From 99ff31dc750e8ac562b529c3ec374cfbb3e66157 Mon Sep 17 00:00:00 2001 From: Aurelien Jacobs Date: Tue, 8 May 2007 23:25:31 +0000 Subject: move aac and ac3 parsers in their own files Originally committed as revision 8941 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/aac_parser.c | 97 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 97 insertions(+) create mode 100644 libavcodec/aac_parser.c (limited to 'libavcodec/aac_parser.c') diff --git a/libavcodec/aac_parser.c b/libavcodec/aac_parser.c new file mode 100644 index 0000000000..3962289917 --- /dev/null +++ b/libavcodec/aac_parser.c @@ -0,0 +1,97 @@ +/* + * 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 */ + 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) +{ + AC3ParseContext *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(AC3ParseContext), + aac_parse_init, + ac3_parse, + NULL, +}; -- cgit v1.2.3