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• 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 frame number or timecode overlay error ; need to reset frame count

  14 juillet 2015, par Samuel Mehr

  I’ve got timecode or frame number overlays as per other posts here, but I’m converting videos from AVCHD 59.94fps to H.264 30fps, and the timecode/frameN overlay is keeping the original values. So as I play the new video, the frames are labeled 1,3,5,7, ... but the video plays at normal speed. Is there an option to reset the frame count to the output video’s framerate ?

  thanks !

  ffmpeg -i in.mov -c:v libx264 -r 30 -s 1280x720 -b:v 5400 -vf "drawtext=fontsize=24:fontfile=FONTHERE.otf: timecode='00\:00\:00\:00': r=30: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000099" testtest.mp4

• Rotate video adding black bars with ffmpeg

  13 février 2016, par MrMoog

  I have a lot of mp4 full hd videos made with a smartphone that looks rotated 90°.

  These files have proper rotation metadata and they plays correctly on VLC or other desktop mediaplayers, but I want to display them on my dvd player that does not handles mp4 metadata.

  So I tried to transpose with ffmpeg but I obtain 1080x1920 files, resolution obviously not recognized by tv/dvd-player.

  How can I transpose, resize and add black side (left/right) bars of the correct dimensions to get "correct" 1920x1080 videos ?