diff options
| -rw-r--r-- | INSTALL | 2 | ||||
| -rw-r--r-- | Makefile.am | 5 | ||||
| -rw-r--r-- | NEWS | 1 | ||||
| -rw-r--r-- | configure.ac | 3 | ||||
| -rw-r--r-- | m4/faad.m4 | 34 | ||||
| -rw-r--r-- | src/decoder/mp4ff_decoder_plugin.c | 448 | ||||
| -rw-r--r-- | src/decoder_list.c | 4 |
7 files changed, 2 insertions, 495 deletions
@@ -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. diff --git a/Makefile.am b/Makefile.am index cf023a6b..4a1dfea7 100644 --- a/Makefile.am +++ b/Makefile.am @@ -499,7 +499,6 @@ DECODER_LIBS = \ $(MAD_LIBS) \ $(MPG123_LIBS) \ $(OPUS_LIBS) \ - $(MP4FF_LIBS) \ $(FFMPEG_LIBS) \ $(MPCDEC_LIBS) \ $(ADPLUG_LIBS) \ @@ -549,10 +548,6 @@ if HAVE_FAAD libdecoder_plugins_a_SOURCES += src/decoder/faad_decoder_plugin.c endif -if HAVE_MP4 -libdecoder_plugins_a_SOURCES += src/decoder/mp4ff_decoder_plugin.c -endif - if HAVE_XIPH libdecoder_plugins_a_SOURCES += \ src/decoder/XiphTags.c src/decoder/XiphTags.h \ @@ -5,6 +5,7 @@ ver 0.18 (2012/??/??) - flac: support FLAC files inside archives - opus: new decoder plugin for the Opus codec - vorbis: skip 16 bit quantisation, provide float samples + - mp4ff: obsolete plugin removed * encoder: - opus: new encoder plugin for the Opus codec - vorbis: accept floating point input samples diff --git a/configure.ac b/configure.ac index 073141d7..f6bb64a8 100644 --- a/configure.ac +++ b/configure.ac @@ -872,7 +872,6 @@ dnl ----------------------------------- FAAD ---------------------------------- AM_PATH_FAAD() AM_CONDITIONAL(HAVE_FAAD, test x$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]) @@ -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) diff --git a/src/decoder/mp4ff_decoder_plugin.c b/src/decoder/mp4ff_decoder_plugin.c deleted file mode 100644 index ca78a22d..00000000 --- a/src/decoder/mp4ff_decoder_plugin.c +++ /dev/null @@ -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, -}; diff --git a/src/decoder_list.c b/src/decoder_list.c index faf71f77..80f6db1f 100644 --- a/src/decoder_list.c +++ b/src/decoder_list.c @@ -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 |