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• Batch screenshots with FFmpeg

  20 juillet 2017, par cCanuck

  I have this bat file that works with another software and I would like to modify it to work with FFmpeg but I haven’t been able to get the syntax or arguments correct

  I essentially have a series of 1hr videos on an external HDD I want to extract an screenshot every 15mins to folder on my computer. I could do it successfully with FFmpeg from the CMD but not in a bat file.

  @echo off
  
  pause
  
  
  
  FOR /F "tokens=*" %%A IN ('dir E:\BIW\*.asf /b') DO "c:\Program Files (x86)\FrameShots3\fsconsole.exe" -ms 900000 -width 1920 -height 1080 -filetype JPEG -filename BIW_ -filecounter 0000 "E:\BIW\%%A" "C:\timelapse\BIW\"
  
  pause

• When using ffmpeg.exe to create video from sequence of images the result is avi file that show green color why ?

  9 janvier 2014, par user3163653

  In Form1 im taking screenshots of my screen I checked they are ok i save them to the hard disk and see they are ok.

  private void timer1_Tick(object sender, EventArgs e)
  
          {
  
  
  
              using (bitmap = (Bitmap)ScreenCapture.CaptureScreen(true))
  
              {
  
  
  
              Image mScreenImage = new Bitmap(Screen.PrimaryScreen.Bounds.Width,
  
                                       Screen.PrimaryScreen.Bounds.Height);
  
              Graphics g = Graphics.FromImage(bitmap);
  
              g.CopyFromScreen(0, 0, 0, 0, new Size(bitmap.Width, bitmap.Height));
  
              g.Dispose();
  
                  bitmap.Save(@"D:\Testbmp\" + counter.ToString("D6") + ".jpg", ImageFormat.Jpeg);
  
                  counter++;
  
              ffmp.PushFrame((Bitmap)mScreenImage);
  
  
  
              }
  
  
  
          }

  Then in the new class ffmp im using pipe to get the screenshots and build the avi file in real time :

  using System;
  
  using System.Windows.Forms;
  
  using System.Collections.Generic;
  
  using System.Linq;
  
  using System.Text;
  
  using System.Threading.Tasks;
  
  using System.Drawing;
  
  using System.IO.Pipes;
  
  using System.Runtime.InteropServices;
  
  using System.Diagnostics;
  
  using System.IO;
  
  
  
  namespace ScreenVideoRecorder
  
  {
  
      class Ffmpeg
  
      {
  
          NamedPipeServerStream p;
  
          String pipename = "mytestpipe";
  
          System.Diagnostics.Process process;
  
          string ffmpegFileName = "ffmpeg.exe";
  
          string workingDirectory;
  
  
  
          public Ffmpeg()
  
          {
  
              workingDirectory = Path.GetDirectoryName(Application.ExecutablePath);
  
              Logger.Write("workingDirectory: " + workingDirectory);
  
              if (!Directory.Exists(workingDirectory))
  
              {
  
                  Directory.CreateDirectory(workingDirectory);
  
              }
  
              ffmpegFileName = Path.Combine(workingDirectory, ffmpegFileName);
  
              Logger.Write("FfmpegFilename: " + ffmpegFileName);
  
          }
  
  
  
          public void Start(string pathFileName, int BitmapRate)
  
          {
  
              try
  
              {
  
  
  
                  string outPath = pathFileName;
  
                  p = new NamedPipeServerStream(pipename, PipeDirection.Out, 1, PipeTransmissionMode.Byte);
  
  
  
                  ProcessStartInfo psi = new ProcessStartInfo();
  
                  psi.WindowStyle = ProcessWindowStyle.Hidden;
  
                  psi.UseShellExecute = false;
  
                  psi.CreateNoWindow = false;
  
                  psi.FileName = ffmpegFileName;
  
                  psi.WorkingDirectory = workingDirectory;
  
                  psi.Arguments = @"-f rawvideo -pix_fmt yuv420p -video_size 1920x1080 -i \\.\pipe\mytestpipe -map 0 -c:v mpeg4 -r " + BitmapRate + " " + outPath;
  
                  process = Process.Start(psi);
  
                  process.EnableRaisingEvents = false;
  
                  psi.RedirectStandardError = true;
  
                  p.WaitForConnection();
  
              }
  
              catch (Exception err)
  
              {
  
                  Logger.Write("Exception Error: " + err.ToString());
  
              }
  
          }
  
  
  
          public void PushFrame(Bitmap bmp)
  
          {
  
              try
  
              {
  
                  int length;
  
                  // Lock the bitmap's bits.
  
                  //bmp = new Bitmap(1920, 1080);
  
                  Rectangle rect = new Rectangle(0, 0, bmp.Width, bmp.Height);
  
                  //Rectangle rect = new Rectangle(0, 0, 1280, 720);
  
                  System.Drawing.Imaging.BitmapData bmpData =
  
                      bmp.LockBits(rect, System.Drawing.Imaging.ImageLockMode.ReadOnly,
  
                      bmp.PixelFormat);
  
  
  
                  int absStride = Math.Abs(bmpData.Stride);
  
                  // Get the address of the first line.
  
                  IntPtr ptr = bmpData.Scan0;
  
  
  
                  // Declare an array to hold the bytes of the bitmap.
  
                  //length = 3 * bmp.Width * bmp.Height;
  
                  length = absStride * bmpData.Height;
  
                  byte[] rgbValues = new byte[length];
  
  
  
                  //Marshal.Copy(ptr, rgbValues, 0, length);
  
                  int j = bmp.Height - 1;
  
                  for (int i = 0; i < bmp.Height; i++)
  
                  {
  
                      IntPtr pointer = new IntPtr(bmpData.Scan0.ToInt32() + (bmpData.Stride * j));
  
                      System.Runtime.InteropServices.Marshal.Copy(pointer, rgbValues, absStride * (bmp.Height - i - 1), absStride);
  
                      j--;
  
                  }
  
                  p.Write(rgbValues, 0, length);
  
                  bmp.UnlockBits(bmpData);
  
              }
  
              catch(Exception err)
  
              {
  
                  Logger.Write("Error: " + err.ToString());
  
              }
  
  
  
          }
  
  
  
          public void Close()
  
          {
  
              p.Close();
  
          }
  
      }
  
  }

  This is the arguments im using now :

  psi.Arguments = @"-f rawvideo -pix_fmt yuv420p -video_size 1920x1080 -i \\.\pipe\mytestpipe -map 0 -c:v mpeg4 -r " + BitmapRate + " " + outPath;

  Before that I tried to use libx264 instead the mpeg4 but it didn't create the avi file good.
  The problem is that the avi file on the hard disk when I play it show just green color.
  I can play any other avi files on my hard disk but when playing this avi file I see green.

  

  And this is what I see when im playing the avi file :

  

  I tried to use Gspot to find if I need any codec to install and this is what I see :

  

  And I have ffdshow installed already so what should I do now ?

• ffmpeg, opus encoded sound in webm does not work [on hold]

  1er août 2017, par Mockarutan

  I’m having trouble getting Opus encoded sound in the webm container to work. I’m using libopus in ffmpeg.

  The file does work in VLC. But not in ffplay or on YouTube. If I take the raw wav data in a wav file and then convert it to Opus/webm with the ffmpeg.exe that comes pre-compiled. It works in VLC, ffplay and YouTube.

  So ffmpeg can obviously do it correctly, I must be doing something wrong in my code.

  The file my code produces : https://drive.google.com/file/d/0B16rIXjPXJCqcU5HVllIYW1iODg/view?usp=sharing

  Edit, More details that I forgot in my frustration : The file can be opened by ffplay and uploaded to youtube (when I interlace it with VP9 video). But the sound is just "ticks", example : https://www.youtube.com/watch?v=j_ShBbuizeo&feature=youtu.be

  I have read though all example codes that I know of from ffmpeg, but all of them is in the old API, not the send/receive api, so a big part of the code does not apply anymore. This codes works with all other Codes I’ve tested, including H.264+AAC in mp4, VP8+Opus in ogg and raw PCM F32LE in wav. I would have gone with VP8+Opus in ogg if the license was as straight forward as the webm license

  I’ve looked though the source for the ffmpeg.exe command line tool and coped everything applicable in to my code base.

  Here is my code : https://pastebin.com/4c999Uz2

  #include "encoder.h"
  
  #include <algorithm>
  
  #include <iterator>
  
  
  
  extern "C"
  
  {
  
  #include "libavcodec/avcodec.h"
  
  #include "libavdevice/avdevice.h"
  
  #include "libavfilter/avfilter.h"
  
  #include "libavformat/avformat.h"
  
  #include "libavutil/avutil.h"
  
  #include "libavutil/imgutils.h"
  
  #include "libswscale/swscale.h"
  
  #include "libswresample/swresample.h"
  
  
  
      enum InfoCodes
  
      {
  
          ENCODED_VIDEO,
  
          ENCODED_AUDIO,
  
          ENCODED_AUDIO_AND_VIDEO,
  
          NOT_ENOUGH_AUDIO_DATA,
  
      };
  
  
  
      enum ErrorCodes
  
      {
  
          RES_NOT_MUL_OF_TWO = -1,
  
          ERROR_FINDING_VID_CODEC = -2,
  
          ERROR_CONTEXT_CREATION = -3,
  
          ERROR_CONTEXT_ALLOCATING = -4,
  
          ERROR_OPENING_VID_CODEC = -5,
  
          ERROR_OPENING_FILE = -6,
  
          ERROR_ALLOCATING_FRAME = -7,
  
          ERROR_ALLOCATING_PIC_BUF = -8,
  
          ERROR_ENCODING_FRAME_SEND = -9,
  
          ERROR_ENCODING_FRAME_RECEIVE = -10,
  
          ERROR_FINDING_AUD_CODEC = -11,
  
          ERROR_OPENING_AUD_CODEC = -12,
  
          ERROR_INIT_RESMPL_CONTEXT = -13,
  
          ERROR_ENCODING_SAMPLES_SEND = -14,
  
          ERROR_ENCODING_SAMPLES_RECEIVE = -15,
  
          ERROR_WRITING_HEADER = -16,
  
          ERROR_INIT_AUDIO_RESPAMLER = -17,
  
      };
  
  
  
      AVCodecID aud_codec_comp_id = AV_CODEC_ID_OPUS;
  
      AVSampleFormat sample_fmt_comp = AV_SAMPLE_FMT_S16;
  
  
  
      AVCodecID aud_codec_id;
  
      AVSampleFormat sample_fmt;
  
  
  
      char* compressed_cont = "webm";
  
  
  
      AVCodec *aud_codec = NULL;
  
      AVCodecContext *aud_codec_context = NULL;
  
      AVFormatContext *outctx;
  
      AVStream *audio_st;
  
      AVFrame *aud_frame;
  
      SwrContext *audio_swr_ctx;
  
      uint8_t **dst_data = NULL;
  
      AVRational conv_time_base;
  
  
  
      int aud_frame_counter;
  
      int dst_nb_samples, src_nb_samples, max_dst_nb_samples;;
  
      int src_rate, dst_rate;
  
      int dst_nb_channels;
  
      int dst_linesize;
  
  
  
      void write_error_to_file(const char* name, int value)
  
      {
  
          int buf_size = 100;
  
          char* buf = new char[buf_size];
  
  
  
          int ret = av_strerror(value, buf, buf_size);
  
  
  
          FILE *f;
  
          fopen_s(&amp;f, name, "w");
  
          if (f != NULL)
  
          {
  
              if (ret != 0)
  
                  fprintf(f, "Error erroring: , \n", ret);
  
              else
  
                  fprintf(f, "Error code: %s\n", buf);
  
  
  
              fclose(f);
  
          }
  
      }
  
  
  
      void write_value_to_file(const char* name, float value)
  
      {
  
          FILE *f;
  
          fopen_s(&amp;f, name, "w");
  
          if (f != NULL)
  
          {
  
              fprintf(f, "Value: %f\n", value);
  
  
  
              fclose(f);
  
          }
  
      }
  
  
  
      char* concat(const char *s1, const char *s2)
  
      {
  
          char *result = (char*)malloc(strlen(s1) + strlen(s2) + 1);
  
  
  
          strcpy(result, s1);
  
          strcat(result, s2);
  
  
  
          return result;
  
      }
  
  
  
      int setup_audio_codec()
  
      {
  
          aud_codec_id = aud_codec_comp_id;
  
          sample_fmt = sample_fmt_comp;
  
  
  
          // Fixup audio codec
  
          if (aud_codec == NULL)
  
          {
  
              aud_codec = avcodec_find_encoder(aud_codec_id);
  
              avcodec_register(aud_codec);
  
          }
  
  
  
          if (!aud_codec)
  
              return ERROR_FINDING_AUD_CODEC;
  
  
  
          return 0;
  
      }
  
  
  
  
  
      int initialize_audio_stream(AVFormatContext *local_outctx, int sample_rate, int per_frame_audio_samples, int audio_bitrate)
  
      {
  
          aud_codec_context = avcodec_alloc_context3(aud_codec);
  
          if (!aud_codec_context)
  
              return ERROR_CONTEXT_CREATION;
  
  
  
          /* select other audio parameters supported by the encoder */
  
          aud_codec_context->bit_rate = audio_bitrate;
  
          aud_codec_context->sample_rate = sample_rate;
  
          aud_codec_context->sample_fmt = sample_fmt;
  
          aud_codec_context->channel_layout = AV_CH_LAYOUT_STEREO;
  
          aud_codec_context->channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);
  
  
  
          aud_codec_context->codec = aud_codec;
  
          aud_codec_context->codec_id = aud_codec_id;
  
  
  
          AVRational time_base;
  
          time_base.num = per_frame_audio_samples;
  
          time_base.den = aud_codec_context->sample_rate;
  
          aud_codec_context->time_base = time_base;
  
  
  
          int ret = avcodec_open2(aud_codec_context, aud_codec, NULL);
  
  
  
          if (ret &lt; 0)
  
              return ERROR_OPENING_AUD_CODEC;
  
  
  
          local_outctx->audio_codec = aud_codec;
  
          local_outctx->audio_codec_id = aud_codec_id;
  
  
  
          audio_st = avformat_new_stream(local_outctx, aud_codec);
  
  
  
          avcodec_parameters_from_context(audio_st->codecpar, aud_codec_context);
  
  
  
          conv_time_base.num = aud_codec_context->frame_size;
  
          conv_time_base.den = aud_codec_context->sample_rate;
  
  
  
          aud_frame = av_frame_alloc();
  
          aud_frame->nb_samples = aud_codec_context->frame_size;
  
          aud_frame->format = aud_codec_context->sample_fmt;
  
          aud_frame->channel_layout = aud_codec_context->channel_layout;
  
          aud_frame->sample_rate = aud_codec_context->sample_rate;
  
  
  
          int buffer_size;
  
          if (aud_codec_context->frame_size == 0)
  
          {
  
              buffer_size = per_frame_audio_samples * 2 * 4;
  
              aud_frame->nb_samples = per_frame_audio_samples;
  
          }
  
          else
  
          {
  
              buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
  
                  aud_codec_context->sample_fmt, 0);
  
          }
  
  
  
          if (av_sample_fmt_is_planar(sample_fmt))
  
              ret = av_frame_get_buffer(aud_frame, buffer_size / 2);
  
          else
  
              ret = av_frame_get_buffer(aud_frame, buffer_size);
  
  
  
          if (!aud_frame || ret &lt; 0)
  
              return ERROR_ALLOCATING_FRAME;
  
  
  
          audio_swr_ctx = swr_alloc();
  
          if (!audio_swr_ctx)
  
              return ERROR_INIT_AUDIO_RESPAMLER;
  
  
  
          /* set options */
  
          av_opt_set_int(audio_swr_ctx, "in_channel_layout", aud_codec_context->channel_layout, 0);
  
          av_opt_set_int(audio_swr_ctx, "in_sample_rate", sample_rate, 0);
  
          av_opt_set_int(audio_swr_ctx, "in_frame_size", per_frame_audio_samples, 0);
  
          av_opt_set_sample_fmt(audio_swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
  
  
          av_opt_set_int(audio_swr_ctx, "out_channel_layout", aud_codec_context->channel_layout, 0);
  
          av_opt_set_int(audio_swr_ctx, "out_sample_rate", aud_codec_context->sample_rate, 0);
  
          av_opt_set_int(audio_swr_ctx, "out_frame_size", aud_codec_context->frame_size, 0);
  
          av_opt_set_sample_fmt(audio_swr_ctx, "out_sample_fmt", aud_codec_context->sample_fmt, 0);
  
  
  
          /* initialize the resampling context */
  
          if ((ret = swr_init(audio_swr_ctx)) &lt; 0)
  
          {
  
              return ERROR_INIT_AUDIO_RESPAMLER;
  
          }
  
  
  
          dst_rate = aud_codec_context->sample_rate;
  
          src_rate = sample_rate;
  
  
  
          src_nb_samples = per_frame_audio_samples;
  
          dst_nb_samples = aud_codec_context->frame_size;
  
  
  
          max_dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
  
  
  
          dst_nb_channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);
  
  
  
          ret = av_samples_alloc_array_and_samples(&amp;dst_data, &amp;dst_linesize, dst_nb_channels, dst_nb_samples, sample_fmt, 0);
  
  
  
          aud_frame_counter = 0;
  
  
  
          return 0;
  
      }
  
  
  
      int initialize_audio_only_encoding(int sample_rate, int per_frame_audio_samples, int audio_bitrate, const char *filename)
  
      {
  
          int ret;
  
  
  
          avcodec_register_all();
  
          av_register_all();
  
  
  
          outctx = avformat_alloc_context();
  
  
  
          char* full_filename;
  
          char* with_dot = concat(filename, ".");
  
          full_filename = concat(with_dot, compressed_cont);
  
          ret = avformat_alloc_output_context2(&amp;outctx, NULL, compressed_cont, full_filename);
  
  
  
          free(with_dot);
  
  
  
          if (ret &lt; 0)
  
          {
  
              free(full_filename);
  
              return ERROR_CONTEXT_CREATION;
  
          }
  
  
  
          ret = setup_audio_codec();
  
          if (ret &lt; 0)
  
              return ret;
  
  
  
          // Setup Audio
  
          ret = initialize_audio_stream(outctx, sample_rate, per_frame_audio_samples, audio_bitrate);
  
          if (ret &lt; 0)
  
              return ret;
  
  
  
          av_dump_format(outctx, 0, full_filename, 1);
  
  
  
          if (!(outctx->oformat->flags &amp; AVFMT_NOFILE))
  
          {
  
              if (avio_open(&amp;outctx->pb, full_filename, AVIO_FLAG_WRITE) &lt; 0)
  
              {
  
                  free(full_filename);
  
                  return ERROR_OPENING_FILE;
  
              }
  
          }
  
  
  
  
  
          free(full_filename);
  
  
  
          ret = avformat_write_header(outctx, NULL);
  
          if (ret &lt; 0)
  
              return ERROR_WRITING_HEADER;
  
  
  
          return 0;
  
      }
  
  
  
      int process_encode_loop(AVFormatContext *local_outctx, AVCodecContext *codec_context, AVStream *stream, AVRational time_base, bool flush)
  
      {
  
          int ret;
  
  
  
          AVPacket pkt;
  
          av_init_packet(&amp;pkt);
  
          pkt.data = NULL;
  
          pkt.size = 0;
  
  
  
          while (true)
  
          {
  
              ret = avcodec_receive_packet(codec_context, &amp;pkt);
  
              if (!ret)
  
              {
  
                  pkt.stream_index = stream->index;
  
                  av_packet_rescale_ts(&amp;pkt, time_base, stream->time_base);
  
                  av_interleaved_write_frame(local_outctx, &amp;pkt);
  
  
  
                  av_packet_unref(&amp;pkt);
  
              }
  
  
  
              if (ret == AVERROR(EAGAIN))
  
                  break;
  
              else if (ret == AVERROR_EOF)
  
                  break;
  
              else if (ret &lt; 0)
  
                  return ERROR_ENCODING_FRAME_RECEIVE;
  
              else if (flush == false)
  
                  break;
  
          }
  
  
  
          return 0;
  
      }
  
  
  
      int write_audio_frame(float_t *aud_sample)
  
      {
  
          int ret;
  
          if (dst_nb_samples > max_dst_nb_samples)
  
          {
  
              av_free(&amp;aud_frame->data[0]);
  
              ret = av_samples_alloc(aud_frame->data, &amp;dst_linesize, dst_nb_channels, dst_nb_samples, sample_fmt, 1);
  
              if (ret &lt; 0)
  
                  return ERROR_INIT_AUDIO_RESPAMLER;
  
  
  
              max_dst_nb_samples = dst_nb_samples;
  
          }
  
  
  
          ret = swr_convert(audio_swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)&amp;aud_sample, src_nb_samples);
  
          if (ret &lt; 0)
  
          {
  
              return ERROR_INIT_AUDIO_RESPAMLER;
  
          }
  
  
  
          aud_frame->data[0] = (uint8_t*)dst_data[0];
  
          aud_frame->extended_data[0] = (uint8_t*)dst_data[0];
  
  
  
          aud_frame->pts = aud_frame_counter++;
  
  
  
          ret = avcodec_send_frame(aud_codec_context, aud_frame);
  
  
  
          if (ret &lt; 0)
  
          {
  
              write_error_to_file("AVERROR.txt", ret);
  
  
  
              return ERROR_ENCODING_SAMPLES_SEND;
  
          }
  
  
  
          ret = process_encode_loop(outctx, aud_codec_context, audio_st, conv_time_base, false);
  
  
  
          if (ret &lt; 0)
  
              return ERROR_ENCODING_SAMPLES_RECEIVE;
  
  
  
          return ENCODED_AUDIO;
  
      }
  
  
  
      int finish_audio_encoding()
  
      {
  
          //fclose(dst_file);
  
  
  
          AVPacket pkt;
  
          av_init_packet(&amp;pkt);
  
          pkt.data = NULL;
  
          pkt.size = 0;
  
  
  
          fflush(stdout);
  
  
  
          int ret = avcodec_send_frame(aud_codec_context, NULL);
  
          if (ret &lt; 0)
  
              return ERROR_ENCODING_FRAME_SEND;
  
  
  
          while (true)
  
          {
  
              ret = avcodec_receive_packet(aud_codec_context, &amp;pkt);
  
              if (!ret)
  
              {
  
                  if (pkt.pts != AV_NOPTS_VALUE)
  
                      pkt.pts = av_rescale_q(pkt.pts, aud_codec_context->time_base, audio_st->time_base);
  
                  if (pkt.dts != AV_NOPTS_VALUE)
  
                      pkt.dts = av_rescale_q(pkt.dts, aud_codec_context->time_base, audio_st->time_base);
  
  
  
                  av_write_frame(outctx, &amp;pkt);
  
                  av_packet_unref(&amp;pkt);
  
              }
  
              if (ret == -AVERROR(AVERROR_EOF))
  
                  break;
  
              else if (ret &lt; 0)
  
                  return ERROR_ENCODING_FRAME_RECEIVE;
  
          }
  
  
  
          av_write_trailer(outctx);
  
  
  
          return 0;
  
      }
  
  
  
      void cleanup()
  
      {
  
          if (aud_frame)
  
          {
  
              av_frame_free(&amp;aud_frame);
  
          }
  
          if (outctx)
  
          {
  
              for (int i = 0; i &lt; outctx->nb_streams; i++)
  
                  av_freep(&amp;outctx->streams[i]);
  
  
  
              avio_close(outctx->pb);
  
              av_free(outctx);
  
          }
  
  
  
          if (aud_codec_context)
  
          {
  
              avcodec_close(aud_codec_context);
  
              av_free(aud_codec_context);
  
          }
  
      }
  
  
  
      void fill_samples(float_t *dst, int nb_samples, int nb_channels, int sample_rate, float_t *t)
  
      {
  
          int i, j;
  
          float_t tincr = 1.0 / sample_rate;
  
          const float_t c = 2 * M_PI * 440.0;
  
          /* generate sin tone with 440Hz frequency and duplicated channels */
  
          for (i = 0; i &lt; nb_samples; i++) {
  
              *dst = sin(c * *t);
  
              for (j = 1; j &lt; nb_channels; j++)
  
                  dst[j] = dst[0];
  
              dst += nb_channels;
  
              *t += tincr;
  
          }
  
      }
  
  
  
      int main()
  
      {
  
          int width = 1920;
  
          int height = 1080;
  
          int frame_rate = 30;
  
          float t = 0, tincr = 0, tincr2 = 0;
  
  
  
          int sec = 20;
  
  
  
          int tot = 3 * width * height;
  
          uint8_t* rgb24Data = new uint8_t[tot];
  
          float_t** aud_samples_planar;
  
          int16_t** aud_samples_s16;
  
          int src_samples_linesize;
  
          int src_nb_samples = 1024;
  
          int src_channels = 2;
  
          int sample_rate = 48000;
  
  
  
          uint8_t **src_data = NULL;
  
  
  
          int ret;
  
  
  
          initialize_audio_only_encoding(48000, src_nb_samples, 256000, "only_sound");
  
  
  
          ret = av_samples_alloc_array_and_samples(&amp;src_data, &amp;src_samples_linesize, src_channels,
  
              src_nb_samples, AV_SAMPLE_FMT_FLT, 0);
  
  
  
  
  
          for (size_t i = 0; i &lt; frame_rate * sec; i++)
  
          {
  
              fill_samples((float *)src_data[0], src_nb_samples, src_channels, sample_rate, &amp;t);
  
              write_audio_frame((float *)src_data[0]);
  
          }
  
  
  
          finish_audio_encoding();
  
  
  
          cleanup();
  
  
  
          return 0;
  
      }
  
  }
  
  </iterator></algorithm>

  Any any suggestion is appreciated, thanks in Advance !

  Some things I tried :

  I’ve looked at the header with the "MediaInfo" app built in to MVKTool :
  https://i.gyazo.com/3b29b41629a28bd526bf7637ce3f2601.png
  It all looks fine to me.

  I’ve also inspected the raw EBML file with EBML-Viewer (https://code.google.com/archive/p/ebml-viewer/) and in there I can se some difference between the files ;

  My file : https://i.gyazo.com/6fa8c540a2698a8a4d3421d363aede0a.png
  File produced with ffmpeg.exe : https://i.gyazo.com/04d60e64ff3c3040ea83e98cdf507530.png

  In my file it’s "Cluster" -> "BlockGroup" -> "Block", " ?"
  In the other it’s just "Cluster" -> "SimpleBlock"
  And in the webm specs, it says both are supported (https://www.webmproject.org/docs/container/)

  But I do not know much about these specific things, just looking for anything.