ffmpeg解碼音訊的兩種方式(二)根據同步位元組解析音訊幀
阿新 • • 發佈:2019-02-18
根據adts同步頭提取aac音訊單幀:
#include "stdafx.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"
//SDL
#include "sdl/SDL.h"
#include "sdl/SDL_thread.h"
};
#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
#define OUTPUT_PCM 1
#define USE_SDL 1
static Uint8 *audio_chunk;
static Uint32 audio_len;
static Uint8 *audio_pos;
void fill_audio(void *udata,Uint8 *stream,int len){
//SDL 2.0
SDL_memset(stream, 0, len);
if(audio_len==0) /* Only play if we have data left */
return;
len=(len>audio_len?audio_len:len); /* Mix as much data as possible */
SDL_MixAudio(stream,audio_pos,len,SDL_MIX_MAXVOLUME);
audio_pos += len;
audio_len -= len;
}
//-----------------
int getADTSframe(unsigned char* buffer, int buf_size, unsigned char* data ,int* data_size){
int size = 0;
if(!buffer || !data || !data_size ){
return -1;
}
while(1){
if(buf_size < 7 ){
return -1;
}
//Sync words
if((buffer[0] == 0xff) && ((buffer[1] & 0xf0) == 0xf0) ){
size |= ((buffer[3] & 0x03) <<11); //high 2 bit
size |= buffer[4]<<3; //middle 8 bit
size |= ((buffer[5] & 0xe0)>>5); //low 3bit
break;
}
--buf_size;
++buffer;
}
if(buf_size < size){
return 1;
}
memcpy(data, buffer, size);
*data_size = size;
return 0;
}
int nWriteBytes=0;
int _tmain(int argc, _TCHAR* argv[])
{
AVFormatContext *pFormatCtx;
int i, audioStream;
AVCodecContext *pCodecCtx;
AVCodec *pCodec;
char url[]="WavinFlag.aac";
//char url[]="1.mp2";
//char url[]="test.g711";
av_register_all();
pCodec=avcodec_find_decoder(AV_CODEC_ID_AAC);
if(pCodec==NULL){
printf("Codec not found.\n");
return -1;
}
pCodecCtx = avcodec_alloc_context3(pCodec);
// Open codec
if(avcodec_open2(pCodecCtx, pCodec,NULL)<0){
printf("Could not open codec.\n");
return -1;
}
FILE *pFile=NULL;
FILE *pInFile = NULL;
#if OUTPUT_PCM
pFile=fopen("output.pcm", "wb");
pInFile=fopen(url, "rb");
#endif
//Out Audio Param
uint64_t out_channel_layout=AV_CH_LAYOUT_STEREO;
int out_nb_samples=1024;
AVSampleFormat out_sample_fmt=AV_SAMPLE_FMT_S16;
int out_sample_rate=44100;
int out_channels=av_get_channel_layout_nb_channels(out_channel_layout);
//輸出記憶體大小
int out_buffer_size=av_samples_get_buffer_size(NULL,out_channels ,out_nb_samples,out_sample_fmt, 1);
uint8_t *out_buffer=(uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE*2);
AVFrame *pFrame = av_frame_alloc();
avcodec_get_frame_defaults(pFrame);
//SDL------------------
#if USE_SDL
//Init
if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
printf( "Could not initialize SDL - %s\n", SDL_GetError());
return -1;
}
//SDL_AudioSpec
SDL_AudioSpec wanted_spec;
wanted_spec.freq = out_sample_rate;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = out_channels;
wanted_spec.silence = 0;
wanted_spec.samples = out_nb_samples/*pCodecCtx->frame_size*/;
wanted_spec.callback = fill_audio;
wanted_spec.userdata = pCodecCtx;
if (SDL_OpenAudio(&wanted_spec, NULL)<0){
printf("can't open audio.\n");
return -1;
}
#endif
////printf("Bitrate:\t %3d\n", pFormatCtx->bit_rate);
//printf("Decoder Name:\t %s\n", pCodecCtx->codec->long_name);
//printf("Channels:\t %d\n", pCodecCtx->channels);
//printf("Sample per Second\t %d \n", pCodecCtx->sample_rate);
uint32_t ret,len = 0;
int got_picture;
int index = 0;
struct SwrContext *au_convert_ctx;
au_convert_ctx = swr_alloc();
//au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate,
// pCodecCtx->channel_layout,pCodecCtx->sample_fmt , pCodecCtx->sample_rate,0, NULL);
//swr_init(au_convert_ctx);
int data_size = 0;
int size = 0;
int cnt=0;
int offset=0;
FILE *myout=stdout;
unsigned char *aacframe=(unsigned char *)malloc(1024*5);
unsigned char *aacbuffer=(unsigned char *)malloc(1024*1024);
FILE *ifile = fopen(url, "rb");
if(!ifile){
printf("Open file error");
return -1;
}
printf(" aac NUM - Profile - Frequency - Size \n");
while(!feof(ifile)){
data_size = fread(aacbuffer+offset, 1, 1024*1024-offset, ifile);
unsigned char* input_data = aacbuffer+offset;
while(1)
{
int ret=getADTSframe(input_data, data_size, aacframe, &size);
if(ret==-1){
break;
}else if(ret==1){
memcpy(aacbuffer,input_data,data_size);
offset=data_size;
break;
}
char profile_str[10]={0};
char frequence_str[10]={0};
unsigned char profile=aacframe[2]&0xC0;
profile=profile>>6;
switch(profile){
case 0: sprintf(profile_str,"Main");break;
case 1: sprintf(profile_str,"LC");break;
case 2: sprintf(profile_str,"SSR");break;
default:sprintf(profile_str,"unknown");break;
}
unsigned char sampling_frequency_index=aacframe[2]&0x3C;
sampling_frequency_index=sampling_frequency_index>>2;
switch(sampling_frequency_index){
case 0: sprintf(frequence_str,"96000Hz");break;
case 1: sprintf(frequence_str,"88200Hz");break;
case 2: sprintf(frequence_str,"64000Hz");break;
case 3: sprintf(frequence_str,"48000Hz");break;
case 4: sprintf(frequence_str,"44100Hz");break;
case 5: sprintf(frequence_str,"32000Hz");break;
case 6: sprintf(frequence_str,"24000Hz");break;
case 7: sprintf(frequence_str,"22050Hz");break;
case 8: sprintf(frequence_str,"16000Hz");break;
case 9: sprintf(frequence_str,"12000Hz");break;
case 10: sprintf(frequence_str,"11025Hz");break;
case 11: sprintf(frequence_str,"8000Hz");break;
default:sprintf(frequence_str,"unknown");break;
}
AVPacket avpkt;
av_init_packet(&avpkt);
avpkt.data = (uint8_t *)aacframe;
avpkt.size = size;
while(avpkt.size>0)
{
ret = avcodec_decode_audio4( pCodecCtx, pFrame,&got_picture, &avpkt);
if ( ret < 0 ) {
printf("Error in decoding audio frame.\n");
return -1;
}
//需要解碼後初始化,否則pCodecCtx的引數尚未獲取到(純淨版不用預先指定引數)
static bool bSwrInit=false;
if(!bSwrInit){
au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate,
pCodecCtx->channel_layout,pCodecCtx->sample_fmt , pCodecCtx->sample_rate,0, NULL);
swr_init(au_convert_ctx);
bSwrInit=true;
}
if ( got_picture > 0 ){
int nBytes = swr_convert(au_convert_ctx,&out_buffer, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)pFrame->data , pFrame->nb_samples);
if(nBytes<0){
printf("swr_convert failed \n");
return 0;
}
nWriteBytes = av_samples_get_buffer_size(NULL, out_channels, nBytes, out_sample_fmt, 1);
if(nWriteBytes<0){
printf("av_samples_get_buffer_size failed \n");
return 0;
}
fwrite(out_buffer, 1, nWriteBytes, pFile);
}
//設定音訊資料緩衝,PCM資料
audio_chunk = (Uint8 *) out_buffer;
//設定音訊資料長度
audio_len = nWriteBytes;
audio_pos = audio_chunk;
//回放音訊資料
SDL_PauseAudio(0);
while(audio_len>0)//等待直到音訊資料播放完畢!
SDL_Delay(1);
avpkt.data+=ret;
avpkt.size-=ret;
}
av_free_packet(&avpkt);
data_size -= size;
input_data += size;
cnt++;
}
}
fclose(ifile);
free(aacbuffer);
fclose(pFile);
#if USE_SDL
SDL_CloseAudio();//關閉音訊裝置
SDL_Quit();
#endif
swr_free(&au_convert_ctx);
av_free(out_buffer);
avcodec_close(pCodecCtx);
return 0;
}