Sur d’autres sites (8456)

• ValueError : I/O operation on closed file when making animation

  3 juillet 2018, par user3851187

  I am using matplotlib and ffmpeg to do some animations. I usually code on a remote server because the code runs faster ; we are having some issues making animations on the remote server. Here is an example of code that works perfectly on my local mac but does not work remotely.

  import matplotlib as mpl
  
  mpl.use('agg')
  
  import matplotlib as mpl 
  
  from matplotlib import animation
  
  import pylab
  
  
  
  def init():
  
      pylab.plot(pylab.arange(10), [0]*10)
  
  
  
  def redraw(frame):
  
      pylab.plot(pylab.arange(10), pylab.arange(10) * frame)
  
  
  
  fig = pylab.figure()
  
  ani = animation.FuncAnimation(fig, redraw, frames=10, interval=1000, init_func=init)
  
  ani.save('animation.mp4')

  I get the animation I want on my local machine (macOS Sierra). When I run it on the remote host (Debian GNU/Linux 8 (jessie)), I get the following error message after 5 frames

  Traceback (most recent call last):
  
    File "animation.py", line 14, in <module>
  
      ani.save('animation.mp4')
  
    File "/usr/local/lib/python2.7/dist-packages/matplotlib/animation.py", line 1200, in save
  
      writer.grab_frame(**savefig_kwargs)
  
    File "/usr/lib/python2.7/contextlib.py", line 35, in __exit__
  
      self.gen.throw(type, value, traceback)
  
    File "/usr/local/lib/python2.7/dist-packages/matplotlib/animation.py", line 241, in saving
  
      self.finish()
  
    File "/usr/local/lib/python2.7/dist-packages/matplotlib/animation.py", line 367, in finish
  
      self.cleanup()
  
    File "/usr/local/lib/python2.7/dist-packages/matplotlib/animation.py", line 405, in cleanup
  
      out, err = self._proc.communicate()
  
    File "/usr/local/lib/python2.7/dist-packages/subprocess32.py", line 724, in communicate
  
      stdout, stderr = self._communicate(input, endtime, timeout)
  
    File "/usr/local/lib/python2.7/dist-packages/subprocess32.py", line 1535, in _communicate
  
      orig_timeout)
  
    File "/usr/local/lib/python2.7/dist-packages/subprocess32.py", line 1591, in _communicate_with_poll
  
      register_and_append(self.stdout, select_POLLIN_POLLPRI)
  
    File "/usr/local/lib/python2.7/dist-packages/subprocess32.py", line 1570, in register_and_append
  
      poller.register(file_obj.fileno(), eventmask)
  
  ValueError: I/O operation on closed file
  
  </module>

  My local machine uses matplotlib version 2.0.0 ; the remote machine uses matplotlib version 2.2.2

  On my local machine I have ffmpeg version 3.2.4

  $ ffmpeg -version
  
  ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
  
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  
  configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.4 --enable-shared - 
  
  -enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables 
  
  --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
  
  libavutil      55. 34.101 / 55. 34.101
  
  libavcodec     57. 64.101 / 57. 64.101
  
  libavformat    57. 56.101 / 57. 56.101
  
  libavdevice    57.  1.100 / 57.  1.100
  
  libavfilter     6. 65.100 /  6. 65.100
  
  libavresample   3.  1.  0 /  3.  1.  0
  
  libswscale      4.  2.100 /  4.  2.100
  
  libswresample   2.  3.100 /  2.  3.100
  
  libpostproc    54.  1.100 / 54.  1.100 

  On the remote host i have ffmpeg version 4.0.1

  ffmpeg -version
  
  ffmpeg version 4.0.1 Copyright (c) 2000-2018 the FFmpeg developers
  
  built with gcc 4.9.2 (Debian 4.9.2-10+deb8u1)
  
  configuration: --prefix=/usr/local
  
  libavutil      56. 14.100 / 56. 14.100
  
  libavcodec     58. 18.100 / 58. 18.100
  
  libavformat    58. 12.100 / 58. 12.100
  
  libavdevice    58.  3.100 / 58.  3.100
  
  libavfilter     7. 16.100 /  7. 16.100
  
  libswscale      5.  1.100 /  5.  1.100
  
  libswresample   3.  1.100 /  3.  1.100

  If I recall correctly I installed ffmpeg locally through homebrew ; I have the anaconda distribution of python. On the remote machine we have the default version of python that comes with Jessie ; I’m not sure how the sysadmin installed ffmpeg.

  I am by no means an expert on ffmpeg, but I have generally never had issues with making animations in matplotlib on my local machine and I would really like to be able to make videos more quickly on the remote machine. Any help would be appreciated !

  Edit
  On the remote machine, the animation works if I use avconv as the writer instead of ffmpeg. I installed avconv locally...which led me to get the same ffmpeg issues locally (probably due to updating shared dependencies). However, I uninstalled ffmpeg and reinstalled it with x264 codec enables Animations in ipython (jupyter) notebook - ValueError : I/O operation on closed file

• What is Google Analytics data sampling and what’s so bad about it ?

  16 août 2019, par Joselyn Khor — Analytics Tips, Development
  What is Google Analytics data sampling, and what’s so bad about it ?

  Google (2019) explains what data sampling is :

  “In data analysis, sampling is the practice of analysing a subset of all data in order to uncover the meaningful information in the larger data set.”[1]

  This is basically saying instead of analysing all of the data, there’s a threshold on how much data is analysed and any data after that will be an assumption based on patterns.

  Google’s (2019) data sampling thresholds :

  Ad-hoc queries of your data are subject to the following general thresholds for sampling :
  [Google] Analytics Standard : 500k sessions at the property level for the date range you are using
  [Google] Analytics 360 : 100M sessions at the view level for the date range you are using (para. 3) [2]

  This threshold is limiting because your data in GA may become more inaccurate as the traffic to your website increases.

  Say you’re looking through all your traffic data from the last year and find you have 5 million page views. Only 500K of that 5 million is accurate ! The data for the remaining 4.5 million (90%) is an assumption based on the 500K sample size.

  This is a key weapon Google uses to sell to large businesses. In order to increase that threshold for more accurate reporting, upgrading to premium Google Analytics 360 for approximately US$150,000 per year seems to be the only choice.

  What’s so bad about data sampling ?

  It’s unfair to say sampled data is to be disregarded completely. There is a calculation ensuring it is representative and can allow you to get good enough insights. However, we don’t encourage it as we don’t just want “good enough” data. We want the actual facts.

  In a recent survey sent to Matomo customers, we found a large proportion of users switched from GA to Matomo due to the data sampling issue.

  The two reasons why data sampling isn’t preferable : 

  1. If the selected sample size is too small, you won’t get a good representative of all the data. 
  2. The bigger your website grows, the more inaccurate your reports will become.

  An example of why we don’t fully trust sampled data is, say you have an ecommerce store and see your GA revenue reports aren’t matching the actual sales data, due to data sampling. In GA you may be seeing revenue for the month as $1 million, instead of actual sales of $800K.

  The sampling here has caused an inaccuracy that could have negative financial implications. What you get in the GA report is an estimated dollar figure rather than the actual sales. Making decisions based on inaccurate data can be costly in this case. 

  Another disadvantage to sampled data is that you might be missing out on opportunities you would’ve noticed if you were given a view of the whole. E.g. not being able to see real patterns occurring due to the data already being predicted. 

  By not getting a chance to see things as they are and only being able to jump to the conclusions and assumptions made by GA is risky. The bigger your business grows, the less you can risk making business decisions based on assumptions that could be inaccurate. 

  If you feel you could be missing out on opportunities because your GA data is sampled data, get 100% accurately reported data. 

  The benefits of 100% accurate data

  Matomo doesn’t use data sampling on any of our products or plans. You get to see all of your data and not a sampled data set.

  Data quality is necessary for high impact decision-making. It’s hard to make strategic changes if you don’t have confidence that your data is reliable and accurate.

  Learn about how Matomo is a serious contender to Google Analytics 360. 

  Now you can import your Google Analytics data directly into your Matomo

  If you’re wanting to make the switch to Matomo but worried about losing all your historic Google Analytics data, you can now import this directly into your Matomo with the Google Analytics Importer tool.

  
  Take the challenge !

  Compare your Google Analytics data (sampled data) against your Matomo data, or if you don’t have Matomo data yet, sign up to our 30-day free trial and start tracking !

  References :

  [1 & 2] About data sampling. (2019). In Analytics Help About data sampling. Retrieved August 14, 2019, from https://support.google.com/analytics/answer/2637192

• libav : encoding/decoding error : get_buffer() failed

  17 août 2015, par MarkB

  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 
  
  #include 
  
  #include 
  
  
  
  #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 
  
  #include 
  
  #include 
  
  
  
  #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;
  
  }