How to decode AAC using avcodec_decode_audio4? - ffmpeg

Question

I changed in my code avcodec_decode_audio3 to avcodec_decode_audio4 and added the frame handling. But now I cannot decode AAC frames anymore.
Why does avcodec_decode_audio4 return -22 (invalid argument)?
Following the answer below, does this have something to do with the parameters in AVContext that need to be set?
I had to use avcodec_decode_audio4 because I updated my ffmpeg and then got the following error:
[NULL # 0xb14f020] Custom get_buffer() for use withavcodec_decode_audio3() detected.
Overriding with avcodec_default_get_buffer
[NULL # 0xb14f020] Please port your application to avcodec_decode_audio4()
According to Buffer error in avcodec_decode_audio4() this is a regression, is there any other solution for this than going back to ffmpeg < 0.8 ?
The decoder using avcodec_decode_audio4:
AVCodec *codec;
AVCodecContext *avCtx;
AVFrame * decoded_frame = NULL;
uint8_t *outbuf = static_cast<uint8_t *>(malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE));
AVPacket avPacket;
main(){
av_register_all();
codec = avcodec_find_decoder(CODEC_ID_AAC);
//set parameters
avCtx = avcodec_alloc_context3(codec);
avCtx->channels = 1;
avCtx->sample_rate = 44100;
avCtx->bit_rate=16;
if (avcodec_open2(avCtx, codec, NULL) < 0) printf("Could not open codec\n");
av_init_packet(&avPacket);
//Main decoder loop
while(1)
my_frame_decoder();
return 0;
}
void my_frame_decoder() {
//get data
...
avPacket.size = numBytes;
avPacket.data = inputBytes;
int len;
while (avPacket.size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = avcodec_alloc_frame())) {
printf("out of memory");
return;
}
} else {
avcodec_get_frame_defaults(decoded_frame);
}
//-------------------->> returns always -22
len = avcodec_decode_audio4(avCtx, decoded_frame, &got_frame, &avPacket);
//do something with the decoded frame
...
avPacket.size -= len;
avPacket.data += len;
}
return;
}

Answer 1

After hours of searching, i found out that the dec_ctx of the avcodec_decode_audio4 must be opened by a dec_codec initialised by av_find_best_stream()
1° av_find_best_stream(in_fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1,
&dec_codec, 0);<br>
2° dec_ctx = m_in_aud_strm->codec;<br>
3° av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);<br>
4° avcodec_open2(dec_ctx, dec_codec, NULL)<br>
.
.
.
5° avcodec_decode_audio4(dec_ctx, pFrame, &got_frame, &pkt);

Answer 2

I think the problem is the parameters set in your codec's context. Please refer to https://www.ffmpeg.org/doxygen/trunk/structAVCodecContext.html for setting the parameters, which has changed from avcodec_decode_audio3 to avcodec_decode_audio4.

Answer 3

No solution but a workaround is to go back to older builds. After testing various builds that work with avcodec_decode_audio3, I thought it might be useful for others to know that ffmpeg-0.10.14.tar.bz2 from https://ffmpeg.org/releases/ works.

Answer 4

<!-- language: c++ -->
#include <windows.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include <errno.h>
#include <fcntl.h>
#include <ctype.h>
#include <math.h>
#include <wctype.h>
#include <wchar.h>
#include <stdarg.h>
#include <stddef.h>
#include <setjmp.h>
#include <locale.h>
#include <signal.h>
#include <limits.h>
#include <float.h>
#include <iso646.h>
#undef NDEBUG
#include <assert.h>
// Use avcodec_send_packet() and avcodec_receive_frame().
//sample code
while (av_read_frame (FormatContext, packet_ptr) >= 0)
{
/* some code */
if (packet_ptr->stream_index == audiostream)
{
assert(NULL == decoded_frame);
decoded_frame = av_frame_alloc();
ret = avcodec_send_packet(pCodecCtx, packet_ptr);
if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF)
{
av_packet_unref (packet_ptr);
if (decoded_frame)
{
av_frame_unref(decoded_frame);
decoded_frame = NULL;
}
continue;
}
else
{
if (0 <= ret)
packet_ptr->size = 0;
ret = avcodec_receive_frame(pCodecCtx, decoded_frame);
if (ret >= 0)
got_frame = 1;
else
{
got_frame = 0;
if (decoded_frame)
{
av_frame_unref(decoded_frame);
decoded_frame = NULL;
}
av_packet_unref (packet_ptr);
continue;
}
}
if(AV_SAMPLE_FMT_FLTP == pCodecCtx->sample_fmt)//AAC sample format for Libav released 10-October-2020 (ffmpeg 4.3.1)
{
//now get the PCM data ready to play or save
int nb_samples = decoded_frame->nb_samples;
int channels = pCodecCtx->channels;
if(channels > 2) //for this small sample only 2 channels...
{
channels = 2;//it will convert multichannel media files to 2 channels, remember this...more code need to be modified
}
int outputBufferLen = nb_samples * channels * 2;
int size_out //the size of the PCM data as sizeof(char)
=outputBufferLen;
char * buf = malloc(size_out);
short *outputBuffer=(short *)buf;
int in_samples = decoded_frame->nb_samples;
int i=0;
float * inputChannel0 = (float *)decoded_frame->extended_data[0];
// Mono
if (pCodecCtx->channels==1)
{
for (i=0; i<in_samples; i++)
{
float sample = *inputChannel0++;
if (sample<-1.0f) sample=-1.0f;
else if (sample>1.0f) sample=1.0f;
outputBuffer[i] = (int16_t) (sample * 32767.0f);//largest positive int16_t
}
}
// Stereo
else
{
float * inputChannel1 = (float *)decoded_frame->extended_data[1];
for (i=0; i < in_samples; i++)
{
float sample = *inputChannel0++;
if (sample<-1.0f) sample=-1.0f;
else if (sample>1.0f) sample=1.0f;
float sample2 = *inputChannel1++;
if (sample2<-1.0f) sample2=-1.0f;
else if (sample2>1.0f) sample2=1.0f;
outputBuffer[i*2] = (int16_t) ((sample) * 32767.0f);
outputBuffer[i*2+1] = (int16_t) ((sample2) * 32767.0f);
}
}
//use buf and size_out here then free the buf
free(buf);
}
}
av_packet_unref (packet_ptr);
if (decoded_frame)
{
av_frame_unref(decoded_frame);
decoded_frame = NULL;
}
}
Hope it helps...