如何使用C语言中的libav将Binary（.bin）文件转换为mp3？

Question

我已经使用my_audio_decode.c创建了一个bin文件。如何通过编码找回原始的mp3文件？

P.S。我已经完成了ffmpeg代码，并已按照编解码器的要求修改为mp3等。您可以在下面看到修改后的代码。

#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;
}

但是代码令人困惑，并且仅使用一个输入作为参数，我需要一个清晰的工作示例c代码，例如它必须使用2个arg，例如输入文件和输出文件。