我是 ffmpeg 的新手，我尝试使用 api-example.c 解码 wma 文件。但是，当我运行该程序时，它给了我一个错误说
“frame_len 溢出”。有谁知道如何解决这个错误？
这是我的代码：
extern "C" {
#include <avcodec.h>
#include "../libavcodec/avcodec.h"
#include <avformat.h>
}
#include <iostream>
#include <assert.h>
#include <windows.h>
#include <mmsystem.h>
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
int main(int argc, char *argv[]) {
avcodec_init();
avcodec_register_all();
//avdevice_register_all();
av_register_all();
AVCodec *codec;
AVCodecContext *c= NULL;
AVCodec *ocodec;
AVCodecContext *oc= NULL;
int out_size, len,out_size2;
FILE *f, *outfile;
uint8_t *outbuf;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
char* outfilename="test.wma";
char* filename="Beethoven's.wma";
AVFormatContext *pFormatCtx;
WAVEFORMATEX* wfx=new WAVEFORMATEX;
int ret;
ret=av_open_input_file(&pFormatCtx, filename, NULL, 0, NULL);
if(ret!=0)
{
std::cout<<"cannot open file!"<<std::endl;
exit(1);
}
if(av_find_stream_info(pFormatCtx)<0)
{
std::cout<<"cannot find stream!"<<std::endl;
exit(1);
}
int audioStream;
AVCodecContext *pCodecCtx;
// Find the first video stream
audioStream=-1;
for(int i=0; i<pFormatCtx->nb_streams; i++)
if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO)
{
audioStream=i;
break;
}
if(audioStream==-1)
{
std::cout<<"cannot find audio!"<<std::endl;
}
// Get a pointer to the codec context for the audio stream
pCodecCtx=pFormatCtx->streams[audioStream]->codec;
av_init_packet(&avpkt);
printf("Audio decoding\n");
/* find the suitable audio decoder */
codec = avcodec_find_decoder(pCodecCtx->codec_id);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
if(codec->capabilities & CODEC_CAP_TRUNCATED)
pCodecCtx->flags|=CODEC_FLAG_TRUNCATED;
//open the codec (for decoding)
int test = avcodec_open(pCodecCtx, codec);
if (test < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
//find mp3 encoder
ocodec = avcodec_find_encoder(CODEC_ID_MP3);
if (!ocodec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
//allocate context
oc= avcodec_alloc_context();
/* put sample parameters */
oc->bit_rate = 64000;
oc->sample_rate = 44100;
oc->channels = 1;
/* open it */
if (avcodec_open(oc, ocodec) < 0) {
fprintf(stderr, "could not open encoding codec\n");
exit(1);
}
//buffer
outbuf = (uint8_t*)malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
//open inputfile
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
//open outputfile
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
//while there is still data
while (avpkt.size > 0) {
std::cout<<"decoding..."<<std::endl;
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
//decode
len = avcodec_decode_audio3(pCodecCtx, (short *)outbuf, &out_size, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (out_size > 0) {
/* if a frame has been decoded, output it */
std::cout<<"1 frame decoded!"<<std::endl;
out_size2 = avcodec_encode_audio(oc, outbuf, out_size, (short*)outbuf);
fwrite(outbuf, 1, out_size2, outfile);
}
//subtract data from whatever decode function returns
avpkt.size -= len;
avpkt.data += len;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
}
我已经坚持了很长时间。请帮我。有人知道我的代码有什么问题吗？
谢谢，
伊扎克