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• FFMPEG merge mp4 file and mp3 file into mp4

  10 avril 2017, par Cường Trần

  I have video file in mp4 format (video.mp4), its length is 20 seconds. From 0 seconds to 10 seconds, the video has sound, and from 10 seconds to 20 seconds, there is no sound.

  I also have mp3 file (audio.mp3) and has length 10 seconds.

  I want to merge video.mp4 and audio.mp3 into result.mp4. The result.mp4 file should have video stream and its audio stream from 01 second to 10 seconds as original and audio stream from 10 seconds to 20 seconds of audio.mp3 as merged.

  I use the command to merge :

  ffmpeg -i video.mp4 -i audio.mp3 -filter_complex "aevalsrc=0:d=10[s1];[s1][1:a]concat=n=2:v=1:a=1[aout]" -c:v copy -map 0:v -map [aout] result.mp4

  But i get the result.mp4 with video : there is no sound from 01-10 seconds, only new sound from 10-20 seconds.

  It is the seem that my command don’t keep the sound from original mp4 file, it has removed it and just keep the new sound.

  Could you please help ?

• FFMPEG- Duration of audio file is inaccurate

  17 septembre 2015, par Tony Than

  I have video file (mp4). I want to detach audio stream (AAC format) from that file and save in PC.
  With below code, Generated aac file canplay now on KM player, but can not play on VLC player. Information of duration displays on player is wrong.
  Please help me with this problem.

   err = avformat_open_input(input_format_context, filename, NULL, NULL);
  
  if (err < 0) {
  
      return err;
  
  }
  
  
  
  /* If not enough info to get the stream parameters, we decode the
  
     first frames to get it. (used in mpeg case for example) */
  
  ret = avformat_find_stream_info(*input_format_context, 0);
  
  if (ret < 0) {
  
      av_log(NULL, AV_LOG_FATAL, "%s: could not find codec parameters\n", filename);
  
      return ret;
  
  }
  
  
  
  /* dump the file content */
  
  av_dump_format(*input_format_context, 0, filename, 0);
  
  
  
  for (size_t i = 0; i < (*input_format_context)->nb_streams; i++) {
  
      AVStream *st = (*input_format_context)->streams[i];
  
      if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
          *input_codec_context = st->codec;
  
          *input_audio_stream = st;
  
  
  
          FILE *file = NULL;
  
          file = fopen("C:\\Users\\MyPC\\Downloads\\Test.aac", "wb");
  
          AVPacket reading_packet;
  
          av_init_packet(&reading_packet);
  
          while (av_read_frame(*input_format_context, &reading_packet) == 0) {
  
              if (reading_packet.stream_index == (int) i) {
  
  
  
                  uint8_t adts_header[7];
  
                  unsigned int obj_type = 0;
  
                  unsigned int num_data_block = (reading_packet.size)/1024;
  
                  int rate_idx = st->codec->sample_rate, channels = st->codec->channels;
  
  
  
                   uint16_t frame_length;
  
  
  
                  // include the header length also
  
                   frame_length = reading_packet.size + 7;
  
  
  
                  /* We want the same metadata */
  
                  /* Generate ADTS header */
  
                  if(adts_header == NULL) return -1;
  
                  /* Sync point over a full byte */
  
                  adts_header[0] = 0xFF;
  
                  /* Sync point continued over first 4 bits + static 4 bits
  
                  * (ID, layer, protection)*/
  
                  adts_header[1] = 0xF1;
  
                  /* Object type over first 2 bits */
  
                  adts_header[2] = obj_type << 6;
  
                  /* rate index over next 4 bits */
  
                  adts_header[2] |= (rate_idx << 2);
  
                  /* channels over last 2 bits */
  
                  adts_header[2] |= (channels & 0x4) >> 2;
  
                  /* channels continued over next 2 bits + 4 bits at zero */
  
                  adts_header[3] = (channels & 0x3) << 6;
  
                  /* frame size over last 2 bits */
  
                  adts_header[3] |= (frame_length & 0x1800) >> 11;
  
                  /* frame size continued over full byte */
  
                  adts_header[4] = (frame_length & 0x1FF8) >> 3;
  
                  /* frame size continued first 3 bits */
  
                  adts_header[5] = (frame_length & 0x7) << 5;
  
                  /* buffer fullness (0x7FF for VBR) over 5 last bits*/
  
                  adts_header[5] |= 0x1F;
  
                  /* buffer fullness (0x7FF for VBR) continued over 6 first bits + 2 zeros
  
                  * number of raw data blocks */
  
                  adts_header[6] = 0xFA;
  
                  adts_header[6] |= num_data_block & 0x03; // Set raw Data blocks.
  
  
  
                  fwrite(adts_header, 1, 7, file);
  
                  fwrite(reading_packet.data, 1, reading_packet.size, file);
  
              }
  
              av_free_packet(&reading_packet);   
  
          }
  
          fclose(file);
  
  
  
          return 0;
  
      }
  
  }

• Channel mapping in FFmpeg conversion of Dolby 7.1 mlp file to 8 channel wav file [closed]

  21 avril 2024, par Stan Duncan

  I have a file with 7.1 audio with the following channel mapping (viewed using MediaInfo) : L R C LFE Ls Rs Lb Rb

  


  I want to convert it to an 8 channel riff64 wave file, so I'm using the following command :

  


  ffmpeg.exe -i "input.mlp" -rf64 auto -c:a pcm_f32le "output.wav"

  


  When I do this, the channel mapping in output.wav changes to (viewed using MediaInfo) : L R C LFE Lb Rb Ls Rs

  


  I'm not sure if this is just a label that gets slapped on there, or if it actually changes the channel positions (i.e., rear surround come out of side surround speakers and vice versa).

  


  Is there a better command I should be using to ensure that the channel mapping is not altered ?