mirror of
https://github.com/FFmpeg/FFmpeg.git
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118 lines
3.6 KiB
C
118 lines
3.6 KiB
C
/*
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* Audio and Video frame extraction
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* Copyright (c) 2003 Fabrice Bellard
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* Copyright (c) 2003 Michael Niedermayer
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "parser.h"
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#include "aac_ac3_parser.h"
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#include "aac_parser.h"
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#include "bitstream.h"
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#include "mpeg4audio.h"
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#define AAC_HEADER_SIZE 7
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int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
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{
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int size, rdb, ch, sr;
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int aot, crc_abs;
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if(get_bits(gbc, 12) != 0xfff)
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return AAC_AC3_PARSE_ERROR_SYNC;
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skip_bits1(gbc); /* id */
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skip_bits(gbc, 2); /* layer */
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crc_abs = get_bits1(gbc); /* protection_absent */
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aot = get_bits(gbc, 2); /* profile_objecttype */
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sr = get_bits(gbc, 4); /* sample_frequency_index */
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if(!ff_mpeg4audio_sample_rates[sr])
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return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
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skip_bits1(gbc); /* private_bit */
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ch = get_bits(gbc, 3); /* channel_configuration */
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if(!ff_mpeg4audio_channels[ch])
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return AAC_AC3_PARSE_ERROR_CHANNEL_CFG;
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skip_bits1(gbc); /* original/copy */
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skip_bits1(gbc); /* home */
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/* adts_variable_header */
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skip_bits1(gbc); /* copyright_identification_bit */
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skip_bits1(gbc); /* copyright_identification_start */
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size = get_bits(gbc, 13); /* aac_frame_length */
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if(size < AAC_HEADER_SIZE)
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return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
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skip_bits(gbc, 11); /* adts_buffer_fullness */
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rdb = get_bits(gbc, 2); /* number_of_raw_data_blocks_in_frame */
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hdr->object_type = aot + 1;
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hdr->chan_config = ch;
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hdr->crc_absent = crc_abs;
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hdr->num_aac_frames = rdb + 1;
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hdr->sampling_index = sr;
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hdr->sample_rate = ff_mpeg4audio_sample_rates[sr];
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hdr->samples = (rdb + 1) * 1024;
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hdr->bit_rate = size * 8 * hdr->sample_rate / hdr->samples;
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return size;
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}
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static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
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int *need_next_header, int *new_frame_start)
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{
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GetBitContext bits;
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AACADTSHeaderInfo hdr;
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int size;
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union {
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uint64_t u64;
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uint8_t u8[8];
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} tmp;
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tmp.u64 = be2me_64(state);
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init_get_bits(&bits, tmp.u8+8-AAC_HEADER_SIZE, AAC_HEADER_SIZE * 8);
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if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
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return 0;
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*need_next_header = 0;
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*new_frame_start = 1;
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hdr_info->sample_rate = hdr.sample_rate;
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hdr_info->channels = ff_mpeg4audio_channels[hdr.chan_config];
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hdr_info->samples = hdr.samples;
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hdr_info->bit_rate = hdr.bit_rate;
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return size;
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}
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static av_cold int aac_parse_init(AVCodecParserContext *s1)
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{
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AACAC3ParseContext *s = s1->priv_data;
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s->header_size = AAC_HEADER_SIZE;
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s->sync = aac_sync;
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return 0;
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}
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AVCodecParser aac_parser = {
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{ CODEC_ID_AAC },
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sizeof(AACAC3ParseContext),
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aac_parse_init,
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ff_aac_ac3_parse,
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ff_parse_close,
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};
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