Mercurial > libavcodec.hg
view aac_parser.c @ 12105:d6e87496883b libavcodec
ARM: set section to .text in 'function' macro
This ensures code always goes into the .text section and avoids the
need to specify it explicitly after changing sections.
author | mru |
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date | Wed, 07 Jul 2010 20:09:41 +0000 |
parents | 61c62ab2218f |
children | ee740a4e80c5 |
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/* * Audio and Video frame extraction * Copyright (c) 2003 Fabrice Bellard * Copyright (c) 2003 Michael Niedermayer * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "parser.h" #include "aac_ac3_parser.h" #include "aac_parser.h" #include "get_bits.h" #include "mpeg4audio.h" int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr) { int size, rdb, ch, sr; int aot, crc_abs; if(get_bits(gbc, 12) != 0xfff) return AAC_AC3_PARSE_ERROR_SYNC; skip_bits1(gbc); /* id */ skip_bits(gbc, 2); /* layer */ crc_abs = get_bits1(gbc); /* protection_absent */ aot = get_bits(gbc, 2); /* profile_objecttype */ sr = get_bits(gbc, 4); /* sample_frequency_index */ if(!ff_mpeg4audio_sample_rates[sr]) return AAC_AC3_PARSE_ERROR_SAMPLE_RATE; skip_bits1(gbc); /* private_bit */ ch = get_bits(gbc, 3); /* channel_configuration */ skip_bits1(gbc); /* original/copy */ skip_bits1(gbc); /* home */ /* adts_variable_header */ skip_bits1(gbc); /* copyright_identification_bit */ skip_bits1(gbc); /* copyright_identification_start */ size = get_bits(gbc, 13); /* aac_frame_length */ if(size < AAC_ADTS_HEADER_SIZE) return AAC_AC3_PARSE_ERROR_FRAME_SIZE; skip_bits(gbc, 11); /* adts_buffer_fullness */ rdb = get_bits(gbc, 2); /* number_of_raw_data_blocks_in_frame */ hdr->object_type = aot + 1; hdr->chan_config = ch; hdr->crc_absent = crc_abs; hdr->num_aac_frames = rdb + 1; hdr->sampling_index = sr; hdr->sample_rate = ff_mpeg4audio_sample_rates[sr]; hdr->samples = (rdb + 1) * 1024; hdr->bit_rate = size * 8 * hdr->sample_rate / hdr->samples; return size; } static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info, int *need_next_header, int *new_frame_start) { GetBitContext bits; AACADTSHeaderInfo hdr; int size; union { uint64_t u64; uint8_t u8[8]; } tmp; tmp.u64 = be2me_64(state); init_get_bits(&bits, tmp.u8+8-AAC_ADTS_HEADER_SIZE, AAC_ADTS_HEADER_SIZE * 8); if ((size = ff_aac_parse_header(&bits, &hdr)) < 0) return 0; *need_next_header = 0; *new_frame_start = 1; hdr_info->sample_rate = hdr.sample_rate; hdr_info->channels = ff_mpeg4audio_channels[hdr.chan_config]; hdr_info->samples = hdr.samples; hdr_info->bit_rate = hdr.bit_rate; return size; } static av_cold int aac_parse_init(AVCodecParserContext *s1) { AACAC3ParseContext *s = s1->priv_data; s->header_size = AAC_ADTS_HEADER_SIZE; s->sync = aac_sync; return 0; } AVCodecParser aac_parser = { { CODEC_ID_AAC }, sizeof(AACAC3ParseContext), aac_parse_init, ff_aac_ac3_parse, ff_parse_close, };