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Support ADTS AAC files in the ffaac decoder (limited to streams containing one

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raw_data_block() per ADTS frame)

Patch by Alex Converse ( alex converse gmail com) based on a patch by Robert
Swain ( robert swain gmail com )

Originally committed as revision 16485 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Robert Swain
2009-01-07 22:09:21 +00:00
parent 600a331c27
commit 158b39126d
3 changed files with 153 additions and 35 deletions
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49
libavcodec/aac.c
View File

@@ -86,6 +86,7 @@
#include "aactab.h"
#include "aacdectab.h"
#include "mpeg4audio.h"
#include "aac_parser.h"
#include <assert.h>
#include <errno.h>
@@ -384,12 +385,24 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
ac->avccontext = avccontext;
if (avccontext->extradata_size <= 0 ||
decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
if (avccontext->extradata_size > 0) {
if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
return -1;
avccontext->sample_rate = ac->m4ac.sample_rate;
} else if (avccontext->channels > 0) {
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8)))
return -1;
if(output_configure(ac, ac->che_pos, new_che_pos))
return -1;
ac->m4ac.sample_rate = avccontext->sample_rate;
} else {
ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0);
return -1;
}
avccontext->sample_fmt = SAMPLE_FMT_S16;
avccontext->sample_rate = ac->m4ac.sample_rate;
avccontext->frame_size = 1024;
AAC_INIT_VLC_STATIC( 0, 144);
@@ -1506,6 +1519,29 @@ static void spectral_to_sample(AACContext * ac) {
}
}
static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) {
int size;
AACADTSHeaderInfo hdr_info;
size = ff_aac_parse_header(gb, &hdr_info);
if (size > 0) {
if (hdr_info.chan_config)
ac->m4ac.chan_config = hdr_info.chan_config;
ac->m4ac.sample_rate = hdr_info.sample_rate;
ac->m4ac.sampling_index = hdr_info.sampling_index;
ac->m4ac.object_type = hdr_info.object_type;
}
if (hdr_info.num_aac_frames == 1) {
if (!hdr_info.crc_absent)
skip_bits(gb, 16);
} else {
ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
return -1;
}
return size;
}
static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) {
AACContext * ac = avccontext->priv_data;
GetBitContext gb;
@@ -1514,6 +1550,13 @@ static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data
init_get_bits(&gb, buf, buf_size*8);
if (show_bits(&gb, 12) == 0xfff) {
if ((err = parse_adts_frame_header(ac, &gb)) < 0) {
av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
return -1;
}
}
// parse
while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
elem_id = get_bits(&gb, 4);

86
libavcodec/aac_parser.c
View File

@@ -22,16 +22,64 @@
#include "parser.h"
#include "aac_ac3_parser.h"
#include "aac_parser.h"
#include "bitstream.h"
#include "mpeg4audio.h"
#define AAC_HEADER_SIZE 7
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 */
if(!ff_mpeg4audio_channels[ch])
return AAC_AC3_PARSE_ERROR_CHANNEL_CFG;
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_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;
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;
int size, rdb, ch, sr;
AACADTSHeaderInfo hdr;
int size;
union {
uint64_t u64;
uint8_t u8[8];
@@ -40,40 +88,14 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
tmp.u64 = be2me_64(state);
init_get_bits(&bits, tmp.u8+8-AAC_HEADER_SIZE, AAC_HEADER_SIZE * 8);
if(get_bits(&bits, 12) != 0xfff)
if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
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(!ff_mpeg4audio_sample_rates[sr])
return 0;
skip_bits1(&bits); /* private_bit */
ch = get_bits(&bits, 3); /* channel_configuration */
if(!ff_mpeg4audio_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 */
hdr_info->channels = ff_mpeg4audio_channels[ch];
hdr_info->sample_rate = ff_mpeg4audio_sample_rates[sr];
hdr_info->samples = (rdb + 1) * 1024;
hdr_info->bit_rate = size * 8 * hdr_info->sample_rate / hdr_info->samples;
*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;
}

53
libavcodec/aac_parser.h Normal file
View File

@@ -0,0 +1,53 @@
/*
* AAC parser prototypes
* 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
*/
#ifndef AVCODEC_AAC_PARSER_H
#define AVCODEC_AAC_PARSER_H
#include <stdint.h>
#include "aac_ac3_parser.h"
#include "bitstream.h"
typedef struct {
uint32_t sample_rate;
uint32_t samples;
uint32_t bit_rate;
uint8_t crc_absent;
uint8_t object_type;
uint8_t sampling_index;
uint8_t chan_config;
uint8_t num_aac_frames;
} AACADTSHeaderInfo;
/**
* Parses AAC frame header.
* Parses the ADTS frame header to the end of the variable header, which is
* the first 54 bits.
* @param gbc[in] BitContext containing the first 54 bits of the frame.
* @param hdr[out] Pointer to struct where header info is written.
* @return Returns 0 on success, -1 if there is a sync word mismatch,
* -2 if the version element is invalid, -3 if the sample rate
* element is invalid, or -4 if the bit rate element is invalid.
*/
int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr);
#endif /* AVCODEC_AAC_PARSER_H */