ffmpeg在开展音频转换时出错

#include <stdlib.h>#include <stdio.h>#include <string.h>#include <math.h>extern "C"{//#include <libavutil/imgutils.h>//#include <libavutil/opt.h>#include <libavcodec/avcodec.h>#include <libavutil/mathematics.h>//#include <libavutil/samplefmt.h>};extern "C"{#pragma comment (lib, "Ws2_32.lib")  #pragma comment (lib, "avcodec.lib")#pragma comment (lib, "avdevice.lib")#pragma comment (lib, "avfilter.lib")#pragma comment (lib, "avformat.lib")#pragma comment (lib, "avutil.lib")//#pragma comment (lib, "swresample.lib")#pragma comment (lib, "swscale.lib")}#define INBUF_SIZE 4096#define AUDIO_INBUF_SIZE 20480#define AUDIO_REFILL_THRESH 4096/* * Audio decoding. */void audio_encode_example(const char * inputfilename ,const char *outputfilename){    AVCodec *codec;    AVCodecContext *c= NULL;    int frame_size,  out_size, outbuf_size;    FILE * fin,*fout;    short *samples;    uint8_t *outbuf;    int numberframe = 0;    int size = 0;    int  FRAME_READ= 0;    printf("Audio encoding\n");    /* find the MP2 encoder */    codec = avcodec_find_encoder(CODEC_ID_AAC);    if (!codec)     {        fprintf(stderr, "codec not found\n");        exit(1);    }    c= avcodec_alloc_context();    /* put sample parameters */    c->bit_rate = 64000;    c->sample_rate = 44100;    c->channels = 2;    /* open it */    if (avcodec_open(c, codec) < 0)    {        fprintf(stderr, "could not open codec\n");        exit(1);    }    /* the codec gives us the frame size, in samples */    frame_size = c->frame_size;                                        samples = (short *)malloc(frame_size * 2 * c->channels);  //* 2 是因为 一般PCM数据都是16bit的 ,c->channels 是声道数    FRAME_READ  = frame_size * 2 * c->channels;    outbuf_size = 10000;    outbuf = (uint8_t *)malloc(outbuf_size);    fin = fopen(inputfilename, "rb+");    if (!fin)    {        fprintf(stderr, "could not open %s\n", inputfilename);        exit(1);    }    fout = fopen(outputfilename, "wb");    if (!fout)     {        fprintf(stderr, "could not open %s\n", outputfilename);        exit(1);    }    for(;;)    {                size = fread(samples, 1,FRAME_READ , fin);        if (size == 0)        {            break;        }        /* encode the samples */        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);        fwrite(outbuf, 1, out_size, fout);        numberframe ++ ;        printf("save frame %d\n",numberframe);            }    fclose(fout);    free(outbuf);    free(samples);    avcodec_close(c);    av_free(c);}int main(){    const char *EncodeOutputFilename_Audio;    const char *EncodeInputFilename_Audio;    EncodeInputFilename_Audio = "11.WAV";    EncodeOutputFilename_Audio = "22.AAC";//    avcodec_init(); //首先，main函数中一开始会去调用avcodec_init()函数，该函数的作用是初始化libavcodec，而我们在使用avcodec编解码库时，该函数必须被调用。    avcodec_register_all();//注册所有的编解码器（codecs），解析器（parsers）以及码流过滤器（bitstream filters）。当然我们也可以使用个别的注册函数来注册我们所要支持的格式。    audio_encode_example(EncodeInputFilename_Audio,EncodeOutputFilename_Audio); //编码    return getchar();}