ffmpeg在进行音频转换时出错
在wav转aac时出以下错误：
Audio encoding
[NULL @ 00392440] Codec is experimental but experimental codecs are not enabled,
try -strict -2
could not open codec
感觉好像是编译的时候没有把一些解码的库也编译进去一样。求高手解答啊。
我编译库的选项是 ./configure --enable-shared --disable-static --enbla-memalign-hack
感觉代码应该没什么问题，因为用从网上下载的SDK就可以正常运行，估计是库的问题，请各位指点一二。
下面是代码:
#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();
}
