renamed decoder plugin sources

Make it X_decoder_plugin.c.

1 files changed, 1235 insertions, 0 deletions

diff --git a/src/decoder/mad_decoder_plugin.c b/src/decoder/mad_decoder_plugin.c
new file mode 100644
index 00000000..87dfbeab
--- /dev/null
+++ b/src/decoder/mad_decoder_plugin.c
@@ -0,0 +1,1235 @@
+/*
+ * 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 "conf.h"
+#include "tag_id3.h"
+#include "audio_check.h"
+
+#include <assert.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <glib.h>
+#include <mad.h>
+
+#ifdef HAVE_ID3TAG
+#include <id3tag.h>
+#endif
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "mad"
+
+#define FRAMES_CUSHION    2000
+
+#define READ_BUFFER_SIZE  40960
+
+enum mp3_action {
+	DECODE_SKIP = -3,
+	DECODE_BREAK = -2,
+	DECODE_CONT = -1,
+	DECODE_OK = 0
+};
+
+enum muteframe {
+	MUTEFRAME_NONE,
+	MUTEFRAME_SKIP,
+	MUTEFRAME_SEEK
+};
+
+/* the number of samples of silence the decoder inserts at start */
+#define DECODERDELAY 529
+
+#define DEFAULT_GAPLESS_MP3_PLAYBACK true
+
+static bool gapless_playback;
+
+static inline int32_t
+mad_fixed_to_24_sample(mad_fixed_t sample)
+{
+	enum {
+		bits = 24,
+		MIN = -MAD_F_ONE,
+		MAX = MAD_F_ONE - 1
+	};
+
+	/* round */
+	sample = sample + (1L << (MAD_F_FRACBITS - bits));
+
+	/* clip */
+	if (sample > MAX)
+		sample = MAX;
+	else if (sample < MIN)
+		sample = MIN;
+
+	/* quantize */
+	return sample >> (MAD_F_FRACBITS + 1 - bits);
+}
+
+static void
+mad_fixed_to_24_buffer(int32_t *dest, const struct mad_synth *synth,
+		       unsigned int start, unsigned int end,
+		       unsigned int num_channels)
+{
+	unsigned int i, c;
+
+	for (i = start; i < end; ++i) {
+		for (c = 0; c < num_channels; ++c)
+			*dest++ = mad_fixed_to_24_sample(synth->pcm.samples[c][i]);
+	}
+}
+
+static bool
+mp3_plugin_init(G_GNUC_UNUSED const struct config_param *param)
+{
+	gapless_playback = config_get_bool(CONF_GAPLESS_MP3_PLAYBACK,
+					   DEFAULT_GAPLESS_MP3_PLAYBACK);
+	return true;
+}
+
+#define MP3_DATA_OUTPUT_BUFFER_SIZE 2048
+
+struct mp3_data {
+	struct mad_stream stream;
+	struct mad_frame frame;
+	struct mad_synth synth;
+	mad_timer_t timer;
+	unsigned char input_buffer[READ_BUFFER_SIZE];
+	int32_t output_buffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
+	float total_time;
+	float elapsed_time;
+	float seek_where;
+	enum muteframe mute_frame;
+	long *frame_offsets;
+	mad_timer_t *times;
+	unsigned long highest_frame;
+	unsigned long max_frames;
+	unsigned long current_frame;
+	unsigned int drop_start_frames;
+	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;
+	unsigned long bit_rate;
+	struct decoder *decoder;
+	struct input_stream *input_stream;
+	enum mad_layer layer;
+};
+
+static void
+mp3_data_init(struct mp3_data *data, struct decoder *decoder,
+	      struct input_stream *input_stream)
+{
+	data->mute_frame = MUTEFRAME_NONE;
+	data->highest_frame = 0;
+	data->max_frames = 0;
+	data->frame_offsets = NULL;
+	data->times = NULL;
+	data->current_frame = 0;
+	data->drop_start_frames = 0;
+	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;
+	data->decoder = decoder;
+	data->input_stream = input_stream;
+	data->layer = 0;
+
+	mad_stream_init(&data->stream);
+	mad_stream_options(&data->stream, MAD_OPTION_IGNORECRC);
+	mad_frame_init(&data->frame);
+	mad_synth_init(&data->synth);
+	mad_timer_reset(&data->timer);
+}
+
+static bool mp3_seek(struct mp3_data *data, long offset)
+{
+	if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL))
+		return false;
+
+	mad_stream_buffer(&data->stream, data->input_buffer, 0);
+	(data->stream).error = 0;
+
+	return true;
+}
+
+static bool
+mp3_fill_buffer(struct mp3_data *data)
+{
+	size_t remaining, length;
+	unsigned char *dest;
+
+	if (data->stream.next_frame != NULL) {
+		remaining = data->stream.bufend - data->stream.next_frame;
+		memmove(data->input_buffer, data->stream.next_frame,
+			remaining);
+		dest = (data->input_buffer) + remaining;
+		length = READ_BUFFER_SIZE - remaining;
+	} else {
+		remaining = 0;
+		length = READ_BUFFER_SIZE;
+		dest = data->input_buffer;
+	}
+
+	/* we've exhausted the read buffer, so give up!, these potential
+	 * mp3 frames are way too big, and thus unlikely to be mp3 frames */
+	if (length == 0)
+		return false;
+
+	length = decoder_read(data->decoder, data->input_stream, dest, length);
+	if (length == 0)
+		return false;
+
+	mad_stream_buffer(&data->stream, data->input_buffer,
+			  length + remaining);
+	(data->stream).error = 0;
+
+	return true;
+}
+
+#ifdef HAVE_ID3TAG
+/* Parse mp3 RVA2 frame. Shamelessly stolen from madplay. */
+static int 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;
+	int found;
+
+	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
+	};
+
+	found = 0;
+
+	/* relative volume adjustment information */
+
+	frame = id3_tag_findframe(tag, "RVA2", 0);
+	if (!frame) return 0;
+
+	id   = id3_field_getlatin1(id3_frame_field(frame, 0));
+	data = id3_field_getbinarydata(id3_frame_field(frame, 1),
+					&length);
+
+	if (!id || !data) return 0;
+
+	/*
+	 * "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].peak = voladj_float;
+			replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = voladj_float;
+
+			g_debug("parseRVA2: Relative Volume "
+				"%+.1f dB adjustment (%s)\n",
+				voladj_float, id);
+
+			found = 1;
+			break;
+		}
+
+		data   += 4 + peak_bytes;
+		length -= 4 + peak_bytes;
+	}
+
+	return found;
+}
+#endif
+
+#ifdef HAVE_ID3TAG
+static struct replay_gain_info *
+parse_id3_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();
+
+	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, "replaygain_track_gain") == 0) {
+			replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = atof(value);
+			found = true;
+		} else if (g_ascii_strcasecmp(key, "replaygain_album_gain") == 0) {
+			replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = atof(value);
+			found = true;
+		} else if (g_ascii_strcasecmp(key, "replaygain_track_peak") == 0) {
+			replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = atof(value);
+			found = true;
+		} else if (g_ascii_strcasecmp(key, "replaygain_album_peak") == 0) {
+			replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = atof(value);
+			found = true;
+		}
+
+		free(key);
+		free(value);
+	}
+
+	if (!found) {
+		/* fall back on RVA2 if no replaygain tags found */
+		found = parse_rva2(tag, replay_gain_info);
+	}
+
+	if (found)
+		return replay_gain_info;
+	replay_gain_info_free(replay_gain_info);
+	return NULL;
+}
+#endif
+
+static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
+			  struct tag **mpd_tag)
+{
+#ifdef HAVE_ID3TAG
+	struct id3_tag *id3_tag = NULL;
+	id3_length_t count;
+	id3_byte_t const *id3_data;
+	id3_byte_t *allocated = NULL;
+
+	count = data->stream.bufend - data->stream.this_frame;
+
+	if (tagsize <= count) {
+		id3_data = data->stream.this_frame;
+		mad_stream_skip(&(data->stream), tagsize);
+	} else {
+		allocated = g_malloc(tagsize);
+		memcpy(allocated, data->stream.this_frame, count);
+		mad_stream_skip(&(data->stream), count);
+
+		while (count < tagsize) {
+			size_t len;
+
+			len = decoder_read(data->decoder, data->input_stream,
+					   allocated + count, tagsize - count);
+			if (len == 0)
+				break;
+			else
+				count += len;
+		}
+
+		if (count != tagsize) {
+			g_debug("error parsing ID3 tag");
+			g_free(allocated);
+			return;
+		}
+
+		id3_data = allocated;
+	}
+
+	id3_tag = id3_tag_parse(id3_data, tagsize);
+	if (id3_tag == NULL) {
+		g_free(allocated);
+		return;
+	}
+
+	if (mpd_tag) {
+		struct tag *tmp_tag = tag_id3_import(id3_tag);
+		if (tmp_tag != NULL) {
+			if (*mpd_tag != NULL)
+				tag_free(*mpd_tag);
+			*mpd_tag = tmp_tag;
+		}
+	}
+
+	if (data->decoder != NULL) {
+		struct replay_gain_info *tmp_rgi =
+			parse_id3_replay_gain_info(id3_tag);
+		if (tmp_rgi != NULL) {
+			decoder_replay_gain(data->decoder, tmp_rgi);
+			replay_gain_info_free(tmp_rgi);
+			data->found_replay_gain = true;
+		}
+	}
+
+	id3_tag_delete(id3_tag);
+
+	g_free(allocated);
+#else /* !HAVE_ID3TAG */
+	(void)mpd_tag;
+
+	/* This code is enabled when libid3tag is disabled.  Instead
+	   of parsing the ID3 frame, it just skips it. */
+
+	mad_stream_skip(&data->stream, tagsize);
+#endif
+}
+
+#ifndef HAVE_ID3TAG
+/**
+ * This function emulates libid3tag when it is disabled.  Instead of
+ * doing a real analyzation of the frame, it just checks whether the
+ * frame begins with the string "ID3".  If so, it returns the full
+ * length.
+ */
+static signed long
+id3_tag_query(const void *p0, size_t length)
+{
+	const char *p = p0;
+
+	return length > 3 && memcmp(p, "ID3", 3) == 0
+		? length
+		: 0;
+}
+#endif /* !HAVE_ID3TAG */
+
+static enum mp3_action
+decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag)
+{
+	enum mad_layer layer;
+
+	if ((data->stream).buffer == NULL
+	    || (data->stream).error == MAD_ERROR_BUFLEN) {
+		if (!mp3_fill_buffer(data))
+			return DECODE_BREAK;
+	}
+	if (mad_header_decode(&data->frame.header, &data->stream)) {
+		if ((data->stream).error == MAD_ERROR_LOSTSYNC &&
+		    (data->stream).this_frame) {
+			signed long tagsize = id3_tag_query((data->stream).
+							    this_frame,
+							    (data->stream).
+							    bufend -
+							    (data->stream).
+							    this_frame);
+
+			if (tagsize > 0) {
+				if (tag && !(*tag)) {
+					mp3_parse_id3(data, (size_t)tagsize,
+						      tag);
+				} else {
+					mad_stream_skip(&(data->stream),
+							tagsize);
+				}
+				return DECODE_CONT;
+			}
+		}
+		if (MAD_RECOVERABLE((data->stream).error)) {
+			return DECODE_SKIP;
+		} else {
+			if ((data->stream).error == MAD_ERROR_BUFLEN)
+				return DECODE_CONT;
+			else {
+				g_warning("unrecoverable frame level error "
+					  "(%s).\n",
+					  mad_stream_errorstr(&data->stream));
+				return DECODE_BREAK;
+			}
+		}
+	}
+
+	layer = data->frame.header.layer;
+	if (!data->layer) {
+		if (layer != MAD_LAYER_II && layer != MAD_LAYER_III) {
+			/* Only layer 2 and 3 have been tested to work */
+			return DECODE_SKIP;
+		}
+		data->layer = layer;
+	} else if (layer != data->layer) {
+		/* Don't decode frames with a different layer than the first */
+		return DECODE_SKIP;
+	}
+
+	return DECODE_OK;
+}
+
+static enum mp3_action
+decodeNextFrame(struct mp3_data *data)
+{
+	if ((data->stream).buffer == NULL
+	    || (data->stream).error == MAD_ERROR_BUFLEN) {
+		if (!mp3_fill_buffer(data))
+			return DECODE_BREAK;
+	}
+	if (mad_frame_decode(&data->frame, &data->stream)) {
+		if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
+			signed long tagsize = id3_tag_query((data->stream).
+							    this_frame,
+							    (data->stream).
+							    bufend -
+							    (data->stream).
+							    this_frame);
+			if (tagsize > 0) {
+				mad_stream_skip(&(data->stream), tagsize);
+				return DECODE_CONT;
+			}
+		}
+		if (MAD_RECOVERABLE((data->stream).error)) {
+			return DECODE_SKIP;
+		} else {
+			if ((data->stream).error == MAD_ERROR_BUFLEN)
+				return DECODE_CONT;
+			else {
+				g_warning("unrecoverable frame level error "
+					  "(%s).\n",
+					  mad_stream_errorstr(&data->stream));
+				return DECODE_BREAK;
+			}
+		}
+	}
+
+	return DECODE_OK;
+}
+
+/* xing stuff stolen from alsaplayer, and heavily modified by jat */
+#define XI_MAGIC (('X' << 8) | 'i')
+#define NG_MAGIC (('n' << 8) | 'g')
+#define IN_MAGIC (('I' << 8) | 'n')
+#define FO_MAGIC (('f' << 8) | 'o')
+
+enum xing_magic {
+	XING_MAGIC_XING, /* VBR */
+	XING_MAGIC_INFO  /* CBR */
+};
+
+struct xing {
+	long flags;             /* valid fields (see below) */
+	unsigned long frames;   /* total number of frames */
+	unsigned long bytes;    /* total number of bytes */
+	unsigned char toc[100]; /* 100-point seek table */
+	long scale;             /* VBR quality */
+	enum xing_magic magic;  /* header magic */
+};
+
+enum {
+	XING_FRAMES = 0x00000001L,
+	XING_BYTES  = 0x00000002L,
+	XING_TOC    = 0x00000004L,
+	XING_SCALE  = 0x00000008L
+};
+
+struct version {
+	unsigned major;
+	unsigned minor;
+};
+
+struct lame {
+	char encoder[10];       /* 9 byte encoder name/version ("LAME3.97b") */
+	struct version version; /* struct containing just the version */
+	float peak;             /* replaygain peak */
+	float track_gain;       /* replaygain track gain */
+	float album_gain;       /* replaygain album gain */
+	int encoder_delay;      /* # of added samples at start of mp3 */
+	int encoder_padding;    /* # of added samples at end of mp3 */
+	int crc;                /* CRC of the first 190 bytes of this frame */
+};
+
+static bool
+parse_xing(struct xing *xing, struct mad_bitptr *ptr, int *oldbitlen)
+{
+	unsigned long bits;
+	int bitlen;
+	int bitsleft;
+	int i;
+
+	bitlen = *oldbitlen;
+
+	if (bitlen < 16)
+		return false;
+
+	bits = mad_bit_read(ptr, 16);
+	bitlen -= 16;
+
+	if (bits == XI_MAGIC) {
+		if (bitlen < 16)
+			return false;
+
+		if (mad_bit_read(ptr, 16) != NG_MAGIC)
+			return false;
+
+		bitlen -= 16;
+		xing->magic = XING_MAGIC_XING;
+	} else if (bits == IN_MAGIC) {
+		if (bitlen < 16)
+			return false;
+
+		if (mad_bit_read(ptr, 16) != FO_MAGIC)
+			return false;
+
+		bitlen -= 16;
+		xing->magic = XING_MAGIC_INFO;
+	}
+	else if (bits == NG_MAGIC) xing->magic = XING_MAGIC_XING;
+	else if (bits == FO_MAGIC) xing->magic = XING_MAGIC_INFO;
+	else
+		return false;
+
+	if (bitlen < 32)
+		return false;
+	xing->flags = mad_bit_read(ptr, 32);
+	bitlen -= 32;
+
+	if (xing->flags & XING_FRAMES) {
+		if (bitlen < 32)
+			return false;
+		xing->frames = mad_bit_read(ptr, 32);
+		bitlen -= 32;
+	}
+
+	if (xing->flags & XING_BYTES) {
+		if (bitlen < 32)
+			return false;
+		xing->bytes = mad_bit_read(ptr, 32);
+		bitlen -= 32;
+	}
+
+	if (xing->flags & XING_TOC) {
+		if (bitlen < 800)
+			return false;
+		for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(ptr, 8);
+		bitlen -= 800;
+	}
+
+	if (xing->flags & XING_SCALE) {
+		if (bitlen < 32)
+			return false;
+		xing->scale = mad_bit_read(ptr, 32);
+		bitlen -= 32;
+	}
+
+	/* Make sure we consume no less than 120 bytes (960 bits) in hopes that
+	 * the LAME tag is found there, and not right after the Xing header */
+	bitsleft = 960 - ((*oldbitlen) - bitlen);
+	if (bitsleft < 0)
+		return false;
+	else if (bitsleft > 0) {
+		mad_bit_read(ptr, bitsleft);
+		bitlen -= bitsleft;
+	}
+
+	*oldbitlen = bitlen;
+
+	return true;
+}
+
+static bool
+parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen)
+{
+	int adj = 0;
+	int name;
+	int orig;
+	int sign;
+	int gain;
+	int i;
+
+	/* Unlike the xing header, the lame tag has a fixed length.  Fail if
+	 * not all 36 bytes (288 bits) are there. */
+	if (*bitlen < 288)
+		return false;
+
+	for (i = 0; i < 9; i++)
+		lame->encoder[i] = (char)mad_bit_read(ptr, 8);
+	lame->encoder[9] = '\0';
+
+	*bitlen -= 72;
+
+	/* This is technically incorrect, since the encoder might not be lame.
+	 * But there's no other way to determine if this is a lame tag, and we
+	 * wouldn't want to go reading a tag that's not there. */
+	if (!g_str_has_prefix(lame->encoder, "LAME"))
+		return false;
+
+	if (sscanf(lame->encoder+4, "%u.%u",
+	           &lame->version.major, &lame->version.minor) != 2)
+		return false;
+
+	g_debug("detected LAME version %i.%i (\"%s\")\n",
+		lame->version.major, lame->version.minor, lame->encoder);
+
+	/* The reference volume was changed from the 83dB used in the
+	 * ReplayGain spec to 89dB in lame 3.95.1.  Bump the gain for older
+	 * versions, since everyone else uses 89dB instead of 83dB.
+	 * Unfortunately, lame didn't differentiate between 3.95 and 3.95.1, so
+	 * it's impossible to make the proper adjustment for 3.95.
+	 * Fortunately, 3.95 was only out for about a day before 3.95.1 was
+	 * released. -- tmz */
+	if (lame->version.major < 3 ||
+	    (lame->version.major == 3 && lame->version.minor < 95))
+		adj = 6;
+
+	mad_bit_read(ptr, 16);
+
+	lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */
+	g_debug("LAME peak found: %f\n", lame->peak);
+
+	lame->track_gain = 0;
+	name = mad_bit_read(ptr, 3); /* gain name */
+	orig = mad_bit_read(ptr, 3); /* gain originator */
+	sign = mad_bit_read(ptr, 1); /* sign bit */
+	gain = mad_bit_read(ptr, 9); /* gain*10 */
+	if (gain && name == 1 && orig != 0) {
+		lame->track_gain = ((sign ? -gain : gain) / 10.0) + adj;
+		g_debug("LAME track gain found: %f\n", lame->track_gain);
+	}
+
+	/* tmz reports that this isn't currently written by any version of lame
+	 * (as of 3.97).  Since we have no way of testing it, don't use it.
+	 * Wouldn't want to go blowing someone's ears just because we read it
+	 * wrong. :P -- jat */
+	lame->album_gain = 0;
+#if 0
+	name = mad_bit_read(ptr, 3); /* gain name */
+	orig = mad_bit_read(ptr, 3); /* gain originator */
+	sign = mad_bit_read(ptr, 1); /* sign bit */
+	gain = mad_bit_read(ptr, 9); /* gain*10 */
+	if (gain && name == 2 && orig != 0) {
+		lame->album_gain = ((sign ? -gain : gain) / 10.0) + adj;
+		g_debug("LAME album gain found: %f\n", lame->track_gain);
+	}
+#else
+	mad_bit_read(ptr, 16);
+#endif
+
+	mad_bit_read(ptr, 16);
+
+	lame->encoder_delay = mad_bit_read(ptr, 12);
+	lame->encoder_padding = mad_bit_read(ptr, 12);
+
+	g_debug("encoder delay is %i, encoder padding is %i\n",
+	      lame->encoder_delay, lame->encoder_padding);
+
+	mad_bit_read(ptr, 80);
+
+	lame->crc = mad_bit_read(ptr, 16);
+
+	*bitlen -= 216;
+
+	return true;
+}
+
+static inline float
+mp3_frame_duration(const struct mad_frame *frame)
+{
+	return mad_timer_count(frame->header.duration,
+			       MAD_UNITS_MILLISECONDS) / 1000.0;
+}
+
+static goffset
+mp3_this_frame_offset(const struct mp3_data *data)
+{
+	goffset offset = data->input_stream->offset;
+
+	if (data->stream.this_frame != NULL)
+		offset -= data->stream.bufend - data->stream.this_frame;
+	else
+		offset -= data->stream.bufend - data->stream.buffer;
+
+	return offset;
+}
+
+static goffset
+mp3_rest_including_this_frame(const struct mp3_data *data)
+{
+	return data->input_stream->size - mp3_this_frame_offset(data);
+}
+
+/**
+ * Attempt to calulcate the length of the song from filesize
+ */
+static void
+mp3_filesize_to_song_length(struct mp3_data *data)
+{
+	goffset rest = mp3_rest_including_this_frame(data);
+
+	if (rest > 0) {
+		float frame_duration = mp3_frame_duration(&data->frame);
+
+		data->total_time = (rest * 8.0) / (data->frame).header.bitrate;
+		data->max_frames = data->total_time / frame_duration +
+			FRAMES_CUSHION;
+	} else {
+		data->max_frames = FRAMES_CUSHION;
+		data->total_time = 0;
+	}
+}
+
+static bool
+mp3_decode_first_frame(struct mp3_data *data, struct tag **tag)
+{
+	struct xing xing;
+	struct lame lame;
+	struct mad_bitptr ptr;
+	int bitlen;
+	enum mp3_action ret;
+
+	/* stfu gcc */
+	memset(&xing, 0, sizeof(struct xing));
+	xing.flags = 0;
+
+	while (true) {
+		do {
+			ret = decode_next_frame_header(data, tag);
+		} while (ret == DECODE_CONT);
+		if (ret == DECODE_BREAK)
+			return false;
+		if (ret == DECODE_SKIP) continue;
+
+		do {
+			ret = decodeNextFrame(data);
+		} while (ret == DECODE_CONT);
+		if (ret == DECODE_BREAK)
+			return false;
+		if (ret == DECODE_OK) break;
+	}
+
+	ptr = data->stream.anc_ptr;
+	bitlen = data->stream.anc_bitlen;
+
+	mp3_filesize_to_song_length(data);
+
+	/*
+	 * if an xing tag exists, use that!
+	 */
+	if (parse_xing(&xing, &ptr, &bitlen)) {
+		data->found_xing = true;
+		data->mute_frame = MUTEFRAME_SKIP;
+
+		if ((xing.flags & XING_FRAMES) && xing.frames) {
+			mad_timer_t duration = data->frame.header.duration;
+			mad_timer_multiply(&duration, xing.frames);
+			data->total_time = ((float)mad_timer_count(duration, MAD_UNITS_MILLISECONDS)) / 1000;
+			data->max_frames = xing.frames;
+		}
+
+		if (parse_lame(&lame, &ptr, &bitlen)) {
+			if (gapless_playback &&
+			    data->input_stream->seekable) {
+				data->drop_start_samples = lame.encoder_delay +
+				                           DECODERDELAY;
+				data->drop_end_samples = lame.encoder_padding;
+			}
+
+			/* Album gain isn't currently used.  See comment in
+			 * parse_lame() for details. -- jat */
+			if (data->decoder != NULL &&
+			    !data->found_replay_gain &&
+			    lame.track_gain) {
+				struct replay_gain_info *rgi
+					= replay_gain_info_new();
+				rgi->tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain;
+				rgi->tuples[REPLAY_GAIN_TRACK].peak = lame.peak;
+				decoder_replay_gain(data->decoder, rgi);
+				replay_gain_info_free(rgi);
+			}
+		}
+	} 
+
+	if (!data->max_frames)
+		return false;
+
+	if (data->max_frames > 8 * 1024 * 1024) {
+		g_warning("mp3 file header indicates too many frames: %lu\n",
+			  data->max_frames);
+		return false;
+	}
+
+	data->frame_offsets = g_malloc(sizeof(long) * data->max_frames);
+	data->times = g_malloc(sizeof(mad_timer_t) * data->max_frames);
+
+	return true;
+}
+
+static void mp3_data_finish(struct mp3_data *data)
+{
+	mad_synth_finish(&data->synth);
+	mad_frame_finish(&data->frame);
+	mad_stream_finish(&data->stream);
+
+	g_free(data->frame_offsets);
+	g_free(data->times);
+}
+
+/* this is primarily used for getting total time for tags */
+static int
+mad_decoder_total_file_time(struct input_stream *is)
+{
+	struct mp3_data data;
+	int ret;
+
+	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);
+
+	return ret;
+}
+
+static bool
+mp3_open(struct input_stream *is, struct mp3_data *data,
+	 struct decoder *decoder, struct tag **tag)
+{
+	mp3_data_init(data, decoder, is);
+	*tag = NULL;
+	if (!mp3_decode_first_frame(data, tag)) {
+		mp3_data_finish(data);
+		if (tag && *tag)
+			tag_free(*tag);
+		return false;
+	}
+
+	return true;
+}
+
+static long
+mp3_time_to_frame(const struct mp3_data *data, double t)
+{
+	unsigned long i;
+
+	for (i = 0; i < data->highest_frame; ++i) {
+		double frame_time =
+			mad_timer_count(data->times[i],
+					MAD_UNITS_MILLISECONDS) / 1000.;
+		if (frame_time >= t)
+			break;
+	}
+
+	return i;
+}
+
+static void
+mp3_update_timer_next_frame(struct mp3_data *data)
+{
+	if (data->current_frame >= data->highest_frame) {
+		/* record this frame's properties in
+		   data->frame_offsets (for seeking) and
+		   data->times */
+		data->bit_rate = (data->frame).header.bitrate;
+
+		if (data->current_frame >= data->max_frames)
+			/* cap data->current_frame */
+			data->current_frame = data->max_frames - 1;
+		else
+			data->highest_frame++;
+
+		data->frame_offsets[data->current_frame] =
+			mp3_this_frame_offset(data);
+
+		mad_timer_add(&data->timer, (data->frame).header.duration);
+		data->times[data->current_frame] = data->timer;
+	} else
+		/* get the new timer value from data->times */
+		data->timer = data->times[data->current_frame];
+
+	data->current_frame++;
+	data->elapsed_time =
+		mad_timer_count(data->timer, MAD_UNITS_MILLISECONDS) / 1000.0;
+}
+
+/**
+ * Sends the synthesized current frame via decoder_data().
+ */
+static enum decoder_command
+mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length)
+{
+	unsigned max_samples;
+
+	max_samples = sizeof(data->output_buffer) /
+		sizeof(data->output_buffer[0]) /
+		MAD_NCHANNELS(&(data->frame).header);
+
+	while (i < pcm_length) {
+		enum decoder_command cmd;
+		unsigned int num_samples = pcm_length - i;
+		if (num_samples > max_samples)
+			num_samples = max_samples;
+
+		i += num_samples;
+
+		mad_fixed_to_24_buffer(data->output_buffer,
+				       &data->synth,
+				       i - num_samples, i,
+				       MAD_NCHANNELS(&(data->frame).header));
+		num_samples *= MAD_NCHANNELS(&(data->frame).header);
+
+		cmd = decoder_data(data->decoder, data->input_stream,
+				   data->output_buffer,
+				   sizeof(data->output_buffer[0]) * num_samples,
+				   data->bit_rate / 1000);
+		if (cmd != DECODE_COMMAND_NONE)
+			return cmd;
+	}
+
+	return DECODE_COMMAND_NONE;
+}
+
+/**
+ * Synthesize the current frame and send it via decoder_data().
+ */
+static enum decoder_command
+mp3_synth_and_send(struct mp3_data *data)
+{
+	unsigned i, pcm_length;
+	enum decoder_command cmd;
+
+	mad_synth_frame(&data->synth, &data->frame);
+
+	if (!data->found_first_frame) {
+		unsigned int samples_per_frame = data->synth.pcm.length;
+		data->drop_start_frames = data->drop_start_samples / samples_per_frame;
+		data->drop_end_frames = data->drop_end_samples / samples_per_frame;
+		data->drop_start_samples = data->drop_start_samples % samples_per_frame;
+		data->drop_end_samples = data->drop_end_samples % samples_per_frame;
+		data->found_first_frame = true;
+	}
+
+	if (data->drop_start_frames > 0) {
+		data->drop_start_frames--;
+		return DECODE_COMMAND_NONE;
+	} else if ((data->drop_end_frames > 0) &&
+		   (data->current_frame == (data->max_frames + 1 - data->drop_end_frames))) {
+		/* stop decoding, effectively dropping all remaining
+		   frames */
+		return DECODE_COMMAND_STOP;
+	}
+
+	if (!data->decoded_first_frame) {
+		i = data->drop_start_samples;
+		data->decoded_first_frame = true;
+	} else
+		i = 0;
+
+	pcm_length = data->synth.pcm.length;
+	if (data->drop_end_samples &&
+	    (data->current_frame == data->max_frames - data->drop_end_frames)) {
+		if (data->drop_end_samples >= pcm_length)
+			pcm_length = 0;
+		else
+			pcm_length -= data->drop_end_samples;
+	}
+
+	cmd = mp3_send_pcm(data, i, pcm_length);
+	if (cmd != DECODE_COMMAND_NONE)
+		return cmd;
+
+	if (data->drop_end_samples &&
+	    (data->current_frame == data->max_frames - data->drop_end_frames))
+		/* stop decoding, effectively dropping
+		 * all remaining samples */
+		return DECODE_COMMAND_STOP;
+
+	return DECODE_COMMAND_NONE;
+}
+
+static bool
+mp3_read(struct mp3_data *data)
+{
+	struct decoder *decoder = data->decoder;
+	enum mp3_action ret;
+	enum decoder_command cmd;
+
+	mp3_update_timer_next_frame(data);
+
+	switch (data->mute_frame) {
+	case MUTEFRAME_SKIP:
+		data->mute_frame = MUTEFRAME_NONE;
+		break;
+	case MUTEFRAME_SEEK:
+		if (data->elapsed_time >= data->seek_where)
+			data->mute_frame = MUTEFRAME_NONE;
+		break;
+	case MUTEFRAME_NONE:
+		cmd = mp3_synth_and_send(data);
+		if (cmd == DECODE_COMMAND_SEEK) {
+			unsigned long j;
+
+			assert(data->input_stream->seekable);
+
+			j = mp3_time_to_frame(data,
+					      decoder_seek_where(decoder));
+			if (j < data->highest_frame) {
+				if (mp3_seek(data, data->frame_offsets[j])) {
+					data->current_frame = j;
+					decoder_command_finished(decoder);
+				} else
+					decoder_seek_error(decoder);
+			} else {
+				data->seek_where = decoder_seek_where(decoder);
+				data->mute_frame = MUTEFRAME_SEEK;
+				decoder_command_finished(decoder);
+			}
+		} else if (cmd != DECODE_COMMAND_NONE)
+			return false;
+	}
+
+	while (true) {
+		bool skip = false;
+
+		do {
+			struct tag *tag = NULL;
+
+			ret = decode_next_frame_header(data, &tag);
+
+			if (tag != NULL) {
+				decoder_tag(decoder, data->input_stream, tag);
+				tag_free(tag);
+			}
+		} while (ret == DECODE_CONT);
+		if (ret == DECODE_BREAK)
+			return false;
+		else if (ret == DECODE_SKIP)
+			skip = true;
+
+		if (data->mute_frame == MUTEFRAME_NONE) {
+			do {
+				ret = decodeNextFrame(data);
+			} while (ret == DECODE_CONT);
+			if (ret == DECODE_BREAK)
+				return false;
+		}
+
+		if (!skip && ret == DECODE_OK)
+			break;
+	}
+
+	return ret != DECODE_BREAK;
+}
+
+static void
+mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
+{
+	struct mp3_data data;
+	GError *error = NULL;
+	struct tag *tag = NULL;
+	struct audio_format audio_format;
+
+	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;
+	}
+
+	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);
+
+	if (tag != NULL) {
+		decoder_tag(decoder, input_stream, tag);
+		tag_free(tag);
+	}
+
+	while (mp3_read(&data)) ;
+
+	if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
+	    data.mute_frame == MUTEFRAME_SEEK)
+		decoder_command_finished(decoder);
+
+	mp3_data_finish(&data);
+}
+
+static struct tag *
+mad_decoder_stream_tag(struct input_stream *is)
+{
+	struct tag *tag;
+	int total_time;
+
+	total_time = mad_decoder_total_file_time(is);
+	if (total_time < 0)
+		return NULL;
+
+	tag = tag_new();
+	tag->time = total_time;
+	return tag;
+}
+
+static const char *const mp3_suffixes[] = { "mp3", "mp2", NULL };
+static const char *const mp3_mime_types[] = { "audio/mpeg", NULL };
+
+const struct decoder_plugin mad_decoder_plugin = {
+	.name = "mad",
+	.init = mp3_plugin_init,
+	.stream_decode = mp3_decode,
+	.stream_tag = mad_decoder_stream_tag,
+	.suffixes = mp3_suffixes,
+	.mime_types = mp3_mime_types
+};