[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
[phone]
Regards,
Aasim
[phone]

[Carl Eugen Hoyos]

Post by Amir Raza
This is my first mail to ffmpeg , apologies if make some mistakes.

User questions on the development mailing list are not welcome,
as is cross-posting

Post by Amir Raza
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.

You are not writing a wav file, this is visible because you are not
including libavformat header files which would be necessary to
write a wav file.
I suspect you are writing an f32le file which should be playable with
something like:
$ ffplay -f f32le -ac 2 -ar 48k filename.
(Depending on the properties of the aac input file, the example is
for 48kHz stereo)

I thought that adts files cannot be read without libavformat but if
you see an output filesize>0, this is apparently not true.

Carl Eugen