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• Can ffmpeg copy metadata/ID3 from FLAC to MP3 ?

  9 avril 2019, par krypterro

  I have a bit of Python code that loops through audio files, finds .FLAC files, and then uses the Python subcommand to run ffmpeg. It works. The audio is fine, but even though I see the metadata in the shell, it doesn’t transfer the data to the ID3 tags in the MP3, and I am using the example found in the previous post here. Here’s the command :

  cmd = 'ffmpeg -y -i "' + src + '" -codec:a libmp3lame -q:a 0 -map_metadata 0 -id3v2_version 3 -write_id3v1 1 "' + dst + '"'

  Which works out to :

  ffmpeg -y -i "source.flac" -codec:a libmp3lame -q:a 0 -map_metadata 0 -id3v2_version 3 -write_id3v1 1 "destination.mp3"

  And here is the log dump :

  /usr/local/bin/python3.7 /home/krypterro/PycharmProjects/mediaman/RipFLAC.py
  
  1 Music Files Found
  
  2019-04-09 14:32:47.758 | INFO     | __main__:main:31 - Start of program
  
  2019-04-09 14:32:48.110 | DEBUG    | __main__:ripmp3:206 - Running Command: ffmpeg -y -i "/home/krypterro/audio/music_in/Visions/01-grimes-laughing_and_not_being_normal.flac" -acodec libmp3lame -ab 192000 "/home/krypterro/audio/music_out/Visions/01-grimes-laughing_and_not_being_normal.mp3"
  
  ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    WARNING: library configuration mismatch
  
    avcodec     configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared --enable-version3 --disable-doc --disable-programs --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, flac, from '/home/krypterro/audio/music_in/Visions/01-grimes-laughing_and_not_being_normal.flac':
  
    Metadata:
  
      ARTIST          : Grimes
  
      TITLE           : Laughing And Not Being Normal
  
      ALBUM           : Art Angels
  
      DATE            : 2015
  
      track           : 1
  
      GENRE           : Electronic
  
      disc            : 1
  
      TOTALDISCS      : 1
  
      TOTALTRACKS     : 15
  
      LANGUAGE        : English
  
      RIP DATE        : 2015-12-12
  
      RETAIL DATE     : 2015-00-00
  
      MEDIA           : CD
  
      ENCODER         : FLAC 1.2.1 -8 -V
  
      RIPPING TOOL    : EAC 1.0 Beta 3
  
      RELEASE TYPE    : Retail
  
      ORGANIZATION    : 4AD
  
      CATALOG         : CAD3535CD
  
    Duration: 00:01:47.51, start: 0.000000, bitrate: 743 kb/s
  
      Stream #0:0: Audio: flac, 44100 Hz, stereo, s16
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (flac (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  Output #0, mp3, to '/home/krypterro/audio/music_out/Visions/01-grimes-laughing_and_not_being_normal.mp3':
  
    Metadata:
  
      TPE1            : Grimes
  
      TIT2            : Laughing And Not Being Normal
  
      TALB            : Art Angels
  
      TDRC            : 2015
  
      TRCK            : 1
  
      TCON            : Electronic
  
      TPOS            : 1
  
      TOTALDISCS      : 1
  
      TOTALTRACKS     : 15
  
      TLAN            : English
  
      RIP DATE        : 2015-12-12
  
      RETAIL DATE     : 2015-00-00
  
      MEDIA           : CD
  
      CATALOG         : CAD3535CD
  
      RIPPING TOOL    : EAC 1.0 Beta 3
  
      RELEASE TYPE    : Retail
  
      ORGANIZATION    : 4AD
  
      TSSE            : Lavf57.83.100
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc57.107.100 libmp3lame
  
  size=    2522kB time=00:01:47.52 bitrate= 192.1kbits/s speed=  41x    
  
  video:0kB audio:2521kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.042501%
  
  2019-04-09 14:32:50.811 | INFO     | __main__:main:73 - End of program
  
  
  
  Process finished with exit code 0

• 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++;
  
  }

• lavu : Adding ARIB STD-B67 (hybrid log-gamma) enum value and transfer function.

  21 avril 2016, par Neil Birkbeck

  lavu : Adding ARIB STD-B67 (hybrid log-gamma) enum value and transfer function.
  Adding hybrid log-gamma (https://en.wikipedia.org/wiki/Hybrid_Log-Gamma)
  
  based on the standardization in ARIB STD-B67 :
  
  http://www.arib.or.jp/english/html/overview/doc/2-STD-B67v1_0.pdf
  The choice of enum value of 18 is consistent with HEVC :
  
  http://phenix.it-sudparis.eu/jct/doc_end_user/current_document.php?id=10481
  And also with latest proposal for color format in mkv :
  
  https://mailarchive.ietf.org/arch/search/?email_list=cellar&gbt=1&q=Colour+Format+proposal
  The implementation assumes a nominal input range of [0, 1], which is
  
  consistent with HEVC.
  Signed-off-by : Neil Birkbeck <neil.birkbeck@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavutil/color_utils.c
  • [DH] libavutil/pixdesc.c
  • [DH] libavutil/pixfmt.h
  • [DH] tests/ref/fate/color_utils