音视频流媒体开发【八】-AAC ADTS格式分析
AAC⾳频格式：Advanced Audio Coding(⾼级⾳频解码)，是⼀种由MPEG-4标准定义的有损⾳频压缩格式，由Fraunhofer发展，Dolby, Sony和AT&T是主要的贡献者。
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ADIF：Audio Data Interchange Format ⾳频数据交换格式。这种格式的特征是可以确定的找到这个⾳频数据的开始，不需进⾏在⾳频数据流中间开始的解码，即它的解码必须在明确定义的开始处进⾏。故这种格式常⽤在磁盘⽂件中。
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ADTS的全称是Audio Data Transport Stream。是AAC⾳频的传输流格式。AAC⾳频格式在MPEG-2（ISO-13318-7 2003）中有定义。AAC后来⼜被采⽤到MPEG-4标准中。这种格式的特征是它是⼀个有同步字的⽐特流，解码可以在这个流中任何位置开始。它的特征类似于mp3数据流格式。
简单说，ADTS可以在任意帧解码，也就是说它每⼀帧都有头信息。ADIF只有⼀个统⼀的头，所以必须得到所有的数据后解码。
且这两种的header的格式也是不同的，⽬前⼀般编码后的和抽取出的都是ADTS格式的⾳频流。两者具体的组织结构如下所示：
- AAC的ADIF格式⻅下图：
- AAC的ADTS的⼀般格式⻅下图：
空⽩处表示前后帧
有的时候当你编码AAC裸流的时候，会遇到写出来的AAC⽂件并不能在PC和⼿机上播放，很⼤的可能就是AAC⽂件的每⼀帧⾥缺少了ADTS头信息⽂件的包装拼接。
只需要加⼊头⽂件ADTS即可。⼀个AAC原始数据块⻓度是可变的，对原始帧加上ADTS头进⾏ADTS的封装，就形成了ADTS帧。
AAC⾳频⽂件的每⼀帧由ADTS Header和AAC Audio Data组成。结构体如下：
每⼀帧的ADTS的头⽂件都包含了⾳频的采样率，声道，帧⻓度等信息，这样解码器才能解析读取。
⼀般情况下ADTS的头信息都是7个字节，分为2部分：
- adts_fixed_header();
- adts_variable_header();
其⼀为固定头信息，紧接着是可变头信息。固定头信息中的数据每⼀帧都相同，⽽可变头信息则在帧与帧之间可变。
syncword ：同步头 总是0xFFF, all bits must be 1，代表着⼀个ADTS帧的开始
ID：MPEG标识符，0标识MPEG-4，1标识MPEG-2
Layer：always: '00'
protection_absent：表示是否误码校验。Warning, set to 1 if there is no CRC and 0 if there is CRC
profile：表示使⽤哪个级别的AAC，如01 Low Complexity(LC)--- AAC LC。有些芯⽚只⽀持AAC LC 。
在MPEG-2 AAC中定义了3种：
profile的值等于 Audio Object Type的值减1
profile = MPEG-4 Audio Object Type - 1
sampling_frequency_index：表示使⽤的采样率下标，通过这个下标在Sampling Frequencies[ ]数组中查找得知采样率的值。
channel_configuration:表示声道数，⽐如2表示⽴体声双声道
0: Defined in AOT Specifc Config
1: 1 channel: front-center
2: 2 channels: front-left, front-right
3: 3 channels: front-center, front-left, front-right
4: 4 channels: front-center, front-left, front-right, back-center
5: 5 channels: front-center, front-left, front-right, back-left, backright
6: 6 channels: front-center, front-left, front-right, back-left, backright, LFE-channel
7: 8 channels: front-center, front-left, front-right, side-left, side-right, back-left, back-right, LFE-channel
8-15: Reserved
接下来看下adts_variable_header();
frame_length : ⼀个ADTS帧的⻓度包括ADTS头和AAC原始流.
frame length, this value must include 7 or 9 bytes of header length:
aac_frame_length = (protection_absent == 1 ? 7 : 9) + size(AACFrame)
protection_absent=0时, header length=9bytes
protection_absent=1时, header length=7bytes
adts_buffer_fullness：0x7FF 说明是码率可变的码流。
number_of_raw_data_blocks_in_frame：表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧。
所以说number_of_raw_data_blocks_in_frame == 0 表示说ADTS帧中有⼀个AAC数据块。
下⾯是ADTS的AAC⽂件部分：
⾼字节开始算
第⼀帧的帧头7个字节为：0xFF 0xF1 0x4C 0x40 0x20 0xFF 0xFC
分析各个关键数值：
111111111111
0
00
1
01
0011
0
001
0
0
0
0
0000100000111(帧⻓度)
11111111111
00
计算帧⻓度：将⼆进制 0000100000111 转换成⼗进制为263。观察第⼀帧的⻓度确实为263个字节。
计算⽅法:（帧⻓度为13位，使⽤unsigned int来存储帧⻓数值）
unsigned int getFrameLength(unsigned char* str)
{
if ( !str )
{
return 0;
}
unsigned int len = 0;
int f_bit = str[3];
int m_bit = str[4];
int b_bit = str[5];
len += (b_bit>>5);
len += (m_bit<<3);
len += ((f_bit&3)<<11);
return len;
}
案例 extract-aac
.pro
TEMPLATE = app
CONFIG += console
CONFIG -= app_bundle
CONFIG -= qt
SOURCES += main.c
win32 {
INCLUDEPATH += $$PWD/ffmpeg-4.2.1-win32-dev/include
LIBS += $$PWD/ffmpeg-4.2.1-win32-dev/lib/avformat.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/avcodec.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/avdevice.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/avfilter.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/avutil.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/postproc.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/swresample.lib \
$$PWD/ffmpeg-4.2.1-win32-dev/lib/swscale.lib
}
main.c
#include <stdio.h>
#include <libavutil/log.h>
#include <libavformat/avio.h>
#include <libavformat/avformat.h>
#define ADTS_HEADER_LEN 7;
const int sampling_frequencies[] = {
96000, // 0x0
88200, // 0x1
64000, // 0x2
48000, // 0x3
44100, // 0x4
32000, // 0x5
24000, // 0x6
22050, // 0x7
16000, // 0x8
12000, // 0x9
11025, // 0xa
8000 // 0xb
// 0xc d e f是保留的
};
int adts_header(char * const p_adts_header, const int data_length,
const int profile, const int samplerate,
const int channels)
{
int sampling_frequency_index = 3; // 默认使用48000hz
int adtsLen = data_length + 7;
int frequencies_size = sizeof(sampling_frequencies) / sizeof(sampling_frequencies[0]);
int i = 0;
for(i = 0; i < frequencies_size; i++)
{
if(sampling_frequencies[i] == samplerate)
{
sampling_frequency_index = i;
break;
}
}
if(i >= frequencies_size)
{
printf("unsupport samplerate:%d\n", samplerate);
return -1;
}
p_adts_header[0] = 0xff; //syncword:0xfff 高8bits
p_adts_header[1] = 0xf0; //syncword:0xfff 低4bits
p_adts_header[1] |= (0 << 3); //MPEG Version:0 for MPEG-4,1 for MPEG-2 1bit
p_adts_header[1] |= (0 << 1); //Layer:0 2bits
p_adts_header[1] |= 1; //protection absent:1 1bit
p_adts_header[2] = (profile)<<6; //profile:profile 2bits
p_adts_header[2] |= (sampling_frequency_index & 0x0f)<<2; //sampling frequency index:sampling_frequency_index 4bits
p_adts_header[2] |= (0 << 1); //private bit:0 1bit
p_adts_header[2] |= (channels & 0x04)>>2; //channel configuration:channels 高1bit
p_adts_header[3] = (channels & 0x03)<<6; //channel configuration:channels 低2bits
p_adts_header[3] |= (0 << 5); //original：0 1bit
p_adts_header[3] |= (0 << 4); //home：0 1bit
p_adts_header[3] |= (0 << 3); //copyright id bit：0 1bit
p_adts_header[3] |= (0 << 2); //copyright id start：0 1bit
p_adts_header[3] |= ((adtsLen & 0x1800) >> 11); //frame length：value 高2bits
p_adts_header[4] = (uint8_t)((adtsLen & 0x7f8) >> 3); //frame length:value 中间8bits
p_adts_header[5] = (uint8_t)((adtsLen & 0x7) << 5); //frame length:value 低3bits
p_adts_header[5] |= 0x1f; //buffer fullness:0x7ff 高5bits
p_adts_header[6] = 0xfc; //11111100 //buffer fullness:0x7ff 低6bits
// number_of_raw_data_blocks_in_frame：
// 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧。
return 0;
}
int main(int argc, char *argv[])
{
int ret = -1;
char errors[1024];
char *in_filename = NULL;
char *aac_filename = NULL;
FILE *aac_fd = NULL;
int audio_index = -1;
int len = 0;
AVFormatContext *ifmt_ctx = NULL;
AVPacket pkt;
// 设置打印级别
av_log_set_level(AV_LOG_DEBUG);
if(argc < 3)
{
av_log(NULL, AV_LOG_DEBUG, "the count of parameters should be more than three!\n");
return -1;
}
in_filename = argv[1]; // 输入文件
aac_filename = argv[2]; // 输出文件
if(in_filename == NULL || aac_filename == NULL)
{
av_log(NULL, AV_LOG_DEBUG, "src or dts file is null, plz check them!\n");
return -1;
}
aac_fd = fopen(aac_filename, "wb");
if (!aac_fd)
{
av_log(NULL, AV_LOG_DEBUG, "Could not open destination file %s\n", aac_filename);
return -1;
}
// 打开输入文件
if((ret = avformat_open_input(&ifmt_ctx, in_filename, NULL, NULL)) < 0)
{
av_strerror(ret, errors, 1024);
av_log(NULL, AV_LOG_DEBUG, "Could not open source file: %s, %d(%s)\n",
in_filename,
ret,
errors);
return -1;
}
// 获取解码器信息
if((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0)
{
av_strerror(ret, errors, 1024);
av_log(NULL, AV_LOG_DEBUG, "failed to find stream information: %s, %d(%s)\n",
in_filename,
ret,
errors);
return -1;
}
// dump媒体信息
av_dump_format(ifmt_ctx, 0, in_filename, 0);
// 初始化packet
av_init_packet(&pkt);
// 查找audio对应的steam index
audio_index = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
if(audio_index < 0)
{
av_log(NULL, AV_LOG_DEBUG, "Could not find %s stream in input file %s\n",
av_get_media_type_string(AVMEDIA_TYPE_AUDIO),
in_filename);
return AVERROR(EINVAL);
}
// 打印AAC级别
printf("audio profile:%d, FF_PROFILE_AAC_LOW:%d\n",
ifmt_ctx->streams[audio_index]->codecpar->profile,
FF_PROFILE_AAC_LOW);
if(ifmt_ctx->streams[audio_index]->codecpar->codec_id != AV_CODEC_ID_AAC)
{
printf("the media file no contain AAC stream, it's codec_id is %d\n",
ifmt_ctx->streams[audio_index]->codecpar->codec_id);
goto failed;
}
// 读取媒体文件，并把aac数据帧写入到本地文件
while(av_read_frame(ifmt_ctx, &pkt) >=0 )
{
if(pkt.stream_index == audio_index)
{
char adts_header_buf[7] = {0};
adts_header(adts_header_buf, pkt.size,
ifmt_ctx->streams[audio_index]->codecpar->profile,
ifmt_ctx->streams[audio_index]->codecpar->sample_rate,
ifmt_ctx->streams[audio_index]->codecpar->channels);
fwrite(adts_header_buf, 1, 7, aac_fd); // 写adts header , ts流不适用，ts流分离出来的packet带了adts header
len = fwrite( pkt.data, 1, pkt.size, aac_fd); // 写adts data
if(len != pkt.size)
{
av_log(NULL, AV_LOG_DEBUG, "warning, length of writed data isn't equal pkt.size(%d, %d)\n",
len,
pkt.size);
}
}
av_packet_unref(&pkt);
}
failed:
// 关闭输入文件
if(ifmt_ctx)
{
avformat_close_input(&ifmt_ctx);
}
if(aac_fd)
{
fclose(aac_fd);
}
return 0;
}
添加believe.mp4文件到build-07-02-extract-aac-Desktop_Qt_5_10_1_MinGW_32bit-Debug目录
设置
运行
[NULL @ 00fecf00] Opening 'believe.mp4' for reading
[file @ 00fd5200] Setting default whitelist 'file,crypto'
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] ISO: File Type Major Brand: isom
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] Processing st: 0, edit list 0 - media time: 0, duration: 3414021
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] Processing st: 1, edit list 0 - media time: 678, duration: 10680624
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] skip 678 audio samples from curr_cts: 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] Before avformat_find_stream_info() pos: 7825770 bytes read:127924 seeks:1 nb_streams:2
[h264 @ 00fe20c0] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 8(PPS), nal_ref_idc: 3
[aac @ 00ff4a80] skip 678 / discard 0 samples due to side data
[aac @ 00ff4a80] skip 678/1024 samples
[h264 @ 00fe20c0] nal_unit_type: 6(SEI
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 00fe20c0] Format yuv420p chosen by get_format()
[h264 @ 00fe20c0] Reinit context to 1920x1088, pix_fmt:
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] All info found
[mov,mp4,m4a,3gp,3g2,mj2 @ 00fecf00] After avformat
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'believe.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf56.38.102
comment : www.ieway.cn
Duration: 00:03:42.53, start: 0.000000, bitrate: 281 kb/s
Stream #0:0(und), 1, 1/15360: Video: h264 (Constrained Baseline), 1 reference frame (avc1 / 0x31637661), yuv420p(left), 1920x1080 (1920x1088), 0/1, 150 kb/s, 14.46 fps, 15 tbr, 15360 tbn, 30 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und), 1, 1/48000: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
audio profile:1, FF_PROFILE_AAC_LOW:1
[AVIOContext @ 0111e5c0] Statistics: 7869163 bytes read, 2 seeks
