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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 encoding live audio to aac issue

  12 juillet 2015, par Ruurd Adema

  I’m trying to encode live raw audio coming from a Blackmagic Decklink input card to a mov file with AAC encoding.

  The issue is that the audio sounds distorted and plays to fast.

  I created the software based on a couple of examples/tutorials including the Dranger tutorial and examples on Github (and of course the examples in the FFmpeg codebase).

  Honestly, at this moment I don’t exactly know what the cause of the problem is. I’m thinking about PTS/DTS values or a timebase mismatch (because of the too fast playout), I tried a lot of things, including working with an av_audio_fifo.

  • When outputting to the mov file with the AV_CODEC_ID_PCM_S16LE codec, everything works well
  • When outputting to the mov file with the AV_CODEC_ID_AAC codec, the problems occur
  • When writing RAW audio VLC media info shows :
    Type : Audio, Codec : PCM S16 LE (sowt), Language : English, Channels : Stereo, Sample rate : 48000 Hz, Bits per sample.
  • When writing with AAC codec VLC media info shows :
    Type : Audio, Codec : MPEG AAC Audio (mp4a), Language : English, Channels : Stereo, Sample rate : 48000 Hz.

  Any idea(s) of what’s causing the problems ?

  Code

  // Create output context
  
  
  
  output_filename = "/root/movies/encoder_debug.mov";
  
  output_format_name = "mov";
  
  if (avformat_alloc_output_context2(&output_fmt_ctx, NULL, output_format_name, output_filename) < 0)
  
  {
  
      printf("[ERROR] Unable to allocate output format context for output: %s\n", output_filename);
  
  } 
  
  
  
  // Create audio output stream
  
  
  
  static AVStream *encoder_add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
  
  {
  
      AVCodecContext  *c;
  
      AVCodec         *codec;
  
      AVStream        *st;
  
  
  
      st = avformat_new_stream(oc, NULL);
  
      if (!st) 
  
      {
  
          printf("[ERROR] Could not allocate new audio stream!\n");
  
          exit(-1);
  
      }
  
  
  
      c                 = st->codec;
  
      c->codec_id       = codec_id;
  
      c->codec_type     = AVMEDIA_TYPE_AUDIO;
  
      c->sample_fmt     = AV_SAMPLE_FMT_S16;
  
      c->sample_rate    = decklink_config()->audio_samplerate;
  
      c->channels       = decklink_config()->audio_channel_count;
  
      c->channel_layout = av_get_default_channel_layout(decklink_config()->audio_channel_count);
  
      c->time_base.den  = decklink_config()->audio_samplerate;
  
      c->time_base.num  = 1;
  
  
  
      if (codec_id == AV_CODEC_ID_AAC)
  
      {
  
          c->bit_rate       = 96000;  
  
          //c->profile = FF_PROFILE_AAC_MAIN; //FIXME Generates error: "Unable to set the AOT 1: Invalid config"
  
          // Allow the use of the experimental AAC encoder
  
          c->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; 
  
      }
  
  
  
      // Some formats want stream headers to be seperate (global?)
  
      if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
      {
  
          c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
      }
  
  
  
      codec = avcodec_find_encoder(c->codec_id);
  
      if (!codec)
  
      {
  
          printf("[ERROR] Audio codec not found\n");
  
          exit(-1);
  
      }
  
  
  
      if (avcodec_open2(c, codec, NULL) < 0)
  
      {
  
          printf("[ERROR] Could not open audio codec\n");
  
          exit(-1);
  
      }
  
  
  
      return st;
  
  }
  
  
  
  // En then, at every incoming frame this function gets called:
  
  
  
  void encoder_handle_incoming_frame(IDeckLinkVideoInputFrame *videoframe, IDeckLinkAudioInputPacket *audiopacket)
  
  {
  
      void *pixels = NULL;
  
      int   pitch = 0;
  
      int got_packet = 0;
  
  
  
      void *audiopacket_data          = NULL;
  
      long  audiopacket_sample_count  = 0;
  
      long  audiopacket_size          = 0;
  
      long  audiopacket_channel_count = 2;
  
  
  
      if (audiopacket)
  
      {  
  
          AVPacket pkt = {0,0,0,0,0,0,0,0,0,0,0,0,0,0};
  
          AVFrame *frame;
  
          BMDTimeValue audio_pts;
  
          int requested_size;
  
          static int last_pts1, last_pts2 = 0;
  
  
  
          audiopacket_sample_count  = audiopacket->GetSampleFrameCount();
  
          audiopacket_channel_count = decklink_config()->audio_channel_count;
  
          audiopacket_size          = audiopacket_sample_count * (decklink_config()->audio_sampletype/8) * audiopacket_channel_count;
  
  
  
          audiopacket->GetBytes(&audiopacket_data);
  
  
  
          av_init_packet(&pkt);    
  
  
  
          printf("\n=== Audiopacket: %d ===\n", audio_stream->codec->frame_number);
  
  
  
          if (AUDIO_TYPE == AV_CODEC_ID_PCM_S16LE)
  
          {
  
              audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
  
  
  
              pkt.pts          = audio_pts;
  
              pkt.dts          = pkt.pts; 
  
              pkt.flags       |= AV_PKT_FLAG_KEY;                 // TODO: Make sure if this still applies
  
              pkt.stream_index = audio_stream->index;
  
              pkt.data         = (uint8_t *)audiopacket_data;
  
              pkt.size         = audiopacket_size;
  
  
  
              printf("[PACKET] size:              %d\n", pkt.size);
  
              printf("[PACKET] pts:               %li\n", pkt.pts);
  
              printf("[PACKET] pts delta:         %li\n", pkt.pts - last_pts2);
  
              printf("[PACKET] duration:          %d\n", pkt.duration);
  
              last_pts2 = pkt.pts;
  
  
  
              av_interleaved_write_frame(output_fmt_ctx, &pkt);
  
          } 
  
          else if (AUDIO_TYPE == AV_CODEC_ID_AAC)
  
          {
  
              frame = av_frame_alloc();
  
              frame->format = audio_stream->codec->sample_fmt;
  
              frame->channel_layout = audio_stream->codec->channel_layout;
  
              frame->sample_rate = audio_stream->codec->sample_rate;
  
              frame->nb_samples = audiopacket_sample_count;
  
  
  
              requested_size = av_samples_get_buffer_size(NULL, audio_stream->codec->channels, audio_stream->codec->frame_size, audio_stream->codec->sample_fmt, 1);
  
  
  
              audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
  
  
  
              printf("[DEBUG] Sample format:      %d\n", frame->format);
  
              printf("[DEBUG] Channel layout:     %li\n", frame->channel_layout);
  
              printf("[DEBUG] Sample rate:        %d\n", frame->sample_rate);
  
              printf("[DEBUG] NB Samples:         %d\n", frame->nb_samples);
  
              printf("[DEBUG] Datasize:           %li\n", audiopacket_size);
  
              printf("[DEBUG] Requested datasize: %d\n", requested_size);
  
              printf("[DEBUG] Too less/much:      %li\n", audiopacket_size - requested_size);
  
              printf("[DEBUG] Framesize:          %d\n", audio_stream->codec->frame_size);
  
              printf("[DEBUG] Audio pts:          %li\n", audio_pts);
  
              printf("[DEBUG] Audio pts delta:    %li\n", audio_pts - last_pts1);
  
              last_pts1 = audio_pts;
  
  
  
              frame->pts = audio_pts;
  
  
  
              if (avcodec_fill_audio_frame(frame, audiopacket_channel_count, audio_stream->codec->sample_fmt, (const uint8_t *)audiopacket_data, audiopacket_size, 0) < 0)
  
              {
  
                  printf("[ERROR] Filling audioframe failed!\n");
  
                  exit(-1);
  
              }
  
  
  
              got_packet = 0;
  
              if (avcodec_encode_audio2(audio_stream->codec, &pkt, frame, &got_packet) != 0)
  
              {
  
                  printf("[ERROR] Encoding audio failed\n");
  
              }
  
  
  
              if (got_packet) 
  
              {
  
                  pkt.stream_index = audio_stream->index;
  
                  pkt.flags       |= AV_PKT_FLAG_KEY; 
  
  
  
                  //printf("[PACKET] size:              %d\n", pkt.size);
  
                  //printf("[PACKET] pts:               %li\n", pkt.pts);
  
                  //printf("[PACKET] pts delta:         %li\n", pkt.pts - last_pts2);
  
                  //printf("[PACKET] duration:          %d\n", pkt.duration);
  
                  //printf("[PACKET] timebase codec:    %d/%d\n", audio_stream->codec->time_base.num, audio_stream->codec->time_base.den);
  
                  //printf("[PACKET] timebase stream:   %d/%d\n", audio_stream->time_base.num, audio_stream->time_base.den);
  
                  last_pts2 = pkt.pts;
  
  
  
                  av_interleaved_write_frame(output_fmt_ctx, &pkt);
  
              }
  
              av_frame_free(&frame); 
  
          }
  
  
  
          av_free_packet(&pkt);
  
      }
  
      else
  
      {
  
          printf("[WARNING] No audiopacket received!\n");
  
      }
  
  
  
      static int count = 0;
  
      count++;
  
  }

• ffmpeg dshow command is listing device name different from actual device name

  20 octobre 2015, par Somanshu

  i have used the command:ffmpeg -f dshow -list_devices true -i dummy
  to list the devices.
  But if device name is in language other than english then device name shown by ffmpeg and actual differ.

  How can i solve this mismatch problem.

  This image shows the device name in spanish

  This image shows the device name as read by ffmpeg