mpd/src/inputPlugins/mp4_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_FAAD
#include "../utils.h"
#include "../audio.h"
#include "../log.h"
#include "../pcm_utils.h"
#include "../inputStream.h"
#include "../outputBuffer.h"
#include "../decode.h"
#include "../os_compat.h"
#include "../mp4ff/mp4ff.h"
#include <faad.h>
/* all code here is either based on or copied from FAAD2's frontend code */
static int mp4_getAACTrack(mp4ff_t * infile)
{
/* find AAC track */
int i, rc;
int numTracks = mp4ff_total_tracks(infile);
for (i = 0; i < numTracks; i++) {
unsigned char *buff = NULL;
unsigned int buff_size = 0;
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
mp4AudioSpecificConfig mp4ASC;
#else
unsigned long dummy1_32;
unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
dummy7_8, dummy8_8;
#endif
mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff) {
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
#else
rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
&dummy3_8, &dummy4_8,
&dummy5_8, &dummy6_8,
&dummy7_8, &dummy8_8);
#endif
free(buff);
if (rc < 0)
continue;
return i;
}
}
/* can't decode this */
return -1;
}
static uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer,
uint32_t length)
{
return readFromInputStream((InputStream *) inStream, buffer, 1, length);
}
static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
{
return seekInputStream((InputStream *) inStream, position, SEEK_SET);
}
static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
InputStream * inStream)
{
mp4ff_t *mp4fh;
mp4ff_callback_t *mp4cb;
int32_t track;
float time;
int32_t scale;
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
unsigned char *mp4Buffer;
unsigned int mp4BufferSize;
unsigned long sampleRate;
unsigned char channels;
long sampleId;
long numSamples;
int eof = 0;
long dur;
unsigned int sampleCount;
char *sampleBuffer;
size_t sampleBufferLen;
unsigned int initial = 1;
float *seekTable;
long seekTableEnd = -1;
int seekPositionFound = 0;
long offset;
mpd_uint16 bitRate = 0;
int seeking = 0;
mp4cb = xmalloc(sizeof(mp4ff_callback_t));
mp4cb->read = mp4_inputStreamReadCallback;
mp4cb->seek = mp4_inputStreamSeekCallback;
mp4cb->user_data = inStream;
mp4fh = mp4ff_open_read(mp4cb);
if (!mp4fh) {
ERROR("Input does not appear to be a mp4 stream.\n");
free(mp4cb);
return -1;
}
track = mp4_getAACTrack(mp4fh);
if (track < 0) {
ERROR("No AAC track found in mp4 stream.\n");
mp4ff_close(mp4fh);
free(mp4cb);
return -1;
}
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
config->downMatrix = 1;
#endif
#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
config->dontUpSampleImplicitSBR = 0;
#endif
faacDecSetConfiguration(decoder, config);
dc->audioFormat.bits = 16;
mp4Buffer = NULL;
mp4BufferSize = 0;
mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
if (faacDecInit2
(decoder, mp4Buffer, mp4BufferSize, &sampleRate, &channels) < 0) {
ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
free(mp4cb);
return -1;
}
dc->audioFormat.sampleRate = sampleRate;
dc->audioFormat.channels = channels;
time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
if (mp4Buffer)
free(mp4Buffer);
if (scale < 0) {
ERROR("Error getting audio format of mp4 AAC track.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
free(mp4cb);
return -1;
}
dc->totalTime = ((float)time) / scale;
numSamples = mp4ff_num_samples(mp4fh, track);
time = 0.0;
seekTable = xmalloc(sizeof(float) * numSamples);
for (sampleId = 0; sampleId < numSamples && !eof; sampleId++) {
if (dc->seek)
seeking = 1;
if (seeking && seekTableEnd > 1 &&
seekTable[seekTableEnd] >= dc->seekWhere) {
int i = 2;
while (seekTable[i] < dc->seekWhere)
i++;
sampleId = i - 1;
time = seekTable[sampleId];
}
dur = mp4ff_get_sample_duration(mp4fh, track, sampleId);
offset = mp4ff_get_sample_offset(mp4fh, track, sampleId);
if (sampleId > seekTableEnd) {
seekTable[sampleId] = time;
seekTableEnd = sampleId;
}
if (sampleId == 0)
dur = 0;
if (offset > dur)
dur = 0;
else
dur -= offset;
time += ((float)dur) / scale;
if (seeking && time > dc->seekWhere)
seekPositionFound = 1;
if (seeking && seekPositionFound) {
seekPositionFound = 0;
clearOutputBuffer(cb);
seeking = 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();
}
if (seeking)
continue;
if (mp4ff_read_sample(mp4fh, track, sampleId, &mp4Buffer,
&mp4BufferSize) == 0) {
eof = 1;
continue;
}
#ifdef HAVE_FAAD_BUFLEN_FUNCS
sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer,
mp4BufferSize);
#else
sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer);
#endif
if (mp4Buffer)
free(mp4Buffer);
if (frameInfo.error > 0) {
ERROR("faad2 error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
eof = 1;
break;
}
if (dc->state != DECODE_STATE_DECODE) {
channels = frameInfo.channels;
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
#endif
dc->audioFormat.sampleRate = scale;
dc->audioFormat.channels = frameInfo.channels;
getOutputAudioFormat(&(dc->audioFormat),
&(cb->audioFormat));
dc->state = DECODE_STATE_DECODE;
}
if (channels * (dur + offset) > frameInfo.samples) {
dur = frameInfo.samples / channels;
offset = 0;
}
sampleCount = (unsigned long)(dur * channels);
if (sampleCount > 0) {
initial = 0;
bitRate = frameInfo.bytesconsumed * 8.0 *
frameInfo.channels * scale /
frameInfo.samples / 1000 + 0.5;
}
sampleBufferLen = sampleCount * 2;
sampleBuffer += offset * channels * 2;
sendDataToOutputBuffer(cb, inStream, dc, 1, sampleBuffer,
sampleBufferLen, time, bitRate, NULL);
if (dc->stop) {
eof = 1;
break;
}
}
free(seekTable);
faacDecClose(decoder);
mp4ff_close(mp4fh);
free(mp4cb);
if (dc->state != DECODE_STATE_DECODE)
return -1;
if (dc->seek && seeking) {
clearOutputBuffer(cb);
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(cb);
return 0;
}
static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound)
{
MpdTag *ret = NULL;
InputStream inStream;
mp4ff_t *mp4fh;
mp4ff_callback_t *cb;
int32_t track;
int32_t time;
int32_t scale;
int i;
*mp4MetadataFound = 0;
if (openInputStream(&inStream, file) < 0) {
DEBUG("mp4DataDup: Failed to open file: %s\n", file);
return NULL;
}
cb = xmalloc(sizeof(mp4ff_callback_t));
cb->read = mp4_inputStreamReadCallback;
cb->seek = mp4_inputStreamSeekCallback;
cb->user_data = &inStream;
mp4fh = mp4ff_open_read(cb);
if (!mp4fh) {
free(cb);
closeInputStream(&inStream);
return NULL;
}
track = mp4_getAACTrack(mp4fh);
if (track < 0) {
mp4ff_close(mp4fh);
closeInputStream(&inStream);
free(cb);
return NULL;
}
ret = newMpdTag();
time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
if (scale < 0) {
mp4ff_close(mp4fh);
closeInputStream(&inStream);
free(cb);
freeMpdTag(ret);
return NULL;
}
ret->time = ((float)time) / scale + 0.5;
for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
char *item;
char *value;
mp4ff_meta_get_by_index(mp4fh, i, &item, &value);
if (0 == strcasecmp("artist", item)) {
addItemToMpdTag(ret, TAG_ITEM_ARTIST, value);
*mp4MetadataFound = 1;
} else if (0 == strcasecmp("title", item)) {
addItemToMpdTag(ret, TAG_ITEM_TITLE, value);
*mp4MetadataFound = 1;
} else if (0 == strcasecmp("album", item)) {
addItemToMpdTag(ret, TAG_ITEM_ALBUM, value);
*mp4MetadataFound = 1;
} else if (0 == strcasecmp("track", item)) {
addItemToMpdTag(ret, TAG_ITEM_TRACK, value);
*mp4MetadataFound = 1;
} else if (0 == strcasecmp("disc", item)) { /* Is that the correct id? */
addItemToMpdTag(ret, TAG_ITEM_DISC, value);
*mp4MetadataFound = 1;
} else if (0 == strcasecmp("genre", item)) {
addItemToMpdTag(ret, TAG_ITEM_GENRE, value);
*mp4MetadataFound = 1;
} else if (0 == strcasecmp("date", item)) {
addItemToMpdTag(ret, TAG_ITEM_DATE, value);
*mp4MetadataFound = 1;
}
free(item);
free(value);
}
mp4ff_close(mp4fh);
closeInputStream(&inStream);
free(cb);
return ret;
}
static MpdTag *mp4TagDup(char *file)
{
MpdTag *ret = NULL;
int mp4MetadataFound = 0;
ret = mp4DataDup(file, &mp4MetadataFound);
if (!ret)
return NULL;
if (!mp4MetadataFound) {
MpdTag *temp = id3Dup(file);
if (temp) {
temp->time = ret->time;
freeMpdTag(ret);
ret = temp;
}
}
return ret;
}
static char *mp4_suffixes[] = { "m4a", "mp4", NULL };
static char *mp4_mimeTypes[] = { "audio/mp4", "audio/m4a", NULL };
InputPlugin mp4Plugin = {
"mp4",
NULL,
NULL,
NULL,
mp4_decode,
NULL,
mp4TagDup,
INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
mp4_suffixes,
mp4_mimeTypes
};
#else
InputPlugin mp4Plugin;
#endif /* HAVE_FAAD */