libav: encoding/decoding error: get_buffer() failed - c

Question

I am trying to edit the given Audio encoding/decoding examples of the libav located in libav-11.4/doc/examples/avcodec.c.
I simply want to encode a .wav file into aac and then decode back to .wav
But in the decoding step I allways get the following error:
[aac # 0x...] get_buffer() failed
Error while decoding
The Edited example code:
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif
#include "libavcodec/avcodec.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/imgutils.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f, *outfile;
uint16_t *samples;
float t, tincr;
printf("Audio encoding\n");
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "encoder does not support %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* 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) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, outfile);
av_free_packet(&pkt);
}
}
fclose(f);
fclose(outfile);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Audio decoding\n");
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* 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 */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
}
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
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(c);
av_free(c);
av_frame_free(&decoded_frame);
}
int main(int argc, char **argv)
{
const char *filename;
/* register all the codecs */
avcodec_register_all();
filename = argv[1];
audio_encode_example("test.aac", filename);
audio_decode_example("out.wav", "test.aac");
return 0;
}
EDIT:
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif
#include "libavcodec/avcodec.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/imgutils.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"
#include "libavformat/avformat.h"
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f, *outfile;
uint16_t *samples;
float t, tincr;
printf("Audio encoding\n");
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "encoder does not support %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* 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) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, outfile);
av_free_packet(&pkt);
}
}
fclose(f);
fclose(outfile);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Audio decoding\n");
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* 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);
}
*/
AVFormatContext *s = avformat_alloc_context();
int ret = avformat_open_input(&s, filename, NULL, NULL);
if (ret != 0)
{
abort();
}
// Retrieve stream information
printf("sdfsdf\n");
if(avformat_find_stream_info(s, NULL)<0)
{
printf("Could not find stream info");// Couldn't find stream information
}
// Dump information about file into standard error
av_dump_format(s, 0, filename, 0);
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
//avpkt.data = inbuf;
//avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
int read_ok = av_read_frame(s,&avpkt);
while (read_ok)
{
read_ok = av_read_frame(s,&avpkt);
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
}
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
/*
avpkt.size -= len;
avpkt.data += len;
if (avpkt.size < AUDIO_REFILL_THRESH) {
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(c);
av_free(c);
av_frame_free(&decoded_frame);
}
int main(int argc, char **argv)
{
const char *filename;
/* register all the codecs */
avcodec_register_all();
av_register_all();
filename = argv[1];
audio_encode_example("test.aac", filename);
audio_decode_example("out.wav", "test.aac");
return 0;
}

Answer 1

See this post, you can't use fopen/fread to feed raw AAC data into the decoder, you need to use av_read_frame().