Discussion:
[Libav-user] Bus Error in Audio Encoding
Ramana Jajula
2018-08-29 07:22:29 UTC
Permalink
Hi, I am trying to get back the original audio mp3 file by audio encoding.
I have got the .bin file by doing the audio decoding(.mp3 file) first. Now
I want to get back the .mp3 file by doing audio encoding. Here is the file
for audio decoding my_audio_decoding.c
<https://codepad.co/snippet/NeHLFtzR>.(took
2 args input file and output file). Now I am doing audio encoding, file
below

my_audio_encoding.c

#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>

#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>

#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>

/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const 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(const AVCodec *codec)
{
const int *p;
int best_samplerate = 0;

if (!codec->supported_samplerates)
return 44100;

p = codec->supported_samplerates;
while (*p) {
if (!best_samplerate || abs(44100 - *p) < abs(44100 -
best_samplerate))
best_samplerate = *p;
p++;
}
return best_samplerate;
}

/* select layout with the highest channel count */
static int select_channel_layout(const 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;
}

static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
FILE *output)
{
int ret;

/* send the frame for encoding */
ret = avcodec_send_frame(ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending the frame to the encoder\n");
exit(1);
}

/* read all the available output packets (in general there may be any
* number of them */
while (ret >= 0) {
ret = avcodec_receive_packet(ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}

fwrite(pkt->data, 1, pkt->size, output);
av_packet_unref(pkt);
}
}

int main(int argc, char **argv)
{
const char *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket *pkt;
int i, j, k, ret;
FILE *f;
uint16_t *samples;
float t, tincr;

av_register_all();
avcodec_register_all();

if (argc <= 1) {
fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
return 0;
}
filename = argv[1];

/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP3);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}

c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}

/* put sample parameters */
c->bit_rate = 64000;

/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16P;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %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, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}

/* packet for holding encoded output */
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "could not allocate the packet\n");
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;

/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio data buffers\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++) {
/* make sure the frame is writable -- makes a copy if the encoder
* kept a reference internally */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
samples = (uint16_t*)frame->data[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(c, frame, pkt, f);
}

/* flush the encoder */
encode(c, NULL, pkt, f);

fclose(f);

av_frame_free(&frame);
av_packet_free(&pkt);
avcodec_free_context(&c);

return 0;
}

While I am doing this, I am keep getting bus error, please help me with
this.
Gonzalo Garramuño
2018-08-29 15:57:19 UTC
Permalink
Post by Ramana Jajula
Hi, I am trying to get back the original audio mp3 file by audio
encoding. I have got the .bin file by doing the audio decoding(.mp3
file) first. Now I want to get back the .mp3 file by doing audio
encoding. Here is the file for audio decoding my_audio_decoding.c
<https://codepad.co/snippet/NeHLFtzR>.(took 2 args input file and
           fwrite(pkt->data, 1, pkt->size, output);
It was already pointed to you that you are not using libavformat to do
the encoding as you are relying on stdio functions.
Look for the examples with av_interleaved_write_frame() to do it properly.
--
Gonzalo Garramuño
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