[Libav-user] Decoing problem for aac to wav using api's code

[Amir Raza]

Hi experts,
This is my first mail to ffmpeg , apologies if make some mistakes.
I tried many online example codes for decoding aac audio to wav file.
including example codes which is for MP2 codec.
below is one such example code , it doubles the decoded file size (.wav)
but not playable.
Can any one point in right direction or help me in correcting the below
code.
Thanks in advance



#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavutil/frame.h>
#include <libavutil/mem.h>
#include <libavcodec/avcodec.h>
#define AUDIO_INBUF_SIZE 4096
#define AUDIO_REFILL_THRESH 4096
 static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
  FILE *outfile)
{
 int i, ch;
 int ret, data_size;

 /* send the packet with the compressed data to the decoder */
 ret = avcodec_send_packet(dec_ctx, pkt);
 if (ret < 0) {
  fprintf(stderr, "Error submitting the packet to the decoder\n");
  exit(1);
 }
 /* read all the output frames (in general there may be any number of them
*/
 while (ret >= 0)
 {
  ret = avcodec_receive_frame(dec_ctx, frame);
  if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
   return;
  else if (ret < 0)
  {
   fprintf(stderr, "Error during decoding\n");
   exit(1);
  }
  data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
  if (data_size < 0) {
   /* This should not occur, checking just for paranoia */
   fprintf(stderr, "Failed to calculate data size\n");
   exit(1);
  }
  for (i = 0; i < frame->nb_samples; i++)
   for (ch = 0; ch < dec_ctx->channels; ch++)
    fwrite(frame->data[ch] + data_size * i, 1, data_size, outfile);
 }
}
int main(int argc, char **argv)
{
 const char *outfilename, *filename;
 const AVCodec *codec;
 AVCodecContext *c = NULL;
 AVCodecParserContext *parser = NULL;
 int len, ret;
 FILE *f, *outfile;
 uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
 uint8_t *data;
 size_t   data_size;
 AVPacket *pkt;
 AVFrame *decoded_frame = NULL;
 if (argc <= 2) {
  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
  exit(0);
 }
 filename = argv[1];
 outfilename = argv[2];
 pkt = av_packet_alloc();

 /* find the MPEG audio decoder */
 codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
 if (!codec) {
  fprintf(stderr, "Codec not found\n");
  exit(1);
 }
 parser = av_parser_init(codec->id);
 if (!parser) {
  fprintf(stderr, "Parser not found\n");
  exit(1);
 }
 c = avcodec_alloc_context3(codec);
 c->channels = 2;
 c->sample_rate = 44100;
 c->bit_rate = 32;
 if (!c) {
  fprintf(stderr, "Could not allocate audio codec context\n");
  exit(1);
 }
 /* open it */
 if (avcodec_open2(c, codec, NULL) < 0) {
  fprintf(stderr, "Could not open codec\n");
  exit(1);
 }
 f = fopen(filename, "rb");
 if (!f) {
  fprintf(stderr, "Could not open %s\n", filename);
  exit(1);
 }
 outfile = fopen(outfilename, "wb");
 if (!outfile) {
  av_free(c);
  exit(1);
 }
 /* decode until eof */
 data = inbuf;
 data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
 while (data_size > 0) {
  if (!decoded_frame) {
   if (!(decoded_frame = av_frame_alloc())) {
    fprintf(stderr, "Could not allocate audio frame\n");
    exit(1);
   }
  }
  ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
   data, data_size,
   AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
  if (ret < 0) {
   fprintf(stderr, "Error while parsing\n");
   exit(1);
  }
  data += ret;
  data_size -= ret;
  if (pkt->size)
   decode(c, pkt, decoded_frame, outfile);
  if (data_size < AUDIO_REFILL_THRESH) {
   memmove(inbuf, data, data_size);
   data = inbuf;
   len = fread(data + data_size, 1,
    AUDIO_INBUF_SIZE - data_size, f);
   if (len > 0)
    data_size += len;
  }
 }
 /* flush the decoder */
 pkt->data = NULL;
 pkt->size = 0;
 decode(c, pkt, decoded_frame, outfile);
 fclose(outfile);
 fclose(f);
 avcodec_free_context(&c);
 av_parser_close(parser);
 av_frame_free(&decoded_frame);
 av_packet_free(&pkt);
 return 0;
}

Regards,
Aasim
+91 8197948544
Regards,
Aasim
+91 8197948544
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