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• How To Specify FFMPEG Input Volume For Multple Input Files

  30 septembre 2014, par gbd

  I have 6 similar/separate audio input files feeding into ffmpeg - no video. I’m mixing the 6 input channels down to a stereo output using amix and that works fine. But now I need to change the volume of each individual input channel before the mix down - or maybe as part of the mix down. I’ve looked at and tried aeval (which seems very slow) in the form

   'aeval=val(0)*volChg1:c=same|val(1)*volChg2:c=same|val(2)*volChg3:c=same|val(3)*volChg4:c=same|val(4)*volChg5:c=same|val(5)*volChg6:c=same'

  but that only seems to change the volume of the first channel.

  So the whole input part of my ffmpeg expression looks something like this right now :

  -i inFilePath1 -i inFilePath2 -i inFilePath3 -i inFilePath4 -i inFilePath5 -i inFilePath6 
  
  -filter_complex 'aeval=val(0)*$volChg1:c=same|val(1)*$volChg2:c=same|val(2)*$volChg3:c=same|val(3)*$volChg4:c=same|val(4)*$volChg5:c=same|val(5)*$volChg6:c=same','amix=inputs=6:duration=longest'

  $volChg1-6 are php variables containing the individual volume multipliers - 0.5 for example should halve the volume of that individual channel while 1.0 would leave it unaffected. How do I do this ?

• With ffmpeg's image to movie feature, is it possible to pass in frames over a period of time versus all at once ?

  2 novembre 2013, par Zack Yoshyaro

  For the purpose of making a time lapse recording of desktop activity, it is possible so "stream" the frame list to ffmpeg over time, rather than all at once in the beginning.

  Currently, it is a two step process.

  1. save individual snapshots to disc

     im = ImageGrab.grab()
     im.save("frame_%s.jpg" % count, 'jpg')

  2. compile those snapshots with ffmpeg via

     ffmpeg -r 1 -pattern_type glob -i '*.jpg' -c:v libx264 out.mp4

  It would be nice if there were a way to merge the two steps so that I'm not flooding my hard drive with thousands of individual snapshots. Is it possible to do this ?

• Duplicate a signal with FFmpeg library [on hold]

  22 septembre 2016, par Meugiwara

  I’m using the FFmpeg library to develop a program in C/C++. At this time, my aim is to duplicate the signal when I open an audio file. My scheme is the next : read an audio file, transform the stereo signal in mono signal and duplicate this mono signal to get two mono signal exactly the same.

  I watched in the Doxygen documentation on the official website, but I don’t find something interesting. Do someone know a good way to do this ?

  Here’s what I have so far :

  #include 
  
  #include 
  
  #include 
  
  
  
  #define __STDC_CONSTANT_MACROS
  
  
  
  #ifdef _WIN32
  
  
  
  extern "C" /* Windows */
  
  {
  
      #include "libswresample/swresample.h"
  
      #include "libavformat/avformat.h"
  
      #include "libavcodec/avcodec.h"
  
      #include "SDL2/SDL.h"
  
  };
  
  
  
  #else
  
  
  
  #ifdef __cplusplus
  
  
  
  extern "C" /* Linux */
  
  {
  
      #endif /* __cplusplus */
  
      #include <libswresample></libswresample>swresample.h>
  
      #include <libavformat></libavformat>avformat.h>
  
      #include <libavcodec></libavcodec>avcodec.h>
  
      #include <sdl2></sdl2>SDL.H>
  
      #ifdef __cplusplus
  
  };
  
  
  
  #endif /* __cplusplus */
  
  
  
  #endif /* _WIN32 */
  
  
  
  #define MAX_AUDIO_FRAME_SIZE 192000 /* 1 second of 48 kHz 32 bits audio*/
  
  #define OUTPUT_PCM 1 /* Output PCM */
  
  #define USE_SDL 1 /* Use SDL */
  
  
  
  /* Les incrémentations */
  
  static Uint32 audio_len;
  
  static Uint8 *audio_chunk;
  
  static Uint8 *audio_pos;
  
  
  
  void    fill_audio(void *udata, Uint8 *stream, int len)
  
  {
  
      SDL_memset(stream, 0, len); /* SDL 2.0 */
  
      if (audio_len == 0)
  
          return;
  
      len = (len > audio_len ? audio_len : len);
  
      SDL_MixAudio(stream, audio_pos, len, SDL_MIX_MAXVOLUME);
  
      audio_pos = audio_pos + len;
  
      audio_len = audio_len - len;
  
  }
  
  
  
  int     reading_sound(char *str)
  
  {
  
      SDL_AudioSpec       wanted_spec;
  
      AVFormatContext     *pFormatCtx;
  
      AVCodecContext      *pCodecCtx;
  
      AVSampleFormat      out_sample_fmt;
  
      AVPacket            *packet;
  
      AVFrame             *pFrame;
  
      AVCodec             *pCodec;
  
      uint64_t            out_channel_layout;
  
      uint8_t             *out_buffer;
  
      int64_t             in_channel_layout;
  
      struct SwrContext   *au_convert_ctx;
  
      int                 out_sample_rate;
  
      int                 out_buffer_size;
  
      int                 out_nb_samples;
  
      int                 out_channels;
  
      int                 audioStream;
  
      int                 got_picture;
  
      int                 index = 0;
  
      int                 ret;
  
      int                 i;
  
      FILE                *pFile = NULL;
  
      /*
  
      //char              *sound = "WavinFlag.aac";
  
      */
  
  
  
      av_register_all();
  
      avformat_network_init();
  
      pFormatCtx = avformat_alloc_context();
  
      if (avformat_open_input(&amp;pFormatCtx, str, NULL, NULL) != 0) /* Ouverture du fichier */
  
      {
  
          fprintf(stderr, "%s\n", "Couldn't open input stream");
  
          return (-1);
  
      }
  
      if (avformat_find_stream_info(pFormatCtx, NULL) &lt; 0) /* Récupérer les informations */
  
      {
  
          fprintf(stderr, "Couldn't find stream information\n");
  
          return (-1);
  
      }
  
      av_dump_format(pFormatCtx, 0, str, false); /* Envoyer les informations utiles sur la sortie d'erreur */
  
      audioStream = -1;
  
      for (i = 0; i &lt; pFormatCtx->nb_streams; i++) /* Trouver le début du son */
  
          if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
  
          {
  
              audioStream = i;
  
              break;
  
          }
  
      if (audioStream == -1)
  
      {
  
          fprintf(stderr, "%s\n", "Didn't find an audio stream");
  
          return (-1);
  
      }
  
      pCodecCtx = pFormatCtx->streams[audioStream]->codec;
  
      pCodec = avcodec_find_decoder(pCodecCtx->codec_id); /* Trouver le décodeur du fichier (.aac, .mp3, ...) */
  
      if (pCodec == NULL)
  
      {
  
          fprintf(stderr, "%s\n", "Codec not found");
  
          return (-1);
  
      }
  
      if (avcodec_open2(pCodecCtx, pCodec, NULL) &lt; 0) /* Ouvrir le décodeur du fichier */
  
      {
  
          fprintf(stderr, "%s\n", "Couldn't open codec");
  
          return (-1);
  
      }
  
  
  
      #if OUTPUT_PCM
  
  
  
          pFile = fopen("output.pcm", "wb"); /* Créer et écrire tout ce qui se passe dans ce fichier */
  
  
  
      #endif /* OUTPUT_PCM */
  
  
  
      packet = (AVPacket *)av_malloc(sizeof(AVPacket)); /* Allouer taille */
  
      av_init_packet(packet);
  
      out_channel_layout = AV_CH_LAYOUT_MONO; /* Canaux de sortie */
  
      out_nb_samples = pCodecCtx->frame_size; /* L'échantillonnage - Le nombre de samples*/
  
      out_sample_fmt = AV_SAMPLE_FMT_S16; /* Format */
  
      out_sample_rate = 44100; /* Fréquence */
  
      out_channels = av_get_channel_layout_nb_channels(out_channel_layout); /* Récupérer le nombre de canaux */
  
      /*
  
      // printf("%d\n", out_channels);
  
      // system("PAUSE");
  
      */
  
      out_buffer_size = av_samples_get_buffer_size(NULL, out_channels, out_nb_samples, out_sample_fmt, 1); /* Taille du buffer */
  
      out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE * 2); /* Allouer taille */
  
      pFrame = av_frame_alloc(); /* Allouer taille */
  
  
  
      #if USE_SDL
  
  
  
          if (SDL_Init(SDL_INIT_VIDEO |SDL_INIT_AUDIO | SDL_INIT_TIMER)) /* Initialiser SDL */
  
          {
  
              fprintf(stderr, "%s - %s\n", "Couldn't initialize SDL", SDL_GetError());
  
              return (-1);
  
          }
  
          /*
  
          // Attribution des valeurs avec les variables au-dessus
  
          */
  
          wanted_spec.freq = out_sample_rate;
  
          wanted_spec.format = AUDIO_S16SYS;
  
          wanted_spec.channels = out_channels;
  
          wanted_spec.silence = 0;
  
          wanted_spec.samples = out_nb_samples;
  
          wanted_spec.callback = fill_audio;
  
          wanted_spec.userdata = pCodecCtx;
  
          if (SDL_OpenAudio(&amp;wanted_spec, NULL) &lt; 0)
  
          {
  
              fprintf(stderr, "%s\n", "Can't open audio");
  
              return (-1);
  
          }
  
  
  
      #endif /* USE_SDL */
  
  
  
      in_channel_layout = av_get_default_channel_layout(pCodecCtx->channels);
  
      au_convert_ctx = swr_alloc();
  
      au_convert_ctx = swr_alloc_set_opts(au_convert_ctx, out_channel_layout, out_sample_fmt, out_sample_rate,
  
      in_channel_layout, pCodecCtx->sample_fmt, pCodecCtx->sample_rate, 0, NULL);
  
      swr_init(au_convert_ctx);
  
      while (av_read_frame(pFormatCtx, packet) >= 0) /* Lecture du fichier */
  
      {
  
          if (packet->stream_index == audioStream)
  
          {
  
              ret = avcodec_decode_audio4(pCodecCtx, pFrame, &amp;got_picture, packet); /* Décoder les packets */
  
              if (ret &lt; 0)
  
              {
  
                  fprintf(stderr, "%s\n", "Error in decoding audio frame");
  
                  return (-1);
  
              }
  
              if (got_picture > 0)
  
              {
  
                  swr_convert(au_convert_ctx, &amp;out_buffer, MAX_AUDIO_FRAME_SIZE, (const uint8_t **)pFrame->data, pFrame->nb_samples);
  
  
  
      #if 1
  
  
  
                  printf("Index : %5d\t Points : %lld\t Packet size : %d\n", index, packet->pts, packet->size); /* Affichage des informations sur la console */
  
  
  
      #endif /* 1 */
  
  
  
      #if OUTPUT_PCM
  
  
  
                  fwrite(out_buffer, 1, out_buffer_size, pFile); /* Faire l'écriture dans le fichier */
  
  
  
      #endif /* OUTPUT_PCM */
  
  
  
                  index = index + 1;
  
              }
  
  
  
      #if USE_SDL
  
  
  
              while (audio_len > 0)
  
                  SDL_Delay(1);
  
              audio_chunk = (Uint8 *)out_buffer;
  
              audio_len = out_buffer_size;
  
              audio_pos = audio_chunk;
  
              SDL_PauseAudio(0);
  
  
  
      #endif /* USE_SDL */
  
  
  
          }
  
          av_free_packet(packet); /* Libérer les packets */
  
      }
  
      swr_free(&amp;au_convert_ctx); /* Libérer la taille de conversion */
  
  
  
      #if USE_SDL
  
  
  
      SDL_CloseAudio(); /* Arrêter le son dans SDL */
  
      SDL_Quit(); /* Quitter SDL */
  
  
  
      #endif /* USE_SDL */
  
  
  
      #if OUTPUT_PCM
  
  
  
      fclose(pFile); /* Fermer le fichier d'écriture */
  
  
  
      #endif /* OUTPUT_PCM */
  
  
  
      av_free(out_buffer);
  
      avcodec_close(pCodecCtx); /* Fermer le décodeur */
  
      avformat_close_input(&amp;pFormatCtx); /* Fermer le fichier lu */
  
      return (0);
  
  }
  
  
  
  int     main(int argc, char **argv)
  
  {
  
  
  
      if (argc != 2)
  
      {
  
          fprintf(stderr, "%s\n", "Usage : ./BASELFI [File]");
  
          system("PAUSE");
  
          return (-1);
  
      }
  
      else
  
      {
  
          reading_sound(argv[1]);
  
          return (0);
  
      }
  
      return (0);
  
  }