FFMPEG实现对AAC解码(采用封装格式实现)

技术在于交流、沟通，转载请注明出处并保持作品的完整性。

原文：https://blog.csdn.net/hiwubihe/article/details/81261022

[音频编解码系列文章]

音频编解码基础
FFMPEG实现音频重采样
FFMPEG实现PCM编码(采用封装格式实现)
FFMPEG实现PCM编码(不采用封装格式实现)
FAAC库实现PCM编码
FAAD库实现RAW格式AAC解码
FAAD库实现RAW格式AAC封装成ADTS格式
FAAD库实现ADTS格式解码
FFMPEG实现对AAC解码(采用封装格式实现)
FFMPEG实现对AAC解码(不采用封装格式实现)

本篇FFMEPG实现对AAC解码，解码结果保存wav格式。对AAC编码文件来说，编码根据音频参数编码，解码根据音频参数重新构建声波，FFMPEG构建的音频存储方式不一定支持播放, 所以需要重采样样本，如AAC解码的样本格式AV_SAMPLE_FMT_FLTP。AAC的解码器如果是外部解码器"aac",解码格式需要AV_SAMPLE_FMT_FLTP，如果是“libvo_aacenc”这个解码器需要格式AV_SAMPLE_FMT_S16。

DEMO源代码如下：

/*******************************************************************************
Copyright (c) wubihe Tech. Co., Ltd. All rights reserved.
--------------------------------------------------------------------------------Date Created:   2014-10-25
Author:         wubihe QQ:1269122125 Email:1269122125@qq.com
Description:    音频格式AAC MP3的解码,结果保存WAV格式
--------------------------------------------------------------------------------
Modification History
DATE          AUTHOR          DESCRIPTION
--------------------------------------------------------------------------------********************************************************************************/#include <stdlib.h>
#include <string.h>
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"};#define INPUT_FILE_NAME       ("huangdun.aac")
#define OUTPUT_FILE_NAME     ("huangdun.wav")
#define  MAXWAVESIZE         (4294967040LU)//写WAV文件头
static int write_wav_header(int iBitPerSample,int iChans,unsigned char ucFormat,int iSampleRate,int iTotalSamples,FILE*pFile)
{unsigned char header[44];unsigned char* p = header;unsigned int bytes = (iBitPerSample + 7) / 8;float data_size = (float)bytes * iTotalSamples;unsigned long word32;*p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F';word32 = (data_size + (44 - 8) < (float)MAXWAVESIZE) ?(unsigned long)data_size + (44 - 8)  :  (unsigned long)MAXWAVESIZE;*p++ = (unsigned char)(word32 >>  0);*p++ = (unsigned char)(word32 >>  8);*p++ = (unsigned char)(word32 >> 16);*p++ = (unsigned char)(word32 >> 24);*p++ = 'W'; *p++ = 'A'; *p++ = 'V'; *p++ = 'E';*p++ = 'f'; *p++ = 'm'; *p++ = 't'; *p++ = ' ';*p++ = 0x10; *p++ = 0x00; *p++ = 0x00; *p++ = 0x00;if (ucFormat == AV_SAMPLE_FMT_FLT){*p++ = 0x03; *p++ = 0x00;} else {*p++ = 0x01; *p++ = 0x00;}*p++ = (unsigned char)(iChans >> 0);*p++ = (unsigned char)(iChans >> 8);word32 = (unsigned long)(iSampleRate + 0.5);*p++ = (unsigned char)(word32 >>  0);*p++ = (unsigned char)(word32 >>  8);*p++ = (unsigned char)(word32 >> 16);*p++ = (unsigned char)(word32 >> 24);word32 = iSampleRate * bytes * iChans;*p++ = (unsigned char)(word32 >>  0);*p++ = (unsigned char)(word32 >>  8);*p++ = (unsigned char)(word32 >> 16);*p++ = (unsigned char)(word32 >> 24);word32 = bytes * iChans;*p++ = (unsigned char)(word32 >>  0);*p++ = (unsigned char)(word32 >>  8);*p++ = (unsigned char)(iBitPerSample >> 0);*p++ = (unsigned char)(iBitPerSample >> 8);*p++ = 'd'; *p++ = 'a'; *p++ = 't'; *p++ = 'a';word32 = data_size < MAXWAVESIZE ?(unsigned long)data_size : (unsigned long)MAXWAVESIZE;*p++ = (unsigned char)(word32 >>  0);*p++ = (unsigned char)(word32 >>  8);*p++ = (unsigned char)(word32 >> 16);*p++ = (unsigned char)(word32 >> 24);return fwrite(header, sizeof(header), 1, pFile);
}int main()
{FILE *pOutFile=NULL;pOutFile = fopen(OUTPUT_FILE_NAME, "wb");if(pOutFile == NULL){return -1;}//注册解码器av_register_all();avformat_network_init();AVFormatContext  * pFormatCtx = avformat_alloc_context();//打开封装格式if(avformat_open_input(&pFormatCtx,INPUT_FILE_NAME,NULL,NULL)!=0){return -1;}//if(avformat_find_stream_info(pFormatCtx,NULL)<0){printf("封装格式查找流失败.\n");return -1;}// Dump valid information onto standard errorav_dump_format(pFormatCtx, 0, INPUT_FILE_NAME, false);// Find the first audio streamint audioStream = -1;for(int i=0; i < pFormatCtx->nb_streams; i++){if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){audioStream=i;break;}}if(audioStream==-1){printf("Didn't find a audio stream.\n");return -1;}// Get a pointer to the codec context for the audio streamAVCodecContext *pCodecCtx=pFormatCtx->streams[audioStream]->codec;// Find the decoder for the audio streamAVCodec            *pCodec=avcodec_find_decoder(pCodecCtx->codec_id);if(pCodec==NULL){printf("Codec not found.\n");return -1;}// Open codecif(avcodec_open2(pCodecCtx, pCodec,NULL)<0){printf("Could not open codec.\n");return -1;}//对AAC编码文件来说，编码根据音频参数编码，解码根据音频参数重新构建声波，FFMPEG构建的//音频存储方式不一定支持播放，所以需要重采样样本 如AAC解码的样本格式是AV_SAMPLE_FMT_FLTPuint64_t iInputLayout                = av_get_default_channel_layout(pCodecCtx->channels);int      iInputChans                = pCodecCtx->channels;AVSampleFormat eInputSampleFormat   = pCodecCtx->sample_fmt;int      iInputSampleRate           = pCodecCtx->sample_rate;uint64_t iOutputLayout              = av_get_default_channel_layout(pCodecCtx->channels);int      iOutputChans               = pCodecCtx->channels;AVSampleFormat eOutputSampleFormat  = AV_SAMPLE_FMT_S16;int         iOutputSampleRate          = pCodecCtx->sample_rate;SwrContext *pSwrCtx = swr_alloc_set_opts(NULL,iOutputLayout, eOutputSampleFormat, iOutputSampleRate,iInputLayout,eInputSampleFormat , iInputSampleRate,0, NULL);swr_init(pSwrCtx);//AVPacket读取原始解码前的数据AVPacket *packet=(AVPacket *)malloc(sizeof(AVPacket));av_init_packet(packet);//1帧数据样本数int iFrameSamples = pCodecCtx->frame_size;// 存储原始数据 int iRawLineSize = 0;int iRawBuffSize  = av_samples_get_buffer_size(&iRawLineSize, iInputChans, iFrameSamples, eInputSampleFormat, 0);uint8_t *pRawBuff = (uint8_t *)av_malloc(iRawBuffSize);//原始数据保存在AVFrame结构体中AVFrame* pRawframe = av_frame_alloc();pRawframe->nb_samples  = iFrameSamples;pRawframe->format        = eInputSampleFormat;pRawframe->channels     = iInputChans;int iReturn = avcodec_fill_audio_frame(pRawframe, iInputChans, eInputSampleFormat, (const uint8_t*)pRawBuff, iRawBuffSize, 0);if(iReturn<0){return -1;}// 存储转换后数据 int iConvertLineSize = 0;int iConvertBuffSize  = av_samples_get_buffer_size(&iConvertLineSize, iOutputChans, iFrameSamples, eOutputSampleFormat, 0);uint8_t *pConvertBuff = (uint8_t *)av_malloc(iConvertBuffSize);//转换后数据保存在AVFrame结构体中AVFrame* pConvertframe = av_frame_alloc();pConvertframe->nb_samples   = iFrameSamples;pConvertframe->format        = eOutputSampleFormat;pConvertframe->channels     = iOutputChans;iReturn = avcodec_fill_audio_frame(pConvertframe, iOutputChans, eOutputSampleFormat, (const uint8_t*)pConvertBuff, iConvertBuffSize, 0);if(iReturn<0){return -1;}int iGetPicture;int iDecodeRet;int iFrameNo = 0;write_wav_header(16,iOutputChans,eOutputSampleFormat,iOutputSampleRate,0,pOutFile);while(av_read_frame(pFormatCtx, packet)>=0){if(packet->stream_index==audioStream){iDecodeRet = avcodec_decode_audio4( pCodecCtx, pRawframe,&iGetPicture, packet);if ( iDecodeRet < 0 ) {printf("Error in decoding audio frame.\n");return -1;}if ( iGetPicture > 0 ){printf("FrameNo:%5d\n",iFrameNo);swr_convert(pSwrCtx, (uint8_t**)pConvertframe->data, iFrameSamples ,(const uint8_t**)pRawframe->data, iFrameSamples );fwrite(pConvertframe->data[0],pConvertframe->linesize[0],1,pOutFile);iFrameNo++;}}av_free_packet(packet);}av_free(pRawBuff);av_free(pConvertBuff);swr_free(&pSwrCtx);avcodec_close(pCodecCtx);avformat_close_input(&pFormatCtx);fclose(pOutFile);printf("Aac encode Success!!\n");getchar();return 0;
}

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