/* 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 "config.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 mp4_context { 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 mp4_context *ctx = user_data; return decoder_read(ctx->decoder, ctx->input_stream, buffer, length); } static uint32_t mp4_seek(void *user_data, uint64_t position) { struct mp4_context *ctx = user_data; return input_stream_seek(ctx->input_stream, position, SEEK_SET) ? 0 : -1; } 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; 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; } *track_r = track; *audio_format = (struct audio_format){ .bits = 16, .channels = channels, .sample_rate = sample_rate, }; if (!audio_format_valid(audio_format)) { g_warning("Invalid audio format: %u:%u:%u\n", audio_format->sample_rate, audio_format->bits, audio_format->channels); faacDecClose(decoder); return NULL; } return decoder; } static void mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) { struct mp4_context ctx = { .decoder = mpd_decoder, .input_stream = input_stream, }; mp4ff_callback_t callback = { .read = mp4_read, .seek = mp4_seek, .user_data = &ctx, }; 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_open_read(&callback); 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, file_time, bit_rate, NULL); } g_free(seek_table); faacDecClose(decoder); mp4ff_close(mp4fh); } static struct tag * mp4_tag_dup(const char *file) { struct tag *ret = NULL; struct input_stream input_stream; struct mp4_context ctx = { .decoder = NULL, .input_stream = &input_stream, }; mp4ff_callback_t callback = { .read = mp4_read, .seek = mp4_seek, .user_data = &ctx, }; mp4ff_t *mp4fh; int32_t track; int32_t file_time; int32_t scale; int i; if (!input_stream_open(&input_stream, file)) { g_warning("Failed to open file: %s", file); return NULL; } mp4fh = mp4ff_open_read(&callback); if (!mp4fh) { input_stream_close(&input_stream); return NULL; } track = mp4_get_aac_track(mp4fh, NULL, NULL, NULL); if (track < 0) { mp4ff_close(mp4fh); input_stream_close(&input_stream); return NULL; } ret = tag_new(); file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track); scale = mp4ff_time_scale(mp4fh, track); if (scale < 0) { mp4ff_close(mp4fh); input_stream_close(&input_stream); tag_free(ret); return NULL; } ret->time = ((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); if (0 == strcasecmp("artist", item)) { tag_add_item(ret, TAG_ITEM_ARTIST, value); } else if (0 == strcasecmp("title", item)) { tag_add_item(ret, TAG_ITEM_TITLE, value); } else if (0 == strcasecmp("album", item)) { tag_add_item(ret, TAG_ITEM_ALBUM, value); } else if (0 == strcasecmp("track", item)) { tag_add_item(ret, TAG_ITEM_TRACK, value); } else if (0 == strcasecmp("disc", item)) { /* Is that the correct id? */ tag_add_item(ret, TAG_ITEM_DISC, value); } else if (0 == strcasecmp("genre", item)) { tag_add_item(ret, TAG_ITEM_GENRE, value); } else if (0 == strcasecmp("date", item)) { tag_add_item(ret, TAG_ITEM_DATE, value); } else if (0 == strcasecmp("writer", item)) { tag_add_item(ret, TAG_ITEM_COMPOSER, value); } free(item); free(value); } mp4ff_close(mp4fh); input_stream_close(&input_stream); return ret; } static const char *const mp4_suffixes[] = { "m4a", "mp4", NULL }; static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL }; const struct decoder_plugin mp4ff_decoder_plugin = { .name = "mp4", .stream_decode = mp4_decode, .tag_dup = mp4_tag_dup, .suffixes = mp4_suffixes, .mime_types = mp4_mime_types, };