一 命令行
ffmpeg -i k.mp4 -acodec copy -vn out.aac
就不解释了前面解释太多了
####C 实现
av_init_packet()
初始化数据包结构体接收抽取出来的数据
av_find_best_steam()
找到最佳的流
av_read_frame()/ 读取 包
av_packet_unref()// 释放包
说一下为啥是读取包 但是是frame
首先基本单位从大到小是 流 包 帧
FFmpeg刚刚开始 读取是一帧一帧的 但是后来因为压缩后帧和未压缩帧大小不同的关系的关系 实际读取改为了包 包里面都是几个的整数帧
int main(int argc, char *argv[])
{
int err_code;
char errors[1024];
char *src_filename = NULL;
char *dst_filename = NULL;
FILE *dst_fd = NULL;
int audio_stream_index = -1;
int len;
AVFormatContext *ofmt_ctx = NULL;
AVOutputFormat *output_fmt = NULL;
AVStream *out_stream = NULL;
AVFormatContext *fmt_ctx = NULL;
AVFrame *frame = NULL;
AVPacket pkt;
av_log_set_level(AV_LOG_DEBUG);
// 这里是控制台输入
if(argc < 3){
av_log(NULL, AV_LOG_DEBUG, "the count of parameters should be more than three!\n");
return -1;
}
src_filename = argv[1];
dst_filename = argv[2];
if(src_filename == NULL || dst_filename == NULL){
av_log(NULL, AV_LOG_DEBUG, "src or dts file is null, plz check them!\n");
return -1;
}
dst_fd = fopen(dst_filename, "wb");
if (!dst_fd) {
av_log(NULL, AV_LOG_DEBUG, "Could not open destination file %s\n", dst_filename);
return -1;
}
/*open input media file, and allocate format context*/
if((err_code = avformat_open_input(&fmt_ctx, src_filename, NULL, NULL)) < 0){
av_strerror(err_code, errors, 1024);
av_log(NULL, AV_LOG_DEBUG, "Could not open source file: %s, %d(%s)\n",
src_filename,
err_code,
errors);
return -1;
}
/*retrieve audio stream*/
if((err_code = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_strerror(err_code, errors, 1024);
av_log(NULL, AV_LOG_DEBUG, "failed to find stream information: %s, %d(%s)\n",
src_filename,
err_code,
errors);
return -1;
}
/*dump input information*/
av_dump_format(fmt_ctx, 0, src_filename, 0);
frame = av_frame_alloc();
if(!frame){
av_log(NULL, AV_LOG_DEBUG, "Could not allocate frame\n");
return AVERROR(ENOMEM);
}
/*initialize packet*/
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
/*find best audio stream*/
// 只选择音频 AVMEDIA_TYPE_AUDIO
audio_stream_index = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
if(audio_stream_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),
src_filename);
return AVERROR(EINVAL);
}
/*
#define FF_PROFILE_AAC_MAIN 0
#define FF_PROFILE_AAC_LOW 1
#define FF_PROFILE_AAC_SSR 2
#define FF_PROFILE_AAC_LTP 3
#define FF_PROFILE_AAC_HE 4
#define FF_PROFILE_AAC_HE_V2 28
#define FF_PROFILE_AAC_LD 22
#define FF_PROFILE_AAC_ELD 38
#define FF_PROFILE_MPEG2_AAC_LOW 128
#define FF_PROFILE_MPEG2_AAC_HE 131
*/
// 定义音频基本参数 profile 选择和通道数 采样率
int aac_type = fmt_ctx->streams[1]->codecpar->profile;
int channels = fmt_ctx->streams[1]->codecpar->channels;
int sample_rate= fmt_ctx->streams[1]->codecpar->sample_rate;
if(fmt_ctx->streams[1]->codecpar->codec_id != AV_CODEC_ID_AAC){
av_log(NULL, AV_LOG_ERROR, "the audio type is not AAC!\n");
goto __ERROR;
}else{
av_log(NULL, AV_LOG_INFO, "the audio type is AAC!\n");
}
/*read frames from media file*/
while(av_read_frame(fmt_ctx, &pkt) >=0 ){
//判断还不是找到的那路最佳流
if(pkt.stream_index == audio_stream_index){
char adts_header_buf[7];
adts_header(adts_header_buf, pkt.size, aac_type, sample_rate, channels);
fwrite(adts_header_buf, 1, 7, dst_fd);
len = fwrite( pkt.data, 1, pkt.size, dst_fd);
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);
}
__ERROR:
/*close input media file*/
avformat_close_input(&fmt_ctx);
if(dst_fd) {
fclose(dst_fd);
}
return 0;
}
#include <stdio.h>
#include <libavutil/log.h>
#include <libavformat/avio.h>
#include <libavformat/avformat.h>
#define ADTS_HEADER_LEN 7;
static int get_audio_obj_type(int aactype){
//AAC HE V2 = AAC LC + SBR + PS
//AAV HE = AAC LC + SBR
//所以无论是 AAC_HEv2 还是 AAC_HE 都是 AAC_LC
switch(aactype){
case 0:
case 2:
case 3:
return aactype+1;
case 1:
case 4:
case 28:
return 2;
default:
return 2;
}
}
static int get_sample_rate_index(int freq, int aactype){
int i = 0;
int freq_arr[13] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000, 7350
};
//如果是 AAC HEv2 或 AAC HE, 则频率减半
if(aactype == 28 || aactype == 4){
freq /= 2;
}
for(i=0; i< 13; i++){
if(freq == freq_arr[i]){
return i;
}
}
return 4;//默认是44100
}
static int get_channel_config(int channels, int aactype){
//如果是 AAC HEv2 通道数减半
if(aactype == 28){
return (channels / 2);
}
return channels;
}
static void adts_header(char *szAdtsHeader, int dataLen, int aactype, int frequency, int channels){
int audio_object_type = get_audio_obj_type(aactype);
int sampling_frequency_index = get_sample_rate_index(frequency, aactype);
int channel_config = get_channel_config(channels, aactype);
printf("aot=%d, freq_index=%d, channel=%d\n", audio_object_type, sampling_frequency_index, channel_config);
int adtsLen = dataLen + 7;
szAdtsHeader[0] = 0xff; //syncword:0xfff 高8bits
szAdtsHeader[1] = 0xf0; //syncword:0xfff 低4bits
szAdtsHeader[1] |= (0 << 3); //MPEG Version:0 for MPEG-4,1 for MPEG-2 1bit
szAdtsHeader[1] |= (0 << 1); //Layer:0 2bits
szAdtsHeader[1] |= 1; //protection absent:1 1bit
szAdtsHeader[2] = (audio_object_type - 1)<<6; //profile:audio_object_type - 1 2bits
szAdtsHeader[2] |= (sampling_frequency_index & 0x0f)<<2; //sampling frequency index:sampling_frequency_index 4bits
szAdtsHeader[2] |= (0 << 1); //private bit:0 1bit
szAdtsHeader[2] |= (channel_config & 0x04)>>2; //channel configuration:channel_config 高1bit
szAdtsHeader[3] = (channel_config & 0x03)<<6; //channel configuration:channel_config 低2bits
szAdtsHeader[3] |= (0 << 5); //original:0 1bit
szAdtsHeader[3] |= (0 << 4); //home:0 1bit
szAdtsHeader[3] |= (0 << 3); //copyright id bit:0 1bit
szAdtsHeader[3] |= (0 << 2); //copyright id start:0 1bit
szAdtsHeader[3] |= ((adtsLen & 0x1800) >> 11); //frame length:value 高2bits
szAdtsHeader[4] = (uint8_t)((adtsLen & 0x7f8) >> 3); //frame length:value 中间8bits
szAdtsHeader[5] = (uint8_t)((adtsLen & 0x7) << 5); //frame length:value 低3bits
szAdtsHeader[5] |= 0x1f; //buffer fullness:0x7ff 高5bits
szAdtsHeader[6] = 0xfc;
}
int main(int argc, char *argv[])
{
int err_code;
char errors[1024];
char *src_filename = NULL;
char *dst_filename = NULL;
FILE *dst_fd = NULL;
int audio_stream_index = -1;
int len;
AVFormatContext *ofmt_ctx = NULL;
AVOutputFormat *output_fmt = NULL;
AVStream *out_stream = NULL;
AVFormatContext *fmt_ctx = NULL;
AVFrame *frame = NULL;
AVPacket pkt;
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;
}
src_filename = argv[1];
dst_filename = argv[2];
if(src_filename == NULL || dst_filename == NULL){
av_log(NULL, AV_LOG_DEBUG, "src or dts file is null, plz check them!\n");
return -1;
}
dst_fd = fopen(dst_filename, "wb");
if (!dst_fd) {
av_log(NULL, AV_LOG_DEBUG, "Could not open destination file %s\n", dst_filename);
return -1;
}
/*open input media file, and allocate format context*/
if((err_code = avformat_open_input(&fmt_ctx, src_filename, NULL, NULL)) < 0){
av_strerror(err_code, errors, 1024);
av_log(NULL, AV_LOG_DEBUG, "Could not open source file: %s, %d(%s)\n",
src_filename,
err_code,
errors);
return -1;
}
/*retrieve audio stream*/
if((err_code = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_strerror(err_code, errors, 1024);
av_log(NULL, AV_LOG_DEBUG, "failed to find stream information: %s, %d(%s)\n",
src_filename,
err_code,
errors);
return -1;
}
/*dump input information*/
av_dump_format(fmt_ctx, 0, src_filename, 0);
frame = av_frame_alloc();
if(!frame){
av_log(NULL, AV_LOG_DEBUG, "Could not allocate frame\n");
return AVERROR(ENOMEM);
}
/*initialize packet*/
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
/*find best audio stream*/
audio_stream_index = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
if(audio_stream_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),
src_filename);
return AVERROR(EINVAL);
}
/*
#define FF_PROFILE_AAC_MAIN 0
#define FF_PROFILE_AAC_LOW 1
#define FF_PROFILE_AAC_SSR 2
#define FF_PROFILE_AAC_LTP 3
#define FF_PROFILE_AAC_HE 4
#define FF_PROFILE_AAC_HE_V2 28
#define FF_PROFILE_AAC_LD 22
#define FF_PROFILE_AAC_ELD 38
#define FF_PROFILE_MPEG2_AAC_LOW 128
#define FF_PROFILE_MPEG2_AAC_HE 131
*/
int aac_type = fmt_ctx->streams[1]->codecpar->profile;
int channels = fmt_ctx->streams[1]->codecpar->channels;
int sample_rate= fmt_ctx->streams[1]->codecpar->sample_rate;
if(fmt_ctx->streams[1]->codecpar->codec_id != AV_CODEC_ID_AAC){
av_log(NULL, AV_LOG_ERROR, "the audio type is not AAC!\n");
goto __ERROR;
}else{
av_log(NULL, AV_LOG_INFO, "the audio type is AAC!\n");
}
/*read frames from media file*/
while(av_read_frame(fmt_ctx, &pkt) >=0 ){
if(pkt.stream_index == audio_stream_index){
char adts_header_buf[7];
adts_header(adts_header_buf, pkt.size, aac_type, sample_rate, channels);
fwrite(adts_header_buf, 1, 7, dst_fd);
len = fwrite( pkt.data, 1, pkt.size, dst_fd);
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);
}
__ERROR:
/*close input media file*/
avformat_close_input(&fmt_ctx);
if(dst_fd) {
fclose(dst_fd);
}
return 0;
}