AAC--ffmpeg解码AAC

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我是靠谱客的博主包容指甲油，最近开发中收集的这篇文章主要介绍AAC--ffmpeg解码AAC，觉得挺不错的，现在分享给大家，希望可以做个参考。

概述

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，例如新版ffmpeg解码AAC解码后得到AV_SAMPLE_FMT_FLTP，该格式SDKL播放不支持该格式

ffmpeg所支持的音频格式中：有的仅内部使用，有的可以外部作为音频文件格式

AVFormatContext	* pFormatCtx = avformat_alloc_context();
avformat_open_input(&pFormatCtx,INPUT_FILE_NAME,NULL,NULL)‘
avformat_find_stream_info(pFormatCtx,NULL);
// Dump valid information onto standard error
av_dump_format(pFormatCtx, 0, INPUT_FILE_NAME, false);
// Find the first audio stream????????int audioStream
p_codec_par = p_fmt_ctx->streams[a_idx]->codecpar;
AVCodec	*pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
AVCodecContext	*p_codec_ctx = avcodec_alloc_context3(p_codec);
avcodec_parameters_to_context(p_codec_ctx, p_codec_par);
// Open codec
avcodec_open2(pCodecCtx, pCodec,NULL);
uint64_t iInputLayout
= av_get_default_channel_layout(pCodecCtx->channels);//使用默认声道分布
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;
//number of audio samples (per channel) described by this frame
pRawframe->format
= eInputSampleFormat;
pRawframe->channels
= iInputChans;
avcodec_fill_audio_frame(pRawframe, iInputChans, eInputSampleFormat, (const uint8_t*)pRawBuff, iRawBuffSize, 0);
/*
swr的接收帧pConvertframe
*/
// 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;
avcodec_fill_audio_frame(pConvertframe, iOutputChans, eOutputSampleFormat, (const uint8_t*)pConvertBuff, iConvertBuffSize, 0);
int iGetPicture;
int iDecodeRet;
int iFrameNo = 0;
write_wav_header(16,iOutputChans,eOutputSampleFormat,iOutputSampleRate,0,pOutFile);//自定义，按照wav格式
while(av_read_frame(pFormatCtx, packet)>=0)
{
if(packet->stream_index==audioStream)
{
avcodec_decode_audio4( pCodecCtx, pRawframe,&iGetPicture, packet);//要解码成功&&拿到pkt
if ( iGetPicture > 0 )
{
printf("FrameNo:%5dn",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;