decoder/mp4ff: delete obsolete plugin

The underlying library has been obsolete for many years.
This commit is contained in:
Max Kellermann 2013-01-28 20:43:01 +01:00
parent 541192c941
commit 88c17926e4
7 changed files with 2 additions and 495 deletions

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@ -91,7 +91,7 @@ Audio File - http://www.68k.org/~michael/audiofile/
For WAVE, AIFF, and AU support. You will need libaudiofile.
FAAD2 - http://www.audiocoding.com/
For MP4/AAC support. You will need libmp4ff.
For MP4/AAC support.
libmpcdec - http://www.musepack.net/
For Musepack support.

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@ -499,7 +499,6 @@ DECODER_LIBS = \
$(MAD_LIBS) \
$(MPG123_LIBS) \
$(OPUS_LIBS) \
$(MP4FF_LIBS) \
$(FFMPEG_LIBS) \
$(MPCDEC_LIBS) \
$(ADPLUG_LIBS) \
@ -549,10 +548,6 @@ if HAVE_FAAD
libdecoder_plugins_a_SOURCES += src/decoder/faad_decoder_plugin.c
endif
if HAVE_MP4
libdecoder_plugins_a_SOURCES += src/decoder/mp4ff_decoder_plugin.c
endif
if HAVE_XIPH
libdecoder_plugins_a_SOURCES += \
src/decoder/XiphTags.c src/decoder/XiphTags.h \

1
NEWS
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@ -5,6 +5,7 @@ ver 0.18 (2012/??/??)
- flac: support FLAC files inside archives
- opus: new decoder plugin for the Opus codec
- vorbis: skip 16 bit quantisation, provide float samples
- mp4ff: obsolete plugin removed
* encoder:
- opus: new encoder plugin for the Opus codec
- vorbis: accept floating point input samples

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@ -872,7 +872,6 @@ dnl ----------------------------------- FAAD ----------------------------------
AM_PATH_FAAD()
AM_CONDITIONAL(HAVE_FAAD, test x$enable_aac = xyes)
AM_CONDITIONAL(HAVE_MP4, test x$enable_mp4 = xyes)
dnl ---------------------------------- ffmpeg ---------------------------------
MPD_AUTO_PKG(ffmpeg, FFMPEG, [libavformat >= 52.31 libavcodec >= 52.20 libavutil >= 49.15],
@ -1148,7 +1147,6 @@ if
test x$enable_mad = xno &&
test x$enable_mikmod = xno; then
test x$enable_modplug = xno &&
test x$enable_mp4 = xno &&
test x$enable_mpc = xno &&
test x$enable_mpg123 = xno &&
test x$enable_opus = xno &&
@ -1639,7 +1637,6 @@ results(mikmod, [MikMod])
results(modplug, [MODPLUG])
results(mad, [MAD])
results(mpg123, [MPG123])
results(mp4, [MP4])
results(mpc, [Musepack])
printf '\n\t'
results(opus, [Opus])

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@ -158,40 +158,6 @@ int main() {
CPPFLAGS=$oldcppflags
fi
if test x$enable_aac = xyes; then
enable_mp4=yes
MP4FF_LIBS="-lmp4ff"
oldcflags=$CFLAGS
oldlibs=$LIBS
oldcppflags=$CPPFLAGS
CFLAGS="$CFLAGS $FAAD_CFLAGS"
LIBS="$LIBS $FAAD_LIBS $MP4FF_LIBS"
CPPFLAGS=$CFLAGS
AC_CHECK_HEADER(mp4ff.h,,enable_mp4=no)
if test x$enable_mp4 = xyes; then
AC_CHECK_LIB(mp4ff,mp4ff_open_read,,enable_mp4=no)
fi
if test x$enable_mp4 = xyes; then
AC_SUBST(MP4FF_LIBS)
AC_DEFINE(HAVE_MP4, 1, [Define to use FAAD2+mp4ff for MP4 decoding])
else
AC_MSG_WARN([libmp4ff needed for MP4 support -- disabling MP4 support])
unset MP4FF_LIBS
fi
CFLAGS=$oldcflags
LIBS=$oldlibs
CPPFLAGS=$oldcppflags
else
enable_mp4=no
FAAD_CFLAGS=""
FAAD_LIBS=""
fi
AC_SUBST(FAAD_CFLAGS)
AC_SUBST(FAAD_LIBS)

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@ -1,448 +0,0 @@
/*
* Copyright (C) 2003-2011 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "decoder_api.h"
#include "audio_check.h"
#include "tag_table.h"
#include "tag_handler.h"
#include <glib.h>
#include <mp4ff.h>
#include <faad.h>
#include <assert.h>
#include <stdlib.h>
#include <unistd.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "mp4ff"
/* all code here is either based on or copied from FAAD2's frontend code */
struct mp4ff_input_stream {
mp4ff_callback_t callback;
struct decoder *decoder;
struct input_stream *input_stream;
};
static int
mp4_get_aac_track(mp4ff_t * infile, faacDecHandle decoder,
uint32_t *sample_rate, unsigned char *channels_r)
{
#ifdef HAVE_FAAD_LONG
/* neaacdec.h declares all arguments as "unsigned long", but
internally expects uint32_t pointers. To avoid gcc
warnings, use this workaround. */
unsigned long *sample_rate_r = (unsigned long*)sample_rate;
#else
uint32_t *sample_rate_r = sample_rate;
#endif
int i, rc;
int num_tracks = mp4ff_total_tracks(infile);
for (i = 0; i < num_tracks; i++) {
unsigned char *buff = NULL;
unsigned int buff_size = 0;
if (mp4ff_get_track_type(infile, i) != 1)
/* not an audio track */
continue;
if (decoder == NULL)
/* have don't have a decoder to initialize -
we're done now, because we found an audio
track */
return i;
mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff == NULL)
continue;
rc = faacDecInit2(decoder, buff, buff_size,
sample_rate_r, channels_r);
free(buff);
if (rc >= 0)
/* found a valid AAC track */
return i;
}
/* can't decode this */
return -1;
}
static uint32_t
mp4_read(void *user_data, void *buffer, uint32_t length)
{
struct mp4ff_input_stream *mis = user_data;
if (length == 0)
/* libmp4ff is known to attempt to read 0 bytes - make
this a special case, because the input_stream API
would not allow this */
return 0;
return decoder_read(mis->decoder, mis->input_stream, buffer, length);
}
static uint32_t
mp4_seek(void *user_data, uint64_t position)
{
struct mp4ff_input_stream *mis = user_data;
return input_stream_lock_seek(mis->input_stream, position, SEEK_SET,
NULL)
? 0 : -1;
}
static const mp4ff_callback_t mpd_mp4ff_callback = {
.read = mp4_read,
.seek = mp4_seek,
};
static mp4ff_t *
mp4ff_input_stream_open(struct mp4ff_input_stream *mis,
struct decoder *decoder,
struct input_stream *input_stream)
{
mis->callback = mpd_mp4ff_callback;
mis->callback.user_data = mis;
mis->decoder = decoder;
mis->input_stream = input_stream;
return mp4ff_open_read(&mis->callback);
}
static faacDecHandle
mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
{
faacDecHandle decoder;
faacDecConfigurationPtr config;
int track;
uint32_t sample_rate;
unsigned char channels;
GError *error = NULL;
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);
track = mp4_get_aac_track(mp4fh, decoder, &sample_rate, &channels);
if (track < 0) {
g_warning("No AAC track found");
faacDecClose(decoder);
return NULL;
}
if (!audio_format_init_checked(audio_format, sample_rate,
SAMPLE_FORMAT_S16, channels,
&error)) {
g_warning("%s", error->message);
g_error_free(error);
faacDecClose(decoder);
return NULL;
}
*track_r = track;
return decoder;
}
static void
mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
{
struct mp4ff_input_stream mis;
mp4ff_t *mp4fh;
int32_t track;
float file_time, total_time;
int32_t scale;
faacDecHandle decoder;
struct audio_format audio_format;
faacDecFrameInfo frame_info;
unsigned char *mp4_buffer;
unsigned int mp4_buffer_size;
long sample_id;
long num_samples;
long dur;
unsigned int sample_count;
char *sample_buffer;
size_t sample_buffer_length;
unsigned int initial = 1;
float *seek_table;
long seek_table_end = -1;
bool seek_position_found = false;
long offset;
uint16_t bit_rate = 0;
bool seeking = false;
double seek_where = 0;
enum decoder_command cmd = DECODE_COMMAND_NONE;
mp4fh = mp4ff_input_stream_open(&mis, mpd_decoder, input_stream);
if (!mp4fh) {
g_warning("Input does not appear to be a mp4 stream.\n");
return;
}
decoder = mp4_faad_new(mp4fh, &track, &audio_format);
if (decoder == NULL) {
mp4ff_close(mp4fh);
return;
}
file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
if (scale < 0) {
g_warning("Error getting audio format of mp4 AAC track.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
return;
}
total_time = ((float)file_time) / scale;
num_samples = mp4ff_num_samples(mp4fh, track);
if (num_samples > (long)(G_MAXINT / sizeof(float))) {
g_warning("Integer overflow.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
return;
}
file_time = 0.0;
seek_table = input_stream->seekable
? g_malloc(sizeof(float) * num_samples)
: NULL;
decoder_initialized(mpd_decoder, &audio_format,
input_stream->seekable,
total_time);
for (sample_id = 0;
sample_id < num_samples && cmd != DECODE_COMMAND_STOP;
sample_id++) {
if (cmd == DECODE_COMMAND_SEEK) {
assert(seek_table != NULL);
seeking = true;
seek_where = decoder_seek_where(mpd_decoder);
}
if (seeking && seek_table_end > 1 &&
seek_table[seek_table_end] >= seek_where) {
int i = 2;
assert(seek_table != NULL);
while (seek_table[i] < seek_where)
i++;
sample_id = i - 1;
file_time = seek_table[sample_id];
}
dur = mp4ff_get_sample_duration(mp4fh, track, sample_id);
offset = mp4ff_get_sample_offset(mp4fh, track, sample_id);
if (seek_table != NULL && sample_id > seek_table_end) {
seek_table[sample_id] = file_time;
seek_table_end = sample_id;
}
if (sample_id == 0)
dur = 0;
if (offset > dur)
dur = 0;
else
dur -= offset;
file_time += ((float)dur) / scale;
if (seeking && file_time >= seek_where)
seek_position_found = true;
if (seeking && seek_position_found) {
seek_position_found = false;
seeking = 0;
decoder_command_finished(mpd_decoder);
}
if (seeking)
continue;
if (mp4ff_read_sample(mp4fh, track, sample_id, &mp4_buffer,
&mp4_buffer_size) == 0)
break;
#ifdef HAVE_FAAD_BUFLEN_FUNCS
sample_buffer = faacDecDecode(decoder, &frame_info, mp4_buffer,
mp4_buffer_size);
#else
sample_buffer = faacDecDecode(decoder, &frame_info, mp4_buffer);
#endif
free(mp4_buffer);
if (frame_info.error > 0) {
g_warning("faad2 error: %s\n",
faacDecGetErrorMessage(frame_info.error));
break;
}
if (frame_info.channels != audio_format.channels) {
g_warning("channel count changed from %u to %u",
audio_format.channels, frame_info.channels);
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
if (frame_info.samplerate != audio_format.sample_rate) {
g_warning("sample rate changed from %u to %lu",
audio_format.sample_rate,
(unsigned long)frame_info.samplerate);
break;
}
#endif
if (audio_format.channels * (unsigned long)(dur + offset) > frame_info.samples) {
dur = frame_info.samples / audio_format.channels;
offset = 0;
}
sample_count = (unsigned long)(dur * audio_format.channels);
if (sample_count > 0) {
initial = 0;
bit_rate = frame_info.bytesconsumed * 8.0 *
frame_info.channels * scale /
frame_info.samples / 1000 + 0.5;
}
sample_buffer_length = sample_count * 2;
sample_buffer += offset * audio_format.channels * 2;
cmd = decoder_data(mpd_decoder, input_stream,
sample_buffer, sample_buffer_length,
bit_rate);
}
g_free(seek_table);
faacDecClose(decoder);
mp4ff_close(mp4fh);
}
static const struct tag_table mp4ff_tags[] = {
{ "album artist", TAG_ALBUM_ARTIST },
{ "writer", TAG_COMPOSER },
{ "band", TAG_PERFORMER },
{ NULL, TAG_NUM_OF_ITEM_TYPES }
};
static enum tag_type
mp4ff_tag_name_parse(const char *name)
{
enum tag_type type = tag_table_lookup_i(mp4ff_tags, name);
if (type == TAG_NUM_OF_ITEM_TYPES)
type = tag_name_parse_i(name);
if (g_ascii_strcasecmp(name, "albumartist") == 0 ||
g_ascii_strcasecmp(name, "album_artist") == 0)
return TAG_ALBUM_ARTIST;
return type;
}
static bool
mp4ff_scan_stream(struct input_stream *is,
const struct tag_handler *handler, void *handler_ctx)
{
struct mp4ff_input_stream mis;
int32_t track;
int32_t file_time;
int32_t scale;
int i;
mp4ff_t *mp4fh = mp4ff_input_stream_open(&mis, NULL, is);
if (mp4fh == NULL)
return false;
track = mp4_get_aac_track(mp4fh, NULL, NULL, NULL);
if (track < 0) {
mp4ff_close(mp4fh);
return false;
}
file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
if (scale < 0) {
mp4ff_close(mp4fh);
return false;
}
tag_handler_invoke_duration(handler, handler_ctx,
((float)file_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);
tag_handler_invoke_pair(handler, handler_ctx, item, value);
enum tag_type type = mp4ff_tag_name_parse(item);
if (type != TAG_NUM_OF_ITEM_TYPES)
tag_handler_invoke_tag(handler, handler_ctx,
type, value);
free(item);
free(value);
}
mp4ff_close(mp4fh);
return true;
}
static const char *const mp4_suffixes[] = {
"m4a",
"m4b",
"mp4",
NULL
};
static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL };
const struct decoder_plugin mp4ff_decoder_plugin = {
.name = "mp4ff",
.stream_decode = mp4_decode,
.scan_stream = mp4ff_scan_stream,
.suffixes = mp4_suffixes,
.mime_types = mp4_mime_types,
};

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@ -40,7 +40,6 @@ extern const struct decoder_plugin mad_decoder_plugin;
extern const struct decoder_plugin mpg123_decoder_plugin;
extern const struct decoder_plugin sndfile_decoder_plugin;
extern const struct decoder_plugin audiofile_decoder_plugin;
extern const struct decoder_plugin mp4ff_decoder_plugin;
extern const struct decoder_plugin faad_decoder_plugin;
extern const struct decoder_plugin mpcdec_decoder_plugin;
extern const struct decoder_plugin modplug_decoder_plugin;
@ -80,9 +79,6 @@ const struct decoder_plugin *const decoder_plugins[] = {
#ifdef HAVE_FAAD
&faad_decoder_plugin,
#endif
#ifdef HAVE_MP4
&mp4ff_decoder_plugin,
#endif
#ifdef HAVE_MPCDEC
&mpcdec_decoder_plugin,
#endif