Mercurial > libavcodec.hg
view aac_parser.c @ 6196:166bef5cad01 libavcodec
add parenthesis, fix warning: i386/dsputil_mmx.c:2618: warning: suggest parentheses around arithmetic in operand of |
author | bcoudurier |
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date | Wed, 30 Jan 2008 23:54:59 +0000 |
parents | ced30500e2b1 |
children | 48759bfbd073 |
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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 "bitstream.h" #define AAC_HEADER_SIZE 7 static const int aac_sample_rates[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350 }; static const int aac_channels[8] = { 0, 1, 2, 3, 4, 5, 6, 8 }; static int aac_sync(const uint8_t *buf, int *channels, int *sample_rate, int *bit_rate, int *samples) { GetBitContext bits; int size, rdb, ch, sr; init_get_bits(&bits, buf, AAC_HEADER_SIZE * 8); if(get_bits(&bits, 12) != 0xfff) return 0; skip_bits1(&bits); /* id */ skip_bits(&bits, 2); /* layer */ skip_bits1(&bits); /* protection_absent */ skip_bits(&bits, 2); /* profile_objecttype */ sr = get_bits(&bits, 4); /* sample_frequency_index */ if(!aac_sample_rates[sr]) return 0; skip_bits1(&bits); /* private_bit */ ch = get_bits(&bits, 3); /* channel_configuration */ if(!aac_channels[ch]) return 0; skip_bits1(&bits); /* original/copy */ skip_bits1(&bits); /* home */ /* adts_variable_header */ skip_bits1(&bits); /* copyright_identification_bit */ skip_bits1(&bits); /* copyright_identification_start */ size = get_bits(&bits, 13); /* aac_frame_length */ if(size < AAC_HEADER_SIZE) return 0; skip_bits(&bits, 11); /* adts_buffer_fullness */ rdb = get_bits(&bits, 2); /* number_of_raw_data_blocks_in_frame */ *channels = aac_channels[ch]; *sample_rate = aac_sample_rates[sr]; *samples = (rdb + 1) * 1024; *bit_rate = size * 8 * *sample_rate / *samples; return size; } static int aac_parse_init(AVCodecParserContext *s1) { AACAC3ParseContext *s = s1->priv_data; s->inbuf_ptr = s->inbuf; s->header_size = AAC_HEADER_SIZE; s->sync = aac_sync; return 0; } AVCodecParser aac_parser = { { CODEC_ID_AAC }, sizeof(AACAC3ParseContext), aac_parse_init, ff_aac_ac3_parse, NULL, };