Read a frame one by one from bin file, restore the data to the same fields in AVFrame structure that you have read it from and send it to the encoder.
From: Ramana Jajula
Sent: Thursday, August 23, 2018 2:10 PM
To: libav-***@ffmpeg.org
Subject: Re: [Libav-user] How to get back the original audio file
No, I didn't get clear. I want to pass the decoded output as input, so that I get original audio file(.mp3). I need a file to achieve this. Actually I tried the example programs on ffmpeg and libavcodec,Â
this is my audio encode c file,
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.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;
  if (argc <= 1) {
    fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
    return 0;
  }
  filename = argv[1];
  /* find the MP2 encoder */
  Â
  avcodec_register_all();
  codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
  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_S16;
  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;
}
I am not getting exactly.Â
On Thu, Aug 23, 2018 at 5:03 PM Strahinja Radman <***@gmail.com> wrote:
Completely true, I should have been more clear. I thought that he wanted to play it in VLC or some other media player.
Â
From: Carl Eugen Hoyos
Sent: Thursday, August 23, 2018 1:32 PM
To: This list is about using libavcodec, libavformat, libavutil, libavdevice and libavfilter.
Subject: Re: [Libav-user] How to get back the original audio file
Â
Post by Strahinja RadmanYou basically wrote raw PCM into a plain file. That file is not usable
without the right header.
Â
This is not true, many programs (including FFmpeg) handle raw
PCM audio just fine.
Â
Please do not top-post here, Carl Eugen
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