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• SDL2 + ffmpeg2 intermittent clicks instead of audio

  25 janvier 2014, par FertoVordalastr

  Using c++, SDL2, SDL2_mix, ffmpeg2.

  Inited SDL2_mix with callback

  Mix_HookMusic(MusicPlayer, &g_audioPos);

  Decoded audio from ogg to AVFrame* by this code :

  while(av_read_frame(m_formatContext, &m_packet) >= 0)
  
  {
  
      if(m_packet.stream_index == m_audioStream)
  
      {
  
          int audio_frame_finished;
  
          avcodec_decode_audio4(m_audioCodecContext, m_audioFrame, &audio_frame_finished, &m_packet);
  
          if(!audio_frame_finished)
  
          {
  
              continue;
  
          }
  
      }
  
  }

  After this code i have frame with some m_audioFrame->data[0] and m_audioFrame->linesize[0] == 4096

  Once in a while my callback being called :

  void MusicPlayer(void *_udata, Uint8 *_stream, int _len)
  
      {
  
          if(!g_audioData)
  
          {
  
              return;
  
          }
  
  
  
          AudioData* audio = reinterpret_cast(g_audioData);
  
  
  
          if(!audio || audio->Frame->linesize[0] == 0)
  
          {
  
              return;
  
          }
  
  
  
          SDL_memcpy((Uint8*)audio->Data + audio->Pos, (Uint8*)audio->Frame->data[0], audio->Frame->linesize[0]);
  
  
  
          audio->Pos+= audio->Frame->linesize[0];
  
          int rest = g_chunkSize - audio->Pos;//frame->linesize[0];
  
  
  
  
  
          if(rest <= 0)
  
          {
  
              SDL_memcpy(_stream, audio->Data, _len);
  
              audio->Pos = 0;
  
              *(int*)_udata += _len;
  
          }
  
      }

  _len == 8192, so i must push 2 frames to fill stream, but all i get - clicks in my speaker. What am i doing wrong ?

  PS : Tried to reopen MIX with Mix_OpenAudio(m_audioCodecContext->sample_rate, AUDIO_S16SYS, m_audioCodecContext->channels, 4096);. Interesting thing is m_audioCodecContext->channels == 2 and when my callback being called _len = 16384. Have no idea what to do. Please help !!!

• ffmpeg glitch while converting DNG sequence into mp4 video

  21 août 2020, par AnnoyedByAds

  I have two problems with FFmpeg, when I use it to join DNG files sequence into mp4 video file. I also need to downgrade the resolution of the video from 6016x3200 to 2030x1080.
First of all I got almost black screen in the resulting video. Had to play with gamma and brightness options. But it was not enough !

  


  New problems :

  


  


  1. something strange happens with aspect ratio in resulting video file : in the first frame aspect is normal, just like in the original picture, but all the rest frames are getting squeezed. can't figure out why this happen !? (see picture attached).

     


  


  2. colors are desaturated. despite the fact that I set "saturation" option to the maximum value. and also, the first frame of the video is different from the rest (while DNG files are all similar, first is no exception)

     


  


  


  I tried prores codec as well, with the same result.

  


  command I use is simple :

  


  ffmpeg.exe -start_number 1 -i "K:\video\copter_R%5d.dng" -c:v libx264 -vf "fps=25,format=yuv420p, eq=gamma=3.2:brightness=0.2:contrast=1.6:saturation=3, scale=w=2030:h=1080" e:\output.mp4


  


  I tried to use different variants of scale parameter : "scale=-1:1080" as well.

  


  Illustration :

  


  

  


  UPDATE : ffmpeg log report for operation :

  


  https://drive.google.com/file/d/1H6bdpU0Eo4WfR3h-SRtgf7WBNYVFRwz2/view?usp=sharing

  


• Convert audio file using another audio file as template in ffmpeg

  3 décembre 2015, par Iulian Onofrei

  I have some .mp3 audio files, with different "configuration" like sample rate, bit rate, etc.

  For my app, one of them is working and the rest, not.

  How can I convert the rest of them using the working file’s "configuration" ?

  Metadata of two sample files :

  ~/Downloads ❯ ffmpeg -i working.mp3 -i not_working.mp3
  
  ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with Apple LLVM version 7.0.0 (clang-700.1.76)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/2.8.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0x7fd2d380da00] Skipping 0 bytes of junk at 33.
  
  [mp3 @ 0x7fd2d380da00] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from 'working.mp3':
  
    Metadata:
  
      encoder         : Lavf52.64.2
  
    Duration: 00:00:00.65, start: 0.000000, bitrate: 64 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 64 kb/s
  
  [mp3 @ 0x7fd2d4008800] Skipping 0 bytes of junk at 417.
  
  Input #1, mp3, from 'not_working.mp3':
  
    Duration: 00:00:01.83, start: 0.025057, bitrate: 46 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 46 kb/s
  
      Metadata:
  
        encoder         : LAME3.99r