);
+	struct tag *tag = vorbis_comments_to_tag(ov_comment(&vf, -1)->user_comments);
 
-	if (!ret)
-		ret = tag_new();
-	ret->time = (int)(ov_time_total(&vf, -1) + 0.5);
+	if (tag == NULL)
+		tag = tag_new();
+	tag->time = (int)(ov_time_total(&vf, -1) + 0.5);
 
 	ov_clear(&vf);
 
-	return ret;
+	return tag;
 }
 
 static const char *const vorbis_suffixes[] = {
@@ -423,7 +409,7 @@ static const char *const vorbis_mime_types[] = {
 const struct decoder_plugin vorbis_decoder_plugin = {
 	.name = "vorbis",
 	.stream_decode = vorbis_stream_decode,
-	.tag_dup = vorbis_tag_dup,
+	.stream_tag = vorbis_stream_tag,
 	.suffixes = vorbis_suffixes,
 	.mime_types = vorbis_mime_types
 };
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_decoder_plugin.c
index 7ad3a62b0..efed98851 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,9 +17,11 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
-#include "../path.h"
-#include "../utils.h"
+#include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
+#include "path.h"
+#include "utils.h"
 
 #include <wavpack/wavpack.h>
 #include <glib.h>
@@ -41,17 +43,17 @@ static struct {
 	const char *name;
 	enum tag_type 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 },
+	{ "artist", TAG_ARTIST },
+	{ "album", TAG_ALBUM },
+	{ "title", TAG_TITLE },
+	{ "track", TAG_TRACK },
+	{ "name", TAG_NAME },
+	{ "genre", TAG_GENRE },
+	{ "date", TAG_DATE },
+	{ "composer", TAG_COMPOSER },
+	{ "performer", TAG_PERFORMER },
+	{ "comment", TAG_COMMENT },
+	{ "disc", TAG_DISC },
 };
 
 /** A pointer type for format converter function. */
@@ -97,19 +99,11 @@ format_samples_int(int bytes_per_sample, void *buffer, uint32_t count)
 		}
 		break;
 	}
+
 	case 3:
+	case 4:
 		/* do nothing */
 		break;
-	case 4: {
-		uint32_t *dst = buffer;
-		assert_static(sizeof(*dst) <= sizeof(*src));
-
-		/* downsample to 24-bit */
-		while (count--) {
-			*dst++ = *src++ >> 8;
-		}
-		break;
-	}
 	}
 }
 
@@ -129,38 +123,61 @@ format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer,
 	}
 }
 
+/**
+ * Choose a MPD sample format from libwavpacks' number of bits.
+ */
+static enum sample_format
+wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample)
+{
+	if (is_float)
+		return SAMPLE_FORMAT_S24_P32;
+
+	switch (bytes_per_sample) {
+	case 1:
+		return SAMPLE_FORMAT_S8;
+
+	case 2:
+		return SAMPLE_FORMAT_S16;
+
+	case 3:
+		return SAMPLE_FORMAT_S24_P32;
+
+	case 4:
+		return SAMPLE_FORMAT_S32;
+
+	default:
+		return SAMPLE_FORMAT_UNDEFINED;
+	}
+}
+
 /*
  * This does the main decoding thing.
  * Requires an already opened WavpackContext.
  */
 static void
-wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
-	       struct replay_gain_info *replay_gain_info)
+wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 {
+	GError *error = NULL;
+	bool is_float;
+	enum sample_format sample_format;
 	struct audio_format audio_format;
 	format_samples_t format_samples;
 	char chunk[CHUNK_SIZE];
 	int samples_requested, samples_got;
-	float total_time, current_time;
+	float total_time;
 	int bytes_per_sample, output_sample_size;
-	int position;
 
-	audio_format.sample_rate = WavpackGetSampleRate(wpc);
-	audio_format.channels = WavpackGetReducedChannels(wpc);
-	audio_format.bits = WavpackGetBitsPerSample(wpc);
-
-	/* round bitwidth to 8-bit units */
-	audio_format.bits = (audio_format.bits + 7) & (~7);
-	/* mpd handles max 24-bit samples */
-	if (audio_format.bits > 24) {
-		audio_format.bits = 24;
-	}
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate,
-			  audio_format.bits,
-			  audio_format.channels);
+	is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0;
+	sample_format =
+		wavpack_bits_to_sample_format(is_float,
+					      WavpackGetBytesPerSample(wpc));
+
+	if (!audio_format_init_checked(&audio_format,
+				       WavpackGetSampleRate(wpc),
+				       sample_format,
+				       WavpackGetNumChannels(wpc), &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return;
 	}
 
@@ -180,8 +197,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 
 	decoder_initialized(decoder, &audio_format, can_seek, total_time);
 
-	position = 0;
-
 	do {
 		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
 			if (can_seek) {
@@ -189,7 +204,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 					audio_format.sample_rate;
 
 				if (WavpackSeekSample(wpc, where)) {
-					position = where;
 					decoder_command_finished(decoder);
 				} else {
 					decoder_seek_error(decoder);
@@ -209,9 +223,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 		if (samples_got > 0) {
 			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
 			              1000 + 0.5);
-			position += samples_got;
-			current_time = position;
-			current_time /= audio_format.sample_rate;
 
 			format_samples(
 				bytes_per_sample, chunk,
@@ -221,8 +232,7 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 			decoder_data(
 				decoder, NULL, chunk,
 				samples_got * output_sample_size,
-				current_time, bitrate,
-				replay_gain_info
+				bitrate
 			);
 		}
 	} while (samples_got > 0);
@@ -246,13 +256,13 @@ wavpack_tag_float(WavpackContext *wpc, const char *key, float *value_r)
 	return true;
 }
 
-static struct replay_gain_info *
-wavpack_replaygain(WavpackContext *wpc)
+static bool
+wavpack_replaygain(struct replay_gain_info *replay_gain_info,
+		   WavpackContext *wpc)
 {
-	struct replay_gain_info *replay_gain_info;
 	bool found = false;
 
-	replay_gain_info = replay_gain_info_new();
+	replay_gain_info_init(replay_gain_info);
 
 	found |= wavpack_tag_float(
 		wpc, "replaygain_track_gain",
@@ -271,13 +281,7 @@ wavpack_replaygain(WavpackContext *wpc)
 		&replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak
 	);
 
-	if (found) {
-		return replay_gain_info;
-	}
-
-	replay_gain_info_free(replay_gain_info);
-
-	return NULL;
+	return found;
 }
 
 /*
@@ -397,13 +401,13 @@ wavpack_input_get_pos(void *id)
 static int
 wavpack_input_set_pos_abs(void *id, uint32_t pos)
 {
-	return input_stream_seek(wpin(id)->is, pos, SEEK_SET) ? 0 : -1;
+	return input_stream_seek(wpin(id)->is, pos, SEEK_SET, NULL) ? 0 : -1;
 }
 
 static int
 wavpack_input_set_pos_rel(void *id, int32_t delta, int mode)
 {
-	return input_stream_seek(wpin(id)->is, delta, mode) ? 0 : -1;
+	return input_stream_seek(wpin(id)->is, delta, mode, NULL) ? 0 : -1;
 }
 
 static int
@@ -452,13 +456,12 @@ wavpack_input_init(struct wavpack_input *isp, struct decoder *decoder,
 	isp->last_byte = EOF;
 }
 
-static bool
-wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc,
+static struct input_stream *
+wavpack_open_wvc(struct decoder *decoder, const char *uri,
 		 struct wavpack_input *wpi)
 {
-	char *utf8url;
+	struct input_stream *is_wvc;
 	char *wvc_url = NULL;
-	bool ret;
 	char first_byte;
 	size_t nbytes;
 
@@ -466,20 +469,15 @@ wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc,
 	 * As we use dc->utf8url, this function will be bad for
 	 * single files. utf8url is not absolute file path :/
 	 */
-	utf8url = decoder_get_uri(decoder);
-	if (utf8url == NULL) {
+	if (uri == NULL)
 		return false;
-	}
 
-	wvc_url = g_strconcat(utf8url, "c", NULL);
-	g_free(utf8url);
-
-	ret = input_stream_open(is_wvc, wvc_url);
+	wvc_url = g_strconcat(uri, "c", NULL);
+	is_wvc = input_stream_open(wvc_url, NULL);
 	g_free(wvc_url);
 
-	if (!ret) {
-		return false;
-	}
+	if (is_wvc == NULL)
+		return NULL;
 
 	/*
 	 * And we try to buffer in order to get know
@@ -490,13 +488,13 @@ wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc,
 	);
 	if (nbytes == 0) {
 		input_stream_close(is_wvc);
-		return false;
+		return NULL;
 	}
 
 	/* push it back */
 	wavpack_input_init(wpi, decoder, is_wvc);
 	wpi->last_byte = first_byte;
-	return true;
+	return is_wvc;
 }
 
 /*
@@ -507,14 +505,15 @@ wavpack_streamdecode(struct decoder * decoder, struct input_stream *is)
 {
 	char error[ERRORLEN];
 	WavpackContext *wpc;
-	struct input_stream is_wvc;
-	int open_flags = OPEN_2CH_MAX | OPEN_NORMALIZE;
+	struct input_stream *is_wvc;
+	int open_flags = OPEN_NORMALIZE;
 	struct wavpack_input isp, isp_wvc;
 	bool can_seek = is->seekable;
 
-	if (wavpack_open_wvc(decoder, &is_wvc, &isp_wvc)) {
+	is_wvc = wavpack_open_wvc(decoder, is->uri, &isp_wvc);
+	if (is_wvc != NULL) {
 		open_flags |= OPEN_WVC;
-		can_seek &= is_wvc.seekable;
+		can_seek &= is_wvc->seekable;
 	}
 
 	if (!can_seek) {
@@ -533,11 +532,11 @@ wavpack_streamdecode(struct decoder * decoder, struct input_stream *is)
 		return;
 	}
 
-	wavpack_decode(decoder, wpc, can_seek, NULL);
+	wavpack_decode(decoder, wpc, can_seek);
 
 	WavpackCloseFile(wpc);
 	if (open_flags & OPEN_WVC) {
-		input_stream_close(&is_wvc);
+		input_stream_close(is_wvc);
 	}
 }
 
@@ -549,11 +548,10 @@ wavpack_filedecode(struct decoder *decoder, const char *fname)
 {
 	char error[ERRORLEN];
 	WavpackContext *wpc;
-	struct replay_gain_info *replay_gain_info;
 
 	wpc = WavpackOpenFileInput(
 		fname, error,
-		OPEN_TAGS | OPEN_WVC | OPEN_2CH_MAX | OPEN_NORMALIZE, 23
+		OPEN_TAGS | OPEN_WVC | OPEN_NORMALIZE, 23
 	);
 	if (wpc == NULL) {
 		g_warning(
@@ -563,13 +561,11 @@ wavpack_filedecode(struct decoder *decoder, const char *fname)
 		return;
 	}
 
-	replay_gain_info = wavpack_replaygain(wpc);
+	struct replay_gain_info replay_gain_info;
+	if (wavpack_replaygain(&replay_gain_info, wpc))
+		decoder_replay_gain(decoder, &replay_gain_info);
 
-	wavpack_decode(decoder, wpc, true, replay_gain_info);
-
-	if (replay_gain_info) {
-		replay_gain_info_free(replay_gain_info);
-	}
+	wavpack_decode(decoder, wpc, true);
 
 	WavpackCloseFile(wpc);
 }
diff --git a/src/decoder/wildmidi_plugin.c b/src/decoder/wildmidi_decoder_plugin.c
index b5e9810f9..66e6c61cf 100644
--- a/src/decoder/wildmidi_plugin.c
+++ b/src/decoder/wildmidi_decoder_plugin.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,7 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
+#include "config.h"
+#include "decoder_api.h"
 
 #include <glib.h>
 
@@ -58,7 +59,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
 {
 	static const struct audio_format audio_format = {
 		.sample_rate = WILDMIDI_SAMPLE_RATE,
-		.bits = 16,
+		.format = SAMPLE_FORMAT_S16,
 		.channels = 2,
 	};
 	midi *wm;
@@ -90,10 +91,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
 		if (len <= 0)
 			break;
 
-		cmd = decoder_data(decoder, NULL, buffer, len,
-				   (float)info->current_sample /
-				   (float)WILDMIDI_SAMPLE_RATE,
-				   0, NULL);
+		cmd = decoder_data(decoder, NULL, buffer, len, 0);
 
 		if (cmd == DECODE_COMMAND_SEEK) {
 			unsigned long seek_where = WILDMIDI_SAMPLE_RATE *
@@ -116,21 +114,17 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
 static struct tag *
 wildmidi_tag_dup(const char *path_fs)
 {
-	midi *wm;
-	const struct _WM_Info *info;
-	struct tag *tag;
-
-	wm = WildMidi_Open(path_fs);
+	midi *wm = WildMidi_Open(path_fs);
 	if (wm == NULL)
 		return NULL;
 
-	info = WildMidi_GetInfo(wm);
+	const struct _WM_Info *info = WildMidi_GetInfo(wm);
 	if (info == NULL) {
 		WildMidi_Close(wm);
 		return NULL;
 	}
 
-	tag = tag_new();
+	struct tag *tag = tag_new();
 	tag->time = info->approx_total_samples / WILDMIDI_SAMPLE_RATE;
 
 	WildMidi_Close(wm);