Error when decode AAC with avcodec_decode_audio4() - c

Question

I'm trying to decode AAC with FFmpeg native decoder and encountered an error
SSR is not implemeted. Update your FFmpeg version to newest from Git. If the problem still occurs, it mean that your file has a feature which has not implemented.
Function avcodec_decode_audio4() return -1163346256. Is this because of FFmpeg version? I downloaded shared and dev version from here. Is this up to date?
Here is the source code:
#include "stdafx.h"
#include "stdio.h"
#include "conio.h"
extern "C"
{
#ifndef __STDC_CONSTANT_MACROS
#define __STDC_CONSTANT_MACROS
#endif
#include <libavcodec\avcodec.h>
#include <libavformat/avformat.h>
}
// compatibility with newer API
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,28,1)
#define av_frame_alloc avcodec_alloc_frame
#define av_frame_free avcodec_free_frame
#endif
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static void audio_decode_example(const char *outfilename, const char *filename);
int main(int argc, char *argv[]) {
audio_decode_example("D:\\sample.pcm","D:\\sample.m4a");
getch();
return 0;
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVFormatContext *pFormatCtx = NULL;
AVCodecContext *pCodecCtxOrig = NULL;
AVCodecContext * pCodecCtx= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_register_all();
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
// Open file to get format context
if(avformat_open_input(&pFormatCtx, filename, NULL, NULL)!=0){
printf("Couldn't open file");
return; // Couldn't open file
}
// Retrieve stream information
if(avformat_find_stream_info(pFormatCtx, NULL)<0){
printf("Couldn't find stream information");
return; // Couldn't find stream information
}
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, filename, 0);
// Find the first audio stream
int audioStream = -1;
int i =0;
for(i=0; i<pFormatCtx->nb_streams; i++) {
if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO) {
audioStream=i;
break;
}
}
if(audioStream==-1) {
printf("Didn't find a audio stream");
return; // Didn't find a audio stream
}
// Get a pointer to the codec context for the audio stream
pCodecCtxOrig=pFormatCtx->streams[audioStream]->codec;
// Find the decoder for the audio stream
codec=avcodec_find_decoder(pCodecCtxOrig->codec_id);
if(codec==NULL) {
fprintf(stderr, "Codec not found\n");
return; // Codec not found
}
pCodecCtx = avcodec_alloc_context3(codec);
if (!pCodecCtx) {
fprintf(stderr, "Could not allocate audio codec context\n");
return;
}
if(avcodec_copy_context(pCodecCtx, pCodecCtxOrig) != 0) {
fprintf(stderr, "Couldn't copy codec context");
return; // Error copying codec context
}
/* open it */
if (avcodec_open2(pCodecCtx, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
return;
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
return;
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(pCodecCtx);
return;
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
return;
}
}
len = avcodec_decode_audio4(pCodecCtx, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding. len = %d \n",len);
return;
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(pCodecCtx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
return;
}
for (i=0; i < decoded_frame->nb_samples; i++)
for (ch=0; ch < pCodecCtx->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
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);
avcodec_close(pCodecCtx);
av_free(pCodecCtx);
av_frame_free(&decoded_frame);
}
I have also read this question: How to decode AAC using avcodec_decode_audio4? but no solution is provided.

Answer 1

f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
return;
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(pCodecCtx);
return;
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
Yeah, that's not going to work. You can't dump raw bytes from some random muxing format (potentially mp4) into a decoder and expect it to work. Use av_read_frame() to read individual audio packets from the muxing format, and feed the resulting AVPacket into the decoder using avcodec_decode_audio4(). See e.g. the dranger api tutorial. I know api-example.c uses the above code, but that unfortunately only works for a very limited subset of cases. Also see the detailed description in the API docs.