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• FFMPEG concat demuxer - how to make file formats compatible ?

  10 avril 2015, par user206481

  I need to automate mp4 concatenation server-side and I’m using FFMPEG. I will get uploads of mp4 files and I want to attach a Title.mp4 and End.mp4 to each one. I am also overlaying a soundtrack (the input videos do not have sound) There is a potential high server load so I’d like to do it as efficiently as possible using ffmpeg’s concat demuxer to avoid re-encoding the video.

  After receiving samples of each file, I am not successful and I believe it is due to mismatched file formats. My result has good Title.mp4 and audio, then when the sample uploaded mp4 is supposed to play there is garbled green/pink/red pixels on the top half of the video, then the End.mp4 plays fine. Here is my ffmpeg command and output :

  $ ffmpeg -f concat -i <(printf "file '%s'\n" Title.mp4 Sample.mp4 End.mp4) -i SoundTrack.wav -c:v copy -strict -2 -y Out.mp4
  
  
  
  ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers
  
  built with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-55)
  
  configuration: --prefix=/home/dpmsmobi/ffmpeg_build --extra-cflags=-I/home/dpmsmobi/ffmpeg_build/include --extra-ldflags=-L/home/dpmsmobi/ffmpeg_build/lib --bindir=/home/dpmsmobi/bin --enable-gpl --enable-nonfree --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264
  
    libavutil      54. 20.100 / 54. 20.100
  
    libavcodec     56. 26.100 / 56. 26.100
  
    libavformat    56. 25.101 / 56. 25.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.102 /  5. 11.102
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, concat, from '/dev/fd/63':
  
    Duration: N/A, start: 0.000000, bitrate: 1810 kb/s
  
      Stream #0:0: Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 768x512 [SAR 1:1 DAR 3:2], 1810 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc
  
  Guessed Channel Layout for  Input Stream #1.0 : stereo
  
  Input #1, wav, from 'SoundTrack.wav':
  
    Metadata:
  
      encoded_by      : Adobe Premiere Pro CC 2014 (Maci
  
      encoder         : Adobe Premiere Pro CC 2014 (Macintosh)
  
      date            : 2015-04-07
  
      creation_time   : 11:12:10
  
      time_reference  : 0
  
    Duration: 00:00:15.06, bitrate: 1551 kb/s
  
      Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s
  
  Output #0, mp4, to 'Out.mp4':
  
    Metadata:
  
      encoder         : Lavf56.25.101
  
      Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 768x512 [SAR 1:1 DAR 3:2], q=2-31, 1810 kb/s, 30 fps, 30 tbr, 30k tbn, 30k tbc
  
      Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.26.100 aac
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (native))
  
  Press [q] to stop, [?] for help
  
  [concat @ 0x1dedc20] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
  
  [concat @ 0x1dedc20] DTS 69750 < 91000 out of order
  
  [mp4 @ 0x1f75060] Non-monotonous DTS in output stream 0:0; previous: 91000, current: 69750; changing to 91001. This may result in incorrect timestamps in the output file.
  
  
  
  <----- many more Non-monotonous DTS messages omitted here ---->
  
  
  
  frame=  427 fps=0.0 q=-1.0 Lsize=    4123kB time=00:00:15.06 bitrate=2242.5kbits/s    
  
  video:3873kB audio:236kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.344173%

  I can successfully concatenate the Title.mp4 to the End.mp4, and I can successfully concatenate two Sample.mp4 files, so I know I’ve got the ffmpeg command right. I can also successfully concat the files using the following ffmpeg command with filter_complex instead of concat demuxer (this takes considerably longer due to re-encoding) :

  ffmpeg -i Title.mp4 -i Sample.mp4 -i End.mp4 -i SoundTrack.wav -filter_complex '[0:0] [1:0] [2:0] concat=n=3:v=1 [v]' -map '[v]' -map 3:0 -crf 20 -strict -2 -y Out2.mp4 

  Here is the MediaInfo output for each type of mp4 file :

  $ mediainfo Title.mp4
  
  General
  
  Complete name                            : Title.mp4
  
  Format                                   : MPEG-4
  
  Format profile                           : Base Media / Version 2
  
  Codec ID                                 : mp42
  
  File size                                : 693 KiB
  
  Duration                                 : 3s 100ms
  
  Overall bit rate mode                    : Variable
  
  Overall bit rate                         : 1 831 Kbps
  
  Encoded date                             : UTC 2015-04-07 19:15:03
  
  Tagged date                              : UTC 2015-04-07 19:15:03
  
  ©TIM                                     : 00:00:00:00
  
  ©TSC                                     : 30
  
  ©TSZ                                     : 1
  
  
  
  Video
  
  ID                                       : 1
  
  Format                                   : AVC
  
  Format/Info                              : Advanced Video Codec
  
  Format profile                           : Main@L3.1
  
  Format settings, CABAC                   : Yes
  
  Format settings, ReFrames                : 3 frames
  
  Codec ID                                 : avc1
  
  Codec ID/Info                            : Advanced Video Coding
  
  Duration                                 : 3s 100ms
  
  Bit rate mode                            : Variable
  
  Bit rate                                 : 1 811 Kbps
  
  Maximum bit rate                         : 3 000 Kbps
  
  Width                                    : 768 pixels
  
  Height                                   : 512 pixels
  
  Display aspect ratio                     : 3:2
  
  Frame rate mode                          : Constant
  
  Frame rate                               : 30.000 fps
  
  Standard                                 : NTSC
  
  Color space                              : YUV
  
  Chroma subsampling                       : 4:2:0
  
  Bit depth                                : 8 bits
  
  Scan type                                : Progressive
  
  Bits/(Pixel*Frame)                       : 0.154
  
  Stream size                              : 685 KiB (99%)
  
  Language                                 : English
  
  Encoded date                             : UTC 2015-04-07 19:15:03
  
  Tagged date                              : UTC 2015-04-07 19:15:03
  
  Color range                              : Limited
  
  
  
  $ mediainfo Sample.mp4
  
  General
  
  Complete name                            : Sample.mp4
  
  Format                                   : MPEG-4
  
  Format profile                           : Base Media
  
  Codec ID                                 : isom
  
  File size                                : 2.93 MiB
  
  Duration                                 : 7s 9ms
  
  Overall bit rate                         : 3 505 Kbps
  
  Encoded date                             : UTC 1970-01-01 00:00:00
  
  Tagged date                              : UTC 1970-01-01 00:00:00
  
  Writing application                      : Lavf52.64.2
  
  
  
  Video
  
  ID                                       : 1
  
  Format                                   : AVC
  
  Format/Info                              : Advanced Video Codec
  
  Format profile                           : Baseline@L3.1
  
  Format settings, CABAC                   : No
  
  Format settings, ReFrames                : 1 frame
  
  Format settings, GOP                     : M=1, N=30
  
  Codec ID                                 : avc1
  
  Codec ID/Info                            : Advanced Video Coding
  
  Duration                                 : 7s 9ms
  
  Bit rate                                 : 3 500 Kbps
  
  Width                                    : 768 pixels
  
  Height                                   : 512 pixels
  
  Display aspect ratio                     : 3:2
  
  Frame rate mode                          : Variable
  
  Frame rate                               : 30.250 fps
  
  Minimum frame rate                       : 23.462 fps
  
  Maximum frame rate                       : 296.053 fps
  
  Color space                              : YUV
  
  Chroma subsampling                       : 4:2:0
  
  Bit depth                                : 8 bits
  
  Scan type                                : Progressive
  
  Bits/(Pixel*Frame)                       : 0.294
  
  Stream size                              : 2.92 MiB (100%)
  
  Language                                 : English
  
  Encoded date                             : UTC 1970-01-01 00:00:00
  
  Tagged date                              : UTC 1970-01-01 00:00:00

  I’m pretty sure it’s the mp42 vs isom Codec ID’s, and potentially the constant vs variable frame rates. I can’t change the input mp4’s but I know their format will stay the same. How can I reformat the Title and End mp4’s to match the input mp4 files so I can use ffmpeg concat demux ?

• FFmpeg C API - syncing video and audio

  10 novembre 2015, par Justin Bradley

  I am trimming video and having a hard getting the audio to sync correctly. The code below is as close as I’ve gotten it work. I’ve tried both re-encoding and not re-encoding the output streams.

  The video trims correctly and is written to the output container. The audio stream also trims correctly, but is written to the front of the output container. For example if the trim length is 10s - the correct portion of audio plays for 10s and then the correct portion of video plays.

  //////// audio stream ////////
  
  const AVStream *input_stream_audio = input_container->streams[audio_stream_index];
  
  const AVCodec *decoder_audio = avcodec_find_decoder(input_stream_audio->codec->codec_id);
  
  if(!decoder_audio) {
  
      cleanup(decoded_packet, output_container, decoded_frame);
  
      avformat_close_input(&input_container);
  
      LOGE("=> Audio decoder not found");
  
      return -1;
  
  }
  
  if(avcodec_open2(input_stream_audio->codec, decoder_audio, NULL) < 0) {
  
      cleanup(decoded_packet, output_container, decoded_frame);
  
      avformat_close_input(&input_container);
  
      LOGE("=> Error opening audio decoder");
  
      return -1;
  
  }
  
  
  
  AVStream *output_stream_audio = avformat_new_stream(output_container, NULL);
  
  if(avcodec_copy_context(output_stream_audio->codec, input_stream_audio->codec) != 0){
  
      LOGE("=> Failed to Copy audio Context ");
  
      return -1;
  
  }
  
  else {
  
      LOGI("=> Copied audio context ");
  
      output_stream_audio->codec->codec_id = input_stream_audio->codec->codec_id;
  
      output_stream_audio->codec->codec_tag = 0;
  
      output_stream_audio->pts = input_stream_audio->pts;
  
      output_stream_audio->time_base.num = input_stream_audio->time_base.num;
  
      output_stream_audio->time_base.den = input_stream_audio->time_base.den;
  
  
  
  }
  
  
  
  if(avio_open(&output_container->pb, output_file, AVIO_FLAG_WRITE) < 0) {
  
      cleanup(decoded_packet, output_container, decoded_frame);
  
      avformat_close_input(&input_container);
  
      LOGE("=> Error opening output file");
  
      return -1;
  
  }
  
  
  
  // allocate frame for conversion
  
  decoded_frame = avcodec_alloc_frame();
  
  if(!decoded_frame) {
  
      cleanup(decoded_packet, output_container, decoded_frame);
  
      avformat_close_input(&input_container);
  
      LOGE("=> Error allocating frame");
  
      return -1;
  
  }
  
  
  
  av_dump_format(input_container, 0, input_file, 0);
  
  avformat_write_header(output_container, NULL);
  
  av_init_packet(&decoded_packet);
  
  
  
  decoded_packet.data = NULL;
  
  decoded_packet.size = 0;
  
  int current_frame_num = 1;
  
  int current_frame_num_audio = 1;
  
  int got_frame, len;
  
  
  
  AVRational default_timebase;
  
  default_timebase.num = 1;
  
  default_timebase.den = AV_TIME_BASE;
  
  
  
  int64_t starttime_int64 = av_rescale_q((int64_t)( 12.0 * AV_TIME_BASE ), AV_TIME_BASE_Q, input_stream->time_base);
  
  int64_t endtime_int64 = av_rescale_q((int64_t)( 18.0 * AV_TIME_BASE ), AV_TIME_BASE_Q, input_stream->time_base);
  
  LOGI("=> starttime_int64:     %" PRId64, starttime_int64);
  
  LOGI("=> endtime_int64:       %" PRId64, endtime_int64);
  
  
  
  int64_t starttime_int64_audio = av_rescale_q((int64_t)( 12.0 * AV_TIME_BASE ), AV_TIME_BASE_Q, input_stream_audio->time_base);
  
  int64_t endtime_int64_audio = av_rescale_q((int64_t)( 18.0 * AV_TIME_BASE ), AV_TIME_BASE_Q, input_stream_audio->time_base);
  
  LOGI("=> starttime_int64_audio:     %" PRId64, starttime_int64_audio);
  
  LOGI("=> endtime_int64_audio:       %" PRId64, endtime_int64_audio);
  
  
  
  // loop input container and decode frames
  
  while(av_read_frame(input_container, &decoded_packet)>=0) {
  
      // video packets
  
      if (decoded_packet.stream_index == video_stream_index) {
  
          len = avcodec_decode_video2(input_stream->codec, decoded_frame, &got_frame, &decoded_packet);
  
          if(len < 0) {
  
              cleanup(decoded_packet, output_container, decoded_frame);
  
              avformat_close_input(&input_container);
  
              LOGE("=> No frames to decode");
  
              return -1;
  
          }
  
          // this is the trim range we're looking for
  
          if(got_frame && decoded_frame->pkt_pts >= starttime_int64 && decoded_frame->pkt_pts <= endtime_int64) {
  
                  av_init_packet(&encoded_packet);
  
                  encoded_packet.data =  NULL;
  
                  encoded_packet.size =  0;
  
  
  
                  ret = avcodec_encode_video2(output_stream->codec, &encoded_packet, decoded_frame, &got_frame);
  
                  if (ret < 0) {
  
                      cleanup(decoded_packet, output_container, decoded_frame);
  
                      avformat_close_input(&input_container);
  
                      LOGE("=> Error encoding frames");
  
                      return ret;
  
                  }
  
                  if(got_frame) {
  
                      if (output_stream->codec->coded_frame->key_frame) {
  
                          encoded_packet.flags |= AV_PKT_FLAG_KEY;
  
                      }
  
  
  
                      encoded_packet.stream_index = output_stream->index;
  
                      encoded_packet.pts = av_rescale_q(current_frame_num, output_stream->codec->time_base, output_stream->time_base);
  
                      encoded_packet.dts = av_rescale_q(current_frame_num, output_stream->codec->time_base, output_stream->time_base);
  
  
  
                      ret = av_interleaved_write_frame(output_container, &encoded_packet);
  
                      if (ret < 0) {
  
                          cleanup(decoded_packet, output_container, decoded_frame);
  
                          avformat_close_input(&input_container);
  
                          LOGE("=> Error encoding frames");
  
                          return ret;
  
                      }
  
                      else {
  
                          current_frame_num +=1;
  
                      }
  
                  }
  
              av_free_packet(&encoded_packet);
  
          }
  
      }
  
      // audio packets
  
      else if(decoded_packet.stream_index == audio_stream_index) {
  
          // this is the trim range we're looking for
  
          if(decoded_packet.pts >= starttime_int64_audio && decoded_packet.pts <= endtime_int64_audio) {
  
              av_init_packet(&encoded_packet);
  
  
  
              encoded_packet.data =  decoded_packet.data;
  
              encoded_packet.size =  decoded_packet.size;
  
              encoded_packet.stream_index = audio_stream_index;
  
              encoded_packet.pts = av_rescale_q(current_frame_num_audio, output_stream_audio->codec->time_base, output_stream_audio->time_base);
  
              encoded_packet.dts = av_rescale_q(current_frame_num_audio, output_stream_audio->codec->time_base, output_stream_audio->time_base);
  
  
  
              ret = av_interleaved_write_frame(output_container, &encoded_packet);
  
              if (ret < 0) {
  
                  cleanup(decoded_packet, output_container, decoded_frame);
  
                  avformat_close_input(&input_container);
  
                  LOGE("=> Error encoding frames");
  
                  return ret;
  
              }
  
              else {
  
                  current_frame_num_audio +=1;
  
              }
  
             av_free_packet(&encoded_packet);
  
          }
  
      }
  
  }

  Edit

  I have slight improvement on the initial code. The audio and video are still not perfectly synced, but the original problem of the audio playing first followed by the video is resolved.

  I’m now writing the decoded packet to the output container rather than re-encoding it.

  In the end though I have the same problem - the trimmed video’s audio and video streams are not perfectly synced.

  // audio packets
  
      else if(decoded_packet.stream_index == audio_stream_index) {
  
          // this is the trim range we're looking for
  
          if(decoded_packet.pts >= starttime_int64_audio && decoded_packet.pts <= endtime_int64_audio) {
  
              ret = av_interleaved_write_frame(output_container, &decoded_packet);
  
              if (ret < 0) {
  
                  cleanup(decoded_packet, output_container, decoded_frame);
  
                  avformat_close_input(&input_container);
  
                  LOGE("=> Error writing audio frame (%s)", av_err2str(ret));
  
                  return ret;
  
              }
  
              else {
  
                  current_frame_num_audio +=1;
  
              }
  
          }
  
          else if(decoded_frame->pkt_pts > endtime_int64_audio) {
  
              audio_copy_complete = true;
  
          }
  
      }

• how to convert a video to H264 video codec format & webm and ogv with watermark via ffmpeg

  5 mars 2015, par ali

  I used this code :

  function convert()
  
  {
  
    exec("$ffmpegPath  -i $inputPath -i $watermark -filter_complex overlay=10:10 $outPath");
  
  }
  
  
  
  convert('test.mp4');
  
  convert('test.webm');
  
  convert('test.ogv');

  This code works...but not in browser !
  I can run output videos via KMPLAYER in desktop (ovg only video - no audio)

  I want to run it on web via videojs.

  mp4 playback with videojs in browsers - audio plays but video is blank.

  I googled and I think I must convert it to h264 format...

  Can somebody change my orginal code to 3 valid codes for h264,webm and ogv ?