#pragma once
#define __STDC_CONSTANT_MACROS
#define _CRT_SECURE_NO_WARNINGSextern "C"
{
#include "libavcodec/avcodec.h"
}//缓冲区大小（缓存5帧数据）
#define AUDIO_INBUF_SIZE 40960  
/*name   depthu8        8s16      16s32      32flt      32dbl      64u8p       8s16p     16s32p     32fltp     32dblp     64s64      64s64p     64//此代码解码的音频文件格式如下://AAC文件(一帧1024字节)，双声道（2），FLTP（32位，4字节）//AAC文件 frame_size 和 nb_samples 大小均为1024//一帧音频所占字节大小//1024*2*4=8192字节
*/
#define AUDIO_REFILL_THRESH 8192using namespace std;#define INPUT_FILE_NAME "d:\\123.aac"
#define OUTPUT_FILE_NAME "d:\\1111.pcm"static int get_format_from_sample_fmt(const char** fmt,	enum AVSampleFormat sample_fmt)
{struct sample_fmt_entry {enum AVSampleFormat sample_fmt; const char* fmt_be, * fmt_le;} sample_fmt_entries[] = {{ AV_SAMPLE_FMT_U8,  "u8",    "u8"    },{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },};*fmt = NULL;for (int i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {struct sample_fmt_entry* entry = &sample_fmt_entries[i];if (sample_fmt == entry->sample_fmt) {*fmt = AV_NE(entry->fmt_be, entry->fmt_le);return 0;}}av_log(NULL, AV_LOG_ERROR, "sample format %s is not supported as output format\n", av_get_sample_fmt_name(sample_fmt));return -1;
}static void decode(AVCodecContext* pCodecContext, AVFrame* pFrame, AVPacket* pPacket, FILE* pFile)
{int ret = avcodec_send_packet(pCodecContext, pPacket);if (ret < 0) {av_log(NULL, AV_LOG_ERROR, "发送数据包到解码器出错。\n");exit(1);}while (ret >= 0) {ret = avcodec_receive_frame(pCodecContext, pFrame);if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {return;}else if (ret < 0) {av_log(NULL, AV_LOG_ERROR, "Error sending a packet for decoding.\n");exit(1);}//获取每个采样点当中每个声道的大小int nDataSize = av_get_bytes_per_sample(pCodecContext->sample_fmt);if (nDataSize < 0) {av_log(NULL, AV_LOG_ERROR, "Failed to calculate data size.\n");exit(1);}//遍历采样点for (int i = 0; i < pFrame->nb_samples; i++) {//遍历声道for (int ch = 0; ch < pCodecContext->ch_layout.nb_channels; ch++) {fwrite(pFrame->data[ch] + nDataSize * i, 1, nDataSize, pFile);}}}
}int main(int argc, char* argv[])
{//初始化inbuf数字默认值uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE] = {0};//获取解码器（此处需要读取的文件是AAC，故）const AVCodec* pCodecOfAAC = avcodec_find_decoder(AV_CODEC_ID_AAC);if (!pCodecOfAAC) {av_log(NULL, AV_LOG_ERROR, "Codec not found.\n");exit(1);}//注册解析器AVCodecParserContext* pCodecParserParser = av_parser_init(pCodecOfAAC->id);if (!pCodecParserParser) {av_log(NULL, AV_LOG_ERROR, "parser not found.\n");exit(1);}//分配解析器上下文AVCodecContext* pCodecContextOfAAC = avcodec_alloc_context3(pCodecOfAAC);if (!pCodecContextOfAAC) {av_log(NULL, AV_LOG_ERROR, "Could not allocate video codec context.\n");exit(1);}//打开解码器if (avcodec_open2(pCodecContextOfAAC, pCodecOfAAC, NULL) < 0) {av_log(NULL, AV_LOG_ERROR, "Could not open codec.\n");exit(1);}//分配AVPacketAVPacket* pPacket = av_packet_alloc();if (!pPacket) {exit(1);}//分配AVFrameAVFrame* pFrame = av_frame_alloc();if (!pFrame) {exit(1);}//打开输入文件FILE* ifile = fopen(INPUT_FILE_NAME, "rb");if (!ifile) {av_log(NULL, AV_LOG_ERROR, "Could not open \s.\n", INPUT_FILE_NAME);exit(1);}//打开输入文件FILE* ofile = fopen(OUTPUT_FILE_NAME, "wb+");if (!ofile) {av_log(NULL, AV_LOG_ERROR, "Could not open \s.\n", OUTPUT_FILE_NAME);exit(1);}//从输入流 ifile 读取数据到 inbuf 所指向的数组中uint8_t* data = inbuf;size_t nDataSize = fread(inbuf, 1, AUDIO_INBUF_SIZE, ifile);while (nDataSize > 0) {//使用注册的解析器 parser 把数据分割成帧int nRet = av_parser_parse2(pCodecParserParser, pCodecContextOfAAC, &pPacket->data, &pPacket->size, data, nDataSize, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);if (nRet < 0) {fprintf(stderr, "Error while parsing\n");exit(1);}//根据使用情况重置数据位置data += nRet;nDataSize -= nRet;//送往解码if (pPacket->size) {decode(pCodecContextOfAAC, pFrame, pPacket, ofile);}//判断缓存区剩余数据是否小于一帧音频大小//小于的话从文件继续读取，之后在送往解码if (nDataSize < AUDIO_REFILL_THRESH) {memmove(inbuf, data, nDataSize);data = inbuf;int nLen = fread(data + nDataSize, 1, AUDIO_INBUF_SIZE - nDataSize, ifile);if (nLen > 0) {nDataSize += nLen;}}}//flush 解码器decode(pCodecContextOfAAC, pFrame, NULL, ofile);//此时就已经解码完了，我们稍后使用ffplay播放下音频//解码出来的pcm数据是没有这些基础数据的，我们需要从元数据获取//打印下基本信息//声道数printf("channels: %d \n", pCodecContextOfAAC->ch_layout.nb_channels);//采样率printf("sample_rate: %d  \n", pCodecContextOfAAC->sample_rate);//一帧音频所占字节代销printf("buffer: %d  \n", av_samples_get_buffer_size(NULL, pCodecContextOfAAC->ch_layout.nb_channels, pCodecContextOfAAC->frame_size, pCodecContextOfAAC->sample_fmt, 1));//采样格式enum AVSampleFormat sfmt = pCodecContextOfAAC->sample_fmt;printf("sample_fmt: %s  \n", av_get_sample_fmt_name(sfmt));//如果为planar，转换为packed格式if (av_sample_fmt_is_planar(sfmt)) {const char* packed = av_get_sample_fmt_name(sfmt);sfmt = av_get_packed_sample_fmt(sfmt);}const char* fmt = NULL;if (get_format_from_sample_fmt(&fmt, sfmt) < 0) {av_log(NULL, AV_LOG_ERROR, "Could not get forma \s.\n", av_get_sample_fmt_name(sfmt));exit(1);}//资源释放fclose(ifile);fclose(ofile);av_parser_close(pCodecParserParser);avcodec_free_context(&pCodecContextOfAAC);av_frame_free(&pFrame);av_packet_free(&pPacket);return 0;
}