text: /*
   * Copyright (c) 2001 Fabrice Bellard
   *
   * Permission is hereby granted, free of charge, to any person obtaining a copy
   * of this software and associated documentation files (the "Software"), to deal
   * in the Software without restriction, including without limitation the rights
   * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
   * copies of the Software, and to permit persons to whom the Software is
   * furnished to do so, subject to the following conditions:
   *
   * The above copyright notice and this permission notice shall be included in
   * all copies or substantial portions of the Software.
   *
   * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
   * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
   * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
   * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
   * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
   * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
   * THE SOFTWARE.
   */
  
  /**
   * @file
   * audio encoding with libavcodec API example.
   *
   * @example encode_audio.c
   */
  
  #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];
  
      /* register all the codecs */
      avcodec_register_all();
  
      /* find the MP2 encoder */
      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;
  }
  