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mpd/src/decoder/mad_plugin.c
Max Kellermann d8db46edfa decoders: added and fixed GLib log domains 
Fixed the log domains of the renamed decoders.  Added G_LOG_DOMAIN
macros in decoders which don't have one already.
2009-02-17 08:51:34 +01:00

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/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../decoder_api.h"
#include "../conf.h"
#include "config.h"
#include "tag_id3.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_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_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))
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 (strcasecmp(key, "replaygain_track_gain") == 0) {
replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = atof(value);
found = true;
} else if (strcasecmp(key, "replaygain_album_gain") == 0) {
replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = atof(value);
found = true;
} else if (strcasecmp(key, "replaygain_track_peak") == 0) {
replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = atof(value);
found = true;
} else if (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
#ifdef HAVE_ID3TAG
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 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 (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;
}
}
id3_tag_delete(id3_tag);
g_free(allocated);
}
#endif
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)
{
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)) {
#ifdef HAVE_ID3TAG
if ((data->stream).error == MAD_ERROR_LOSTSYNC &&
(data->stream).this_frame) {
signed long tagsize = id3_tag_query((data->stream).
this_frame,
(data->stream).
bufend -
(data->stream).
this_frame);
if (tagsize > 0) {
if (tag && !(*tag)) {
mp3_parse_id3(data, (size_t)tagsize,
tag, replay_gain_info_r);
} else {
mad_stream_skip(&(data->stream),
tagsize);
}
return DECODE_CONT;
}
}
#endif
if (MAD_RECOVERABLE((data->stream).error)) {
return DECODE_SKIP;
} else {
if ((data->stream).error == MAD_ERROR_BUFLEN)
return DECODE_CONT;
else {
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)) {
#ifdef HAVE_ID3TAG
if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
signed long tagsize = id3_tag_query((data->stream).
this_frame,
(data->stream).
bufend -
(data->stream).
this_frame);
if (tagsize > 0) {
mad_stream_skip(&(data->stream), tagsize);
return DECODE_CONT;
}
}
#endif
if (MAD_RECOVERABLE((data->stream).error)) {
return DECODE_SKIP;
} else {
if ((data->stream).error == MAD_ERROR_BUFLEN)
return DECODE_CONT;
else {
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 off_t
mp3_this_frame_offset(const struct mp3_data *data)
{
off_t 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 off_t
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)
{
off_t 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 replay_gain_info **replay_gain_info_r)
{
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,
replay_gain_info_r);
} 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 (replay_gain_info_r && !*replay_gain_info_r &&
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;
}
}
}
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 mp3_total_file_time(const char *file)
{
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))
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)
{
mp3_data_init(data, decoder, is);
*tag = NULL;
if (!mp3_decode_first_frame(data, tag, replay_gain_info_r)) {
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,
struct replay_gain_info *replay_gain_info)
{
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->elapsed_time,
data->bit_rate / 1000,
replay_gain_info);
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,
struct replay_gain_info *replay_gain_info)
{
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, replay_gain_info);
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 replay_gain_info **replay_gain_info_r)
{
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,
replay_gain_info_r != NULL
? *replay_gain_info_r : NULL);
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,
replay_gain_info_r);
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_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;
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 (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);
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, &replay_gain_info)) ;
if (replay_gain_info)
replay_gain_info_free(replay_gain_info);
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK &&
data.mute_frame == MUTEFRAME_SEEK)
decoder_command_finished(decoder);
mp3_data_finish(&data);
}
static struct tag *mp3_tag_dup(const char *file)
{
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);
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,
.tag_dup = mp3_tag_dup,
.suffixes = mp3_suffixes,
.mime_types = mp3_mime_types
};
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