mpd/src/inputPlugins/mp3_plugin.c

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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 "../inputPlugin.h"
#ifdef HAVE_MAD
#include "../pcm_utils.h"
#include <mad.h>
#ifdef HAVE_ID3TAG
#include <id3tag.h>
#endif
#include "../log.h"
#include "../utils.h"
#include "../replayGain.h"
#include "../tag.h"
#include "../conf.h"
#include "../os_compat.h"
#define FRAMES_CUSHION 2000
#define READ_BUFFER_SIZE 40960
#define DECODE_SKIP -3
#define DECODE_BREAK -2
#define DECODE_CONT -1
#define DECODE_OK 0
#define MUTEFRAME_SKIP 1
#define MUTEFRAME_SEEK 2
/* the number of samples of silence the decoder inserts at start */
#define DECODERDELAY 529
#define DEFAULT_GAPLESS_MP3_PLAYBACK 1
static int gaplessPlaybackEnabled;
/* this is stolen from mpg321! */
struct audio_dither {
mad_fixed_t error[3];
mad_fixed_t random;
};
static unsigned long prng(unsigned long state)
{
return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
}
static signed long audio_linear_dither(unsigned int bits, mad_fixed_t sample,
struct audio_dither *dither)
{
unsigned int scalebits;
mad_fixed_t output, mask, rnd;
enum {
MIN = -MAD_F_ONE,
MAX = MAD_F_ONE - 1
};
sample += dither->error[0] - dither->error[1] + dither->error[2];
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0] / 2;
output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1));
scalebits = MAD_F_FRACBITS + 1 - bits;
mask = (1L << scalebits) - 1;
rnd = prng(dither->random);
output += (rnd & mask) - (dither->random & mask);
dither->random = rnd;
if (output > MAX) {
output = MAX;
if (sample > MAX)
sample = MAX;
} else if (output < MIN) {
output = MIN;
if (sample < MIN)
sample = MIN;
}
output &= ~mask;
dither->error[0] = sample - output;
return output >> scalebits;
}
/* end of stolen stuff from mpg321 */
static int mp3_plugin_init(void)
{
gaplessPlaybackEnabled = getBoolConfigParam(CONF_GAPLESS_MP3_PLAYBACK,
1);
if (gaplessPlaybackEnabled == CONF_BOOL_UNSET)
gaplessPlaybackEnabled = DEFAULT_GAPLESS_MP3_PLAYBACK;
return 1;
}
/* decoder stuff is based on madlld */
#define MP3_DATA_OUTPUT_BUFFER_SIZE 4096
typedef struct _mp3DecodeData {
struct mad_stream stream;
struct mad_frame frame;
struct mad_synth synth;
mad_timer_t timer;
unsigned char readBuffer[READ_BUFFER_SIZE];
char outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
char *outputPtr;
char *outputBufferEnd;
float totalTime;
float elapsedTime;
int muteFrame;
long *frameOffset;
mad_timer_t *times;
long highestFrame;
long maxFrames;
long currentFrame;
int dropFramesAtStart;
int dropFramesAtEnd;
int dropSamplesAtStart;
int dropSamplesAtEnd;
int foundXing;
int foundFirstFrame;
int decodedFirstFrame;
int flush;
unsigned long bitRate;
InputStream *inStream;
struct audio_dither dither;
enum mad_layer layer;
} mp3DecodeData;
static void initMp3DecodeData(mp3DecodeData * data, InputStream * inStream)
{
data->outputPtr = data->outputBuffer;
data->outputBufferEnd =
data->outputBuffer + MP3_DATA_OUTPUT_BUFFER_SIZE;
data->muteFrame = 0;
data->highestFrame = 0;
data->maxFrames = 0;
data->frameOffset = NULL;
data->times = NULL;
data->currentFrame = 0;
data->dropFramesAtStart = 0;
data->dropFramesAtEnd = 0;
data->dropSamplesAtStart = 0;
data->dropSamplesAtEnd = 0;
data->foundXing = 0;
data->foundFirstFrame = 0;
data->decodedFirstFrame = 0;
data->flush = 1;
data->inStream = inStream;
data->layer = 0;
memset(&(data->dither), 0, sizeof(struct audio_dither));
mad_stream_init(&data->stream);
mad_stream_options(&data->stream, MAD_OPTION_IGNORECRC);
mad_frame_init(&data->frame);
mad_synth_init(&data->synth);
mad_timer_reset(&data->timer);
}
static int seekMp3InputBuffer(mp3DecodeData * data, long offset)
{
if (seekInputStream(data->inStream, offset, SEEK_SET) < 0) {
return -1;
}
mad_stream_buffer(&data->stream, data->readBuffer, 0);
(data->stream).error = 0;
return 0;
}
static int fillMp3InputBuffer(mp3DecodeData * data)
{
size_t readSize;
size_t remaining;
size_t readed;
unsigned char *readStart;
if ((data->stream).next_frame != NULL) {
remaining = (data->stream).bufend - (data->stream).next_frame;
memmove(data->readBuffer, (data->stream).next_frame, remaining);
readStart = (data->readBuffer) + remaining;
readSize = READ_BUFFER_SIZE - remaining;
} else {
readSize = READ_BUFFER_SIZE;
readStart = data->readBuffer, remaining = 0;
}
/* we've exhausted the read buffer, so give up!, these potential
* mp3 frames are way too big, and thus unlikely to be mp3 frames */
if (readSize == 0)
return -1;
readed = readFromInputStream(data->inStream, readStart, (size_t) 1,
readSize);
if (readed <= 0 && inputStreamAtEOF(data->inStream))
return -1;
/* sleep for a fraction of a second! */
else if (readed <= 0) {
readed = 0;
my_usleep(10000);
}
mad_stream_buffer(&data->stream, data->readBuffer, readed + remaining);
(data->stream).error = 0;
return 0;
}
#ifdef HAVE_ID3TAG
static ReplayGainInfo *parseId3ReplayGainInfo(struct id3_tag *tag)
{
int i;
char *key;
char *value;
struct id3_frame *frame;
int found = 0;
ReplayGainInfo *replayGainInfo;
replayGainInfo = newReplayGainInfo();
for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) {
if (frame->nfields < 3)
continue;
key = (char *)
id3_ucs4_latin1duplicate(id3_field_getstring
(&frame->fields[1]));
value = (char *)
id3_ucs4_latin1duplicate(id3_field_getstring
(&frame->fields[2]));
if (strcasecmp(key, "replaygain_track_gain") == 0) {
replayGainInfo->trackGain = atof(value);
found = 1;
} else if (strcasecmp(key, "replaygain_album_gain") == 0) {
replayGainInfo->albumGain = atof(value);
found = 1;
} else if (strcasecmp(key, "replaygain_track_peak") == 0) {
replayGainInfo->trackPeak = atof(value);
found = 1;
} else if (strcasecmp(key, "replaygain_album_peak") == 0) {
replayGainInfo->albumPeak = atof(value);
found = 1;
}
free(key);
free(value);
}
if (found)
return replayGainInfo;
freeReplayGainInfo(replayGainInfo);
return NULL;
}
#endif
#ifdef HAVE_ID3TAG
static void mp3_parseId3Tag(mp3DecodeData * data, size_t tagsize,
MpdTag ** mpdTag, ReplayGainInfo ** replayGainInfo)
{
struct id3_tag *id3Tag = NULL;
id3_length_t count;
id3_byte_t const *id3_data;
id3_byte_t *allocated = NULL;
MpdTag *tmpMpdTag;
ReplayGainInfo *tmpReplayGainInfo;
count = data->stream.bufend - data->stream.this_frame;
if (tagsize <= count) {
id3_data = data->stream.this_frame;
mad_stream_skip(&(data->stream), tagsize);
} else {
allocated = xmalloc(tagsize);
if (!allocated)
goto fail;
memcpy(allocated, data->stream.this_frame, count);
mad_stream_skip(&(data->stream), count);
while (count < tagsize) {
size_t len;
len = readFromInputStream(data->inStream,
allocated + count, (size_t) 1,
tagsize - count);
if (len <= 0 && inputStreamAtEOF(data->inStream)) {
break;
} else if (len <= 0)
my_usleep(10000);
else
count += len;
}
if (count != tagsize) {
DEBUG("mp3_decode: error parsing ID3 tag\n");
goto fail;
}
id3_data = allocated;
}
id3Tag = id3_tag_parse(id3_data, tagsize);
if (!id3Tag)
goto fail;
if (mpdTag) {
tmpMpdTag = parseId3Tag(id3Tag);
if (tmpMpdTag) {
if (*mpdTag)
freeMpdTag(*mpdTag);
*mpdTag = tmpMpdTag;
}
}
if (replayGainInfo) {
tmpReplayGainInfo = parseId3ReplayGainInfo(id3Tag);
if (tmpReplayGainInfo) {
if (*replayGainInfo)
freeReplayGainInfo(*replayGainInfo);
*replayGainInfo = tmpReplayGainInfo;
}
}
id3_tag_delete(id3Tag);
fail:
if (allocated)
free(allocated);
}
#endif
static int decodeNextFrameHeader(mp3DecodeData * data, MpdTag ** tag,
ReplayGainInfo ** replayGainInfo)
{
enum mad_layer layer;
if ((data->stream).buffer == NULL
|| (data->stream).error == MAD_ERROR_BUFLEN) {
if (fillMp3InputBuffer(data) < 0) {
return DECODE_BREAK;
}
}
if (mad_header_decode(&data->frame.header, &data->stream)) {
#ifdef HAVE_ID3TAG
if ((data->stream).error == MAD_ERROR_LOSTSYNC &&
(data->stream).this_frame) {
signed long tagsize = id3_tag_query((data->stream).
this_frame,
(data->stream).
bufend -
(data->stream).
this_frame);
if (tagsize > 0) {
if (tag && !(*tag)) {
mp3_parseId3Tag(data, (size_t)tagsize,
tag, replayGainInfo);
} else {
mad_stream_skip(&(data->stream),
tagsize);
}
return DECODE_CONT;
}
}
#endif
if (MAD_RECOVERABLE((data->stream).error)) {
return DECODE_SKIP;
} else {
if ((data->stream).error == MAD_ERROR_BUFLEN)
return DECODE_CONT;
else {
ERROR("unrecoverable frame level error "
"(%s).\n",
mad_stream_errorstr(&data->stream));
data->flush = 0;
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 int decodeNextFrame(mp3DecodeData * data)
{
if ((data->stream).buffer == NULL
|| (data->stream).error == MAD_ERROR_BUFLEN) {
if (fillMp3InputBuffer(data) < 0) {
return DECODE_BREAK;
}
}
if (mad_frame_decode(&data->frame, &data->stream)) {
#ifdef HAVE_ID3TAG
if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
signed long tagsize = id3_tag_query((data->stream).
this_frame,
(data->stream).
bufend -
(data->stream).
this_frame);
if (tagsize > 0) {
mad_stream_skip(&(data->stream), tagsize);
return DECODE_CONT;
}
}
#endif
if (MAD_RECOVERABLE((data->stream).error)) {
return DECODE_SKIP;
} else {
if ((data->stream).error == MAD_ERROR_BUFLEN)
return DECODE_CONT;
else {
ERROR("unrecoverable frame level error "
"(%s).\n",
mad_stream_errorstr(&data->stream));
data->flush = 0;
return DECODE_BREAK;
}
}
}
return DECODE_OK;
}
/* xing stuff stolen from alsaplayer, and heavily modified by jat */
#define XI_MAGIC (('X' << 8) | 'i')
#define NG_MAGIC (('n' << 8) | 'g')
#define IN_MAGIC (('I' << 8) | 'n')
#define FO_MAGIC (('f' << 8) | 'o')
enum xing_magic {
XING_MAGIC_XING, /* VBR */
XING_MAGIC_INFO /* CBR */
};
struct xing {
long flags; /* valid fields (see below) */
unsigned long frames; /* total number of frames */
unsigned long bytes; /* total number of bytes */
unsigned char toc[100]; /* 100-point seek table */
long scale; /* VBR quality */
enum xing_magic magic; /* header magic */
};
enum {
XING_FRAMES = 0x00000001L,
XING_BYTES = 0x00000002L,
XING_TOC = 0x00000004L,
XING_SCALE = 0x00000008L
};
struct version {
unsigned major;
unsigned minor;
};
struct lame {
char encoder[10]; /* 9 byte encoder name/version ("LAME3.97b") */
struct version version; /* struct containing just the version */
float peak; /* replaygain peak */
float trackGain; /* replaygain track gain */
float albumGain; /* replaygain album gain */
int encoderDelay; /* # of added samples at start of mp3 */
int encoderPadding; /* # of added samples at end of mp3 */
int crc; /* CRC of the first 190 bytes of this frame */
};
static int parse_xing(struct xing *xing, struct mad_bitptr *ptr, int *oldbitlen)
{
unsigned long bits;
int bitlen;
int bitsleft;
int i;
bitlen = *oldbitlen;
if (bitlen < 16) goto fail;
bits = mad_bit_read(ptr, 16);
bitlen -= 16;
if (bits == XI_MAGIC) {
if (bitlen < 16) goto fail;
if (mad_bit_read(ptr, 16) != NG_MAGIC) goto fail;
bitlen -= 16;
xing->magic = XING_MAGIC_XING;
} else if (bits == IN_MAGIC) {
if (bitlen < 16) goto fail;
if (mad_bit_read(ptr, 16) != FO_MAGIC) goto fail;
bitlen -= 16;
xing->magic = XING_MAGIC_INFO;
}
else if (bits == NG_MAGIC) xing->magic = XING_MAGIC_XING;
else if (bits == FO_MAGIC) xing->magic = XING_MAGIC_INFO;
else goto fail;
if (bitlen < 32) goto fail;
xing->flags = mad_bit_read(ptr, 32);
bitlen -= 32;
if (xing->flags & XING_FRAMES) {
if (bitlen < 32) goto fail;
xing->frames = mad_bit_read(ptr, 32);
bitlen -= 32;
}
if (xing->flags & XING_BYTES) {
if (bitlen < 32) goto fail;
xing->bytes = mad_bit_read(ptr, 32);
bitlen -= 32;
}
if (xing->flags & XING_TOC) {
if (bitlen < 800) goto fail;
for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(ptr, 8);
bitlen -= 800;
}
if (xing->flags & XING_SCALE) {
if (bitlen < 32) goto fail;
xing->scale = mad_bit_read(ptr, 32);
bitlen -= 32;
}
/* Make sure we consume no less than 120 bytes (960 bits) in hopes that
* the LAME tag is found there, and not right after the Xing header */
bitsleft = 960 - ((*oldbitlen) - bitlen);
if (bitsleft < 0) goto fail;
else if (bitsleft > 0) {
mad_bit_read(ptr, bitsleft);
bitlen -= bitsleft;
}
*oldbitlen = bitlen;
return 1;
fail:
xing->flags = 0;
return 0;
}
static int parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen)
{
int adj = 0;
int name;
int orig;
int sign;
int gain;
int i;
/* Unlike the xing header, the lame tag has a fixed length. Fail if
* not all 36 bytes (288 bits) are there. */
if (*bitlen < 288)
return 0;
for (i = 0; i < 9; i++)
lame->encoder[i] = (char)mad_bit_read(ptr, 8);
lame->encoder[9] = '\0';
*bitlen -= 72;
/* This is technically incorrect, since the encoder might not be lame.
* But there's no other way to determine if this is a lame tag, and we
* wouldn't want to go reading a tag that's not there. */
if (strncmp(lame->encoder, "LAME", 4) != 0)
return 0;
if (sscanf(lame->encoder+4, "%u.%u",
&lame->version.major, &lame->version.minor) != 2)
return 0;
DEBUG("detected LAME version %i.%i (\"%s\")\n",
lame->version.major, lame->version.minor, lame->encoder);
/* The reference volume was changed from the 83dB used in the
* ReplayGain spec to 89dB in lame 3.95.1. Bump the gain for older
* versions, since everyone else uses 89dB instead of 83dB.
* Unfortunately, lame didn't differentiate between 3.95 and 3.95.1, so
* it's impossible to make the proper adjustment for 3.95.
* Fortunately, 3.95 was only out for about a day before 3.95.1 was
* released. -- tmz */
if (lame->version.major < 3 ||
(lame->version.major == 3 && lame->version.minor < 95))
adj = 6;
mad_bit_read(ptr, 16);
lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */
DEBUG("LAME peak found: %f\n", lame->peak);
lame->trackGain = 0;
name = mad_bit_read(ptr, 3); /* gain name */
orig = mad_bit_read(ptr, 3); /* gain originator */
sign = mad_bit_read(ptr, 1); /* sign bit */
gain = mad_bit_read(ptr, 9); /* gain*10 */
if (gain && name == 1 && orig != 0) {
lame->trackGain = ((sign ? -gain : gain) / 10.0) + adj;
DEBUG("LAME track gain found: %f\n", lame->trackGain);
}
/* tmz reports that this isn't currently written by any version of lame
* (as of 3.97). Since we have no way of testing it, don't use it.
* Wouldn't want to go blowing someone's ears just because we read it
* wrong. :P -- jat */
lame->albumGain = 0;
#if 0
name = mad_bit_read(ptr, 3); /* gain name */
orig = mad_bit_read(ptr, 3); /* gain originator */
sign = mad_bit_read(ptr, 1); /* sign bit */
gain = mad_bit_read(ptr, 9); /* gain*10 */
if (gain && name == 2 && orig != 0) {
lame->albumGain = ((sign ? -gain : gain) / 10.0) + adj;
DEBUG("LAME album gain found: %f\n", lame->trackGain);
}
#else
mad_bit_read(ptr, 16);
#endif
mad_bit_read(ptr, 16);
lame->encoderDelay = mad_bit_read(ptr, 12);
lame->encoderPadding = mad_bit_read(ptr, 12);
DEBUG("encoder delay is %i, encoder padding is %i\n",
lame->encoderDelay, lame->encoderPadding);
mad_bit_read(ptr, 80);
lame->crc = mad_bit_read(ptr, 16);
*bitlen -= 216;
return 1;
}
static int decodeFirstFrame(mp3DecodeData * data,
MpdTag ** tag, ReplayGainInfo ** replayGainInfo)
{
struct xing xing;
struct lame lame;
struct mad_bitptr ptr;
int bitlen;
int ret;
/* stfu gcc */
memset(&xing, 0, sizeof(struct xing));
xing.flags = 0;
while (1) {
while ((ret = decodeNextFrameHeader(data, tag, replayGainInfo)) == DECODE_CONT &&
!dc.stop);
if (ret == DECODE_BREAK || dc.stop) return -1;
if (ret == DECODE_SKIP) continue;
while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
!dc.stop);
if (ret == DECODE_BREAK || dc.stop) return -1;
if (ret == DECODE_OK) break;
}
ptr = data->stream.anc_ptr;
bitlen = data->stream.anc_bitlen;
/*
* Attempt to calulcate the length of the song from filesize
*/
{
size_t offset = data->inStream->offset;
mad_timer_t duration = data->frame.header.duration;
float frameTime = ((float)mad_timer_count(duration,
MAD_UNITS_MILLISECONDS)) / 1000;
if (data->stream.this_frame != NULL)
offset -= data->stream.bufend - data->stream.this_frame;
else
offset -= data->stream.bufend - data->stream.buffer;
if (data->inStream->size >= offset) {
data->totalTime = ((data->inStream->size - offset) *
8.0) / (data->frame).header.bitrate;
data->maxFrames = data->totalTime / frameTime +
FRAMES_CUSHION;
} else {
data->maxFrames = FRAMES_CUSHION;
data->totalTime = 0;
}
}
/*
* if an xing tag exists, use that!
*/
if (parse_xing(&xing, &ptr, &bitlen)) {
data->foundXing = 1;
data->muteFrame = MUTEFRAME_SKIP;
if ((xing.flags & XING_FRAMES) && xing.frames) {
mad_timer_t duration = data->frame.header.duration;
mad_timer_multiply(&duration, xing.frames);
data->totalTime = ((float)mad_timer_count(duration, MAD_UNITS_MILLISECONDS)) / 1000;
data->maxFrames = xing.frames;
}
if (parse_lame(&lame, &ptr, &bitlen)) {
if (gaplessPlaybackEnabled &&
data->inStream->seekable) {
data->dropSamplesAtStart = lame.encoderDelay +
DECODERDELAY;
data->dropSamplesAtEnd = lame.encoderPadding;
}
/* Album gain isn't currently used. See comment in
* parse_lame() for details. -- jat */
if (replayGainInfo && !*replayGainInfo &&
lame.trackGain) {
*replayGainInfo = newReplayGainInfo();
(*replayGainInfo)->trackGain = lame.trackGain;
(*replayGainInfo)->trackPeak = lame.peak;
}
}
}
if (!data->maxFrames) return -1;
data->frameOffset = xmalloc(sizeof(long) * data->maxFrames);
data->times = xmalloc(sizeof(mad_timer_t) * data->maxFrames);
return 0;
}
static void mp3DecodeDataFinalize(mp3DecodeData * data)
{
mad_synth_finish(&data->synth);
mad_frame_finish(&data->frame);
mad_stream_finish(&data->stream);
if (data->frameOffset) free(data->frameOffset);
if (data->times) free(data->times);
}
/* this is primarily used for getting total time for tags */
static int getMp3TotalTime(char *file)
{
InputStream inStream;
mp3DecodeData data;
int ret;
if (openInputStream(&inStream, file) < 0)
return -1;
initMp3DecodeData(&data, &inStream);
if (decodeFirstFrame(&data, NULL, NULL) < 0)
ret = -1;
else
ret = data.totalTime + 0.5;
mp3DecodeDataFinalize(&data);
closeInputStream(&inStream);
return ret;
}
static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
MpdTag ** tag,
ReplayGainInfo ** replayGainInfo)
{
initMp3DecodeData(data, inStream);
*tag = NULL;
if (decodeFirstFrame(data, tag, replayGainInfo) < 0) {
mp3DecodeDataFinalize(data);
if (tag && *tag)
freeMpdTag(*tag);
return -1;
}
return 0;
}
static int mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
{
int samplesPerFrame;
int samplesLeft;
int i;
int ret;
int skip;
if (data->currentFrame >= data->highestFrame) {
mad_timer_add(&data->timer, (data->frame).header.duration);
data->bitRate = (data->frame).header.bitrate;
if (data->currentFrame >= data->maxFrames) {
data->currentFrame = data->maxFrames - 1;
} else {
data->highestFrame++;
}
data->frameOffset[data->currentFrame] = data->inStream->offset;
if (data->stream.this_frame != NULL) {
data->frameOffset[data->currentFrame] -=
data->stream.bufend - data->stream.this_frame;
} else {
data->frameOffset[data->currentFrame] -=
data->stream.bufend - data->stream.buffer;
}
data->times[data->currentFrame] = data->timer;
} else {
data->timer = data->times[data->currentFrame];
}
data->currentFrame++;
data->elapsedTime =
((float)mad_timer_count(data->timer, MAD_UNITS_MILLISECONDS)) /
1000;
switch (data->muteFrame) {
case MUTEFRAME_SKIP:
data->muteFrame = 0;
break;
case MUTEFRAME_SEEK:
if (dc.seekWhere <= data->elapsedTime) {
data->outputPtr = data->outputBuffer;
clearOutputBuffer();
data->muteFrame = 0;
dc.seek = 0;
Initial cut of fork() => pthreads() for decoder and player I initially started to do a heavy rewrite that changed the way processes communicated, but that was too much to do at once. So this change only focuses on replacing the player and decode processes with threads and using condition variables instead of polling in loops; so the changeset itself is quiet small. * The shared output buffer variables will still need locking to guard against race conditions. So in this effect, we're probably just as buggy as before. The reduced context-switching overhead of using threads instead of processes may even make bugs show up more or less often... * Basic functionality appears to be working for playing local (and NFS) audio, including: play, pause, stop, seek, previous, next, and main playlist editing * I haven't tested HTTP streams yet, they should work. * I've only tested ALSA and Icecast. ALSA works fine, Icecast metadata seems to get screwy at times and breaks song advancement in the playlist at times. * state file loading works, too (after some last-minute hacks with non-blocking wakeup functions) * The non-blocking (*_nb) variants of the task management functions are probably overused. They're more lenient and easier to use because much of our code is still based on our previous polling-based system. * It currently segfaults on exit. I haven't paid much attention to the exit/signal-handling routines other than ensuring it compiles. At least the state file seems to work. We don't do any cleanups of the threads on exit, yet. * Update is still done in a child process and not in a thread. To do this in a thread, we'll need to ensure it does proper locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: master - just does waitpid() + fork() in a loop \- main thread \- decoder thread \- player thread At the beginning of every song, the main thread will set a dirty flag and update the state file. This way, if we encounter a song that triggers a segfault killing the main thread, the master will start the replacement main on the next song. * The main thread still wakes up every second on select() to check for signals; which affects power management. [merged r7138 from branches/ew] git-svn-id: https://svn.musicpd.org/mpd/trunk@7240 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2008-04-12 06:08:00 +02:00
decoder_wakeup_player();
}
break;
default:
mad_synth_frame(&data->synth, &data->frame);
if (!data->foundFirstFrame) {
samplesPerFrame = (data->synth).pcm.length;
data->dropFramesAtStart = data->dropSamplesAtStart / samplesPerFrame;
data->dropFramesAtEnd = data->dropSamplesAtEnd / samplesPerFrame;
data->dropSamplesAtStart = data->dropSamplesAtStart % samplesPerFrame;
data->dropSamplesAtEnd = data->dropSamplesAtEnd % samplesPerFrame;
data->foundFirstFrame = 1;
}
if (data->dropFramesAtStart > 0) {
data->dropFramesAtStart--;
break;
} else if ((data->dropFramesAtEnd > 0) &&
(data->currentFrame == (data->maxFrames + 1 - data->dropFramesAtEnd))) {
/* stop decoding, effectively dropping all remaining
* frames */
return DECODE_BREAK;
}
if (data->inStream->metaTitle) {
MpdTag *tag = newMpdTag();
if (data->inStream->metaName) {
addItemToMpdTag(tag,
TAG_ITEM_NAME,
data->inStream->metaName);
}
addItemToMpdTag(tag, TAG_ITEM_TITLE,
data->inStream->metaTitle);
free(data->inStream->metaTitle);
data->inStream->metaTitle = NULL;
freeMpdTag(tag);
}
samplesLeft = (data->synth).pcm.length;
for (i = 0; i < (data->synth).pcm.length; i++) {
mpd_sint16 *sample;
samplesLeft--;
if (!data->decodedFirstFrame &&
(i < data->dropSamplesAtStart)) {
continue;
} else if (data->dropSamplesAtEnd &&
(data->currentFrame == (data->maxFrames - data->dropFramesAtEnd)) &&
(samplesLeft < data->dropSamplesAtEnd)) {
/* stop decoding, effectively dropping
* all remaining samples */
return DECODE_BREAK;
}
sample = (mpd_sint16 *) data->outputPtr;
*sample = (mpd_sint16) audio_linear_dither(16,
(data->synth).pcm.samples[0][i],
&(data->dither));
data->outputPtr += 2;
if (MAD_NCHANNELS(&(data->frame).header) == 2) {
sample = (mpd_sint16 *) data->outputPtr;
*sample = (mpd_sint16) audio_linear_dither(16,
(data->synth).pcm.samples[1][i],
&(data->dither));
data->outputPtr += 2;
}
if (data->outputPtr >= data->outputBufferEnd) {
ret = sendDataToOutputBuffer(data->inStream,
data->inStream->seekable,
data->outputBuffer,
data->outputPtr - data->outputBuffer,
data->elapsedTime,
data->bitRate / 1000,
(replayGainInfo != NULL) ? *replayGainInfo : NULL);
if (ret == OUTPUT_BUFFER_DC_STOP) {
data->flush = 0;
return DECODE_BREAK;
}
data->outputPtr = data->outputBuffer;
if (ret == OUTPUT_BUFFER_DC_SEEK)
break;
}
}
data->decodedFirstFrame = 1;
if (dc.seek && data->inStream->seekable) {
long j = 0;
data->muteFrame = MUTEFRAME_SEEK;
while (j < data->highestFrame && dc.seekWhere >
((float)mad_timer_count(data->times[j],
MAD_UNITS_MILLISECONDS))
/ 1000) {
j++;
}
if (j < data->highestFrame) {
if (seekMp3InputBuffer(data,
data->frameOffset[j]) ==
0) {
data->outputPtr = data->outputBuffer;
clearOutputBuffer();
data->currentFrame = j;
} else
dc.seekError = 1;
data->muteFrame = 0;
dc.seek = 0;
Initial cut of fork() => pthreads() for decoder and player I initially started to do a heavy rewrite that changed the way processes communicated, but that was too much to do at once. So this change only focuses on replacing the player and decode processes with threads and using condition variables instead of polling in loops; so the changeset itself is quiet small. * The shared output buffer variables will still need locking to guard against race conditions. So in this effect, we're probably just as buggy as before. The reduced context-switching overhead of using threads instead of processes may even make bugs show up more or less often... * Basic functionality appears to be working for playing local (and NFS) audio, including: play, pause, stop, seek, previous, next, and main playlist editing * I haven't tested HTTP streams yet, they should work. * I've only tested ALSA and Icecast. ALSA works fine, Icecast metadata seems to get screwy at times and breaks song advancement in the playlist at times. * state file loading works, too (after some last-minute hacks with non-blocking wakeup functions) * The non-blocking (*_nb) variants of the task management functions are probably overused. They're more lenient and easier to use because much of our code is still based on our previous polling-based system. * It currently segfaults on exit. I haven't paid much attention to the exit/signal-handling routines other than ensuring it compiles. At least the state file seems to work. We don't do any cleanups of the threads on exit, yet. * Update is still done in a child process and not in a thread. To do this in a thread, we'll need to ensure it does proper locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: master - just does waitpid() + fork() in a loop \- main thread \- decoder thread \- player thread At the beginning of every song, the main thread will set a dirty flag and update the state file. This way, if we encounter a song that triggers a segfault killing the main thread, the master will start the replacement main on the next song. * The main thread still wakes up every second on select() to check for signals; which affects power management. [merged r7138 from branches/ew] git-svn-id: https://svn.musicpd.org/mpd/trunk@7240 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2008-04-12 06:08:00 +02:00
decoder_wakeup_player();
}
} else if (dc.seek && !data->inStream->seekable) {
dc.seek = 0;
dc.seekError = 1;
Initial cut of fork() => pthreads() for decoder and player I initially started to do a heavy rewrite that changed the way processes communicated, but that was too much to do at once. So this change only focuses on replacing the player and decode processes with threads and using condition variables instead of polling in loops; so the changeset itself is quiet small. * The shared output buffer variables will still need locking to guard against race conditions. So in this effect, we're probably just as buggy as before. The reduced context-switching overhead of using threads instead of processes may even make bugs show up more or less often... * Basic functionality appears to be working for playing local (and NFS) audio, including: play, pause, stop, seek, previous, next, and main playlist editing * I haven't tested HTTP streams yet, they should work. * I've only tested ALSA and Icecast. ALSA works fine, Icecast metadata seems to get screwy at times and breaks song advancement in the playlist at times. * state file loading works, too (after some last-minute hacks with non-blocking wakeup functions) * The non-blocking (*_nb) variants of the task management functions are probably overused. They're more lenient and easier to use because much of our code is still based on our previous polling-based system. * It currently segfaults on exit. I haven't paid much attention to the exit/signal-handling routines other than ensuring it compiles. At least the state file seems to work. We don't do any cleanups of the threads on exit, yet. * Update is still done in a child process and not in a thread. To do this in a thread, we'll need to ensure it does proper locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: master - just does waitpid() + fork() in a loop \- main thread \- decoder thread \- player thread At the beginning of every song, the main thread will set a dirty flag and update the state file. This way, if we encounter a song that triggers a segfault killing the main thread, the master will start the replacement main on the next song. * The main thread still wakes up every second on select() to check for signals; which affects power management. [merged r7138 from branches/ew] git-svn-id: https://svn.musicpd.org/mpd/trunk@7240 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2008-04-12 06:08:00 +02:00
decoder_wakeup_player();
}
}
while (1) {
skip = 0;
while ((ret =
decodeNextFrameHeader(data, NULL,
replayGainInfo)) == DECODE_CONT
&& !dc.stop) ;
if (ret == DECODE_BREAK || dc.stop || dc.seek)
break;
else if (ret == DECODE_SKIP)
skip = 1;
if (!data->muteFrame) {
while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
!dc.stop && !dc.seek) ;
if (ret == DECODE_BREAK || dc.stop || dc.seek)
break;
}
if (!skip && ret == DECODE_OK)
break;
}
if (dc.stop)
return DECODE_BREAK;
return ret;
}
static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
AudioFormat * af)
{
af->bits = 16;
af->sampleRate = (data->frame).header.samplerate;
af->channels = MAD_NCHANNELS(&(data->frame).header);
}
static int mp3_decode(InputStream * inStream)
{
mp3DecodeData data;
MpdTag *tag = NULL;
ReplayGainInfo *replayGainInfo = NULL;
if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) <
0) {
if (!dc.stop) {
ERROR
("Input does not appear to be a mp3 bit stream.\n");
return -1;
}
return 0;
}
initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat));
getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat));
dc.totalTime = data.totalTime;
if (inStream->metaTitle) {
if (tag)
freeMpdTag(tag);
tag = newMpdTag();
addItemToMpdTag(tag, TAG_ITEM_TITLE, inStream->metaTitle);
free(inStream->metaTitle);
inStream->metaTitle = NULL;
if (inStream->metaName) {
addItemToMpdTag(tag, TAG_ITEM_NAME, inStream->metaName);
}
freeMpdTag(tag);
} else if (tag) {
if (inStream->metaName) {
clearItemsFromMpdTag(tag, TAG_ITEM_NAME);
addItemToMpdTag(tag, TAG_ITEM_NAME, inStream->metaName);
}
freeMpdTag(tag);
} else if (inStream->metaName) {
tag = newMpdTag();
if (inStream->metaName) {
addItemToMpdTag(tag, TAG_ITEM_NAME, inStream->metaName);
}
freeMpdTag(tag);
}
dc.state = DECODE_STATE_DECODE;
while (mp3Read(&data, &replayGainInfo) != DECODE_BREAK) ;
/* send last little bit if not dc.stop */
if (!dc.stop && data.outputPtr != data.outputBuffer && data.flush) {
sendDataToOutputBuffer(NULL,
data.inStream->seekable,
data.outputBuffer,
data.outputPtr - data.outputBuffer,
data.elapsedTime, data.bitRate / 1000,
replayGainInfo);
}
if (replayGainInfo)
freeReplayGainInfo(replayGainInfo);
if (dc.seek && data.muteFrame == MUTEFRAME_SEEK) {
clearOutputBuffer();
dc.seek = 0;
Initial cut of fork() => pthreads() for decoder and player I initially started to do a heavy rewrite that changed the way processes communicated, but that was too much to do at once. So this change only focuses on replacing the player and decode processes with threads and using condition variables instead of polling in loops; so the changeset itself is quiet small. * The shared output buffer variables will still need locking to guard against race conditions. So in this effect, we're probably just as buggy as before. The reduced context-switching overhead of using threads instead of processes may even make bugs show up more or less often... * Basic functionality appears to be working for playing local (and NFS) audio, including: play, pause, stop, seek, previous, next, and main playlist editing * I haven't tested HTTP streams yet, they should work. * I've only tested ALSA and Icecast. ALSA works fine, Icecast metadata seems to get screwy at times and breaks song advancement in the playlist at times. * state file loading works, too (after some last-minute hacks with non-blocking wakeup functions) * The non-blocking (*_nb) variants of the task management functions are probably overused. They're more lenient and easier to use because much of our code is still based on our previous polling-based system. * It currently segfaults on exit. I haven't paid much attention to the exit/signal-handling routines other than ensuring it compiles. At least the state file seems to work. We don't do any cleanups of the threads on exit, yet. * Update is still done in a child process and not in a thread. To do this in a thread, we'll need to ensure it does proper locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: master - just does waitpid() + fork() in a loop \- main thread \- decoder thread \- player thread At the beginning of every song, the main thread will set a dirty flag and update the state file. This way, if we encounter a song that triggers a segfault killing the main thread, the master will start the replacement main on the next song. * The main thread still wakes up every second on select() to check for signals; which affects power management. [merged r7138 from branches/ew] git-svn-id: https://svn.musicpd.org/mpd/trunk@7240 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2008-04-12 06:08:00 +02:00
decoder_wakeup_player();
}
flushOutputBuffer();
mp3DecodeDataFinalize(&data);
return 0;
}
static MpdTag *mp3_tagDup(char *file)
{
MpdTag *ret = NULL;
int total_time;
ret = id3Dup(file);
total_time = getMp3TotalTime(file);
if (total_time >= 0) {
if (!ret)
ret = newMpdTag();
ret->time = total_time;
} else {
DEBUG("mp3_tagDup: Failed to get total song time from: %s\n",
file);
}
return ret;
}
static const char *mp3_suffixes[] = { "mp3", "mp2", NULL };
static const char *mp3_mimeTypes[] = { "audio/mpeg", NULL };
InputPlugin mp3Plugin = {
"mp3",
mp3_plugin_init,
NULL,
NULL,
mp3_decode,
NULL,
mp3_tagDup,
INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
mp3_suffixes,
mp3_mimeTypes
};
#else
InputPlugin mp3Plugin;
#endif /* HAVE_MAD */