avcodec_decode_audio2返回值为-1？

我用ffmpeg4.0版本代替旧版本3.2时，解码时发现avcodec_decode_audio2返回值总为-1，我程序中代码如下：

        int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;     //#define AVCODEC_MAX_AUDIO_FRAME_SIZE 2<<20 

        uint8_t * inbuf = (uint8_t *)malloc(out_size);

        FILE* pFileWav;

        int fileSize; 

        pFileWav= fopen("b.pcm","wb+");

        while(av_read_frame(pFormatCtx, &packet)>=0)

        {

               if(packet.stream_index==audioStream) {

                       pktdata = packet.data;

                       pktsize = packet.size;

                       while(pktsize>0)

                       {

                               //解码

                               int len=avcodec_decode_audio2(pCodecCtx,(int16_t *)inbuf,&out_size,pktdata,pktsize);

//                             int len=avcodec_decode_audio3(pCodecCtx,(int16_t *)inbuf,&out_size,&packet);

                               if (len<0)

                               {

                                      printf("Error while decoding.\n");

                                      break;

                               }

                               if(out_size>0)

                               {

                                      fwrite(inbuf,1,out_size,pFileWav);//pcm记录

                                      fflush(pFileWav);

                                      fileSize += out_size;

                               }

                               pktsize -= len;

                               pktdata += len;

                       }

               }   

               av_free_packet(&packet);

        }   

 

后来查了下ffmpeg3.2源代码,发现：

int avcodec_decode_audio(AVCodecContext *avctx, int16_t *samples,

                         int *frame_size_ptr,

                         const uint8_t *buf, int buf_size){

    *frame_size_ptr= AVCODEC_MAX_AUDIO_FRAME_SIZE;

    return avcodec_decode_audio2(avctx, samples, frame_size_ptr, buf, buf_size);

}

int attribute_align_arg avcodec_decode_audio2(AVCodecContext *avctx, int16_t *samples,

                         int *frame_size_ptr,

                         const uint8_t *buf, int buf_size)

{

    int ret;

    if((avctx->codec->capabilities & CODEC_CAP_DELAY) || buf_size){

        //FIXME remove the check below _after_ ensuring that all audio check that the available space is enough

        if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){

            av_log(avctx, AV_LOG_ERROR, "buffer smaller than AVCODEC_MAX_AUDIO_FRAME_SIZE\n");

            return -1;

        }

        if(*frame_size_ptr < FF_MIN_BUFFER_SIZE ||

        *frame_size_ptr < avctx->channels * avctx->frame_size * sizeof(int16_t)){

            av_log(avctx, AV_LOG_ERROR, "buffer %d too small\n", *frame_size_ptr);

            return -1;

        }

        ret = avctx->codec->decode(avctx, samples, frame_size_ptr,

                                buf, buf_size);

        avctx->frame_number++;

    }else{

        ret= 0;

        *frame_size_ptr=0;

    }

    return ret;

}

ffmpeg4.0中代码为：

int attribute_align_arg avcodec_decode_audio2(AVCodecContext *avctx, int16_t *samples,

                         int *frame_size_ptr,

                         const uint8_t *buf, int buf_size)

{

    AVPacket avpkt;

    av_init_packet(&avpkt);

    avpkt.data = buf;

    avpkt.size = buf_size;

    return avcodec_decode_audio3(avctx, samples, frame_size_ptr, &avpkt);

}

#endif

int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,

                         int *frame_size_ptr,

                         AVPacket *avpkt)

{

    int ret;

    if((avctx->codec->capabilities & CODEC_CAP_DELAY) || avpkt->size){

        //FIXME remove the check below _after_ ensuring that all audio check that the available space is enough

        if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){

            av_log(avctx, AV_LOG_ERROR, "buffer smaller than AVCODEC_MAX_AUDIO_FRAME_SIZE\n");

            return -1;

        }

        if(*frame_size_ptr < FF_MIN_BUFFER_SIZE ||

        *frame_size_ptr < avctx->channels * avctx->frame_size * sizeof(int16_t)){

            av_log(avctx, AV_LOG_ERROR, "buffer %d too small\n", *frame_size_ptr);

            return -1;

        }

        ret = avctx->codec->decode(avctx, samples, frame_size_ptr, avpkt);

        avctx->frame_number++;

    }else{

        ret= 0;

        *frame_size_ptr=0;

    }

    return ret;

}

 

即每次调用avcodec_decode_audio2时，若(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE)，则返回值为-1，后来我就顺着这个思路，往上找，过不其然，发现了问题之所在，原因是：在intlen=avcodec_decode_audio2(pCodecCtx,(int16_t *)inbuf,&out_size,pktdata,pktsize)处理后，out_size的值变为4608，而不是AVCODEC_MAX_AUDIO_FRAME_SIZE，所以在4.0中，*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE，返回值自然为-1；

找到了问题，解决自然容易，要么在ffmpeg源码处该，要么在客户端，而我选择了后者，主要是方便：

int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;

        uint8_t * inbuf = (uint8_t *)malloc(out_size);

        FILE* pFileWav;

        int fileSize; 

        pFileWav= fopen("b.pcm","wb+");

        while(av_read_frame(pFormatCtx, &packet)>=0)

        {

               if(packet.stream_index==audioStream) {

                       pktdata = packet.data;

                       pktsize = packet.size;

                       out_size=AVCODEC_MAX_AUDIO_FRAME_SIZE;    //加这一句，就OK了

                       while(pktsize>0)

                       {

                               //解码

                               int len=avcodec_decode_audio2(pCodecCtx,(int16_t *)inbuf,&out_size,pktdata,pktsize);

//                             int len=avcodec_decode_audio3(pCodecCtx,(int16_t *)inbuf,&out_size,&packet);

                               if (len<0)

                               {

                                      printf("Error while decoding.\n");

                                      break;

                               }

                               if(out_size>0)

                               {

                                      fwrite(inbuf,1,out_size,pFileWav);//pcm记录

                                      fflush(pFileWav);

                                      fileSize += out_size;

                               }

                               pktsize -= len;

                               pktdata += len;

                       }

               }   

               av_free_packet(&packet);

        }