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• Metadata in mp3 not working when piping from ffmpeg with album art

  10 février 2019, par Cromon

  In my program I am piping a webm from a stream to ffmpeg and then pipe the output to a http request. Part of the process is adding metadata for the mp3. This has so far worked great. However after adding an image as album art it has started to act unexpected.

  First this is the command line I am using inside the program :

      val parameters = listOf("ffmpeg",
  
              "-i", "-",
  
              "-i", albumImage.absolutePath,
  
              "-map", "0",
  
              "-map", "1",
  
              "-c:v", "copy",
  
              "-f", "mp3",
  
              "-id3v2_version", "4",
  
              "-metadata", "title=${info.title}",
  
              "-metadata", "album=YouTube",
  
              "-metadata", "artist=${info.author}",
  
              "-metadata:s:v", "title=Album Cover",
  
              "-metadata:s:v", "comment=Cover (front)",
  
              "-"
  
      )

  It creates a valid mp3 file and I can find both the metadata and the image in the mp3 file, however when playing it none of them are displayed in VLC or anywhere else. To test various configurations I have converted it to the command line.

  In a first try I have saved the video and the image and stopped using pipes altogether, which results in this :

  ffmpeg -i video.webm -i image.jpeg -map 0 -map 1 -c:v copy -f mp3 -id3v2_version 4 -metadata title="Tiësto & KSHMR feat. Vassy - Secrets (Official Music Video)" -metadata album="YouTube" -metadata artist="Spinnin' Records" -metadata:s:v title="Album Cover" -metadata:s:v comment="Cover (front)" output3.mp3

  In this case all metadata including the album art is displayed in VLC.

  I then recreated the same thing as in my program, piping both video input and audio output, looking like this :

  ffmpeg -i - -i image.jpeg -map 0 -map 1 -c:v copy -f mp3 -id3v2_version 4 -metadata title="Tiësto & KSHMR feat. Vassy - Secrets (Official Music Video)" -metadata album="YouTube" -metadata artist="Spinnin' Records" -metadata:s:v title="Album Cover" -metadata:s:v comment="Cover (front)" - < video.webm > output3.mp3

  This file is the same as my programs output. Neither title nor album nor album image are displayed (however it can play the file)

  To test a few more options I have hardcoded the output file but pipe the input file like this :

  ffmpeg -i - -i image.jpeg -map 0 -map 1 -c:v copy -f mp3 -id3v2_version 4 -metadata title="Tiësto & KSHMR feat. Vassy - Secrets (Official Music Video)" -metadata album="YouTube" -metadata artist="Spinnin’ Records" -metadata:s:v title="Album Cover" -metadata:s:v comment="Cover (front)" output3.mp3 < video.webm

  Now the metadata is working again. When hardcoding the input video and piping the output, its again gone.

  So to sum up : When piping the output of ffmpeg the metadata in the file is not properly working. Interestingly the stderr output of ffmpeg looks quite similar

  Hardcoded output3.mp3 :

  ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
  
  Input #0, matroska,webm, from 'pipe:':
  
    Metadata:
  
      encoder         : google/video-file
  
    Duration: 00:03:39.58, start: -0.007000, bitrate: N/A
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
  
  Input #1, image2, from 'image.jpeg':
  
    Duration: 00:00:00.04, start: 0.000000, bitrate: 1466 kb/s
  
      Stream #1:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 320x180, 25 tbr, 25 tbn, 25 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (opus (native) -> mp3 (libmp3lame))
  
    Stream #1:0 -> #0:1 (copy)
  
  Output #0, mp3, to 'output3.mp3':
  
    Metadata:
  
      TPE1            : Spinnin' Records
  
      TIT2            : Tiësto & KSHMR feat. Vassy - Secrets (Official Music Video)
  
      TALB            : YouTube
  
      TSSE            : Lavf57.83.100
  
      Stream #0:0(eng): Audio: mp3 (libmp3lame), 48000 Hz, stereo, fltp (default)
  
      Metadata:
  
        encoder         : Lavc57.107.100 libmp3lame
  
      Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 320x180, q=2-31, 25 tbr, 25 tbn, 25 tbc
  
      Metadata:
  
        title           : Album Cover
  
        comment         : Cover (front)

  With pipe output :

  ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
  
  Input #0, matroska,webm, from 'pipe:':
  
    Metadata:
  
      encoder         : google/video-file
  
    Duration: 00:03:39.58, start: -0.007000, bitrate: N/A
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
  
  Input #1, image2, from 'image.jpeg':
  
    Duration: 00:00:00.04, start: 0.000000, bitrate: 1466 kb/s
  
      Stream #1:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 320x180, 25 tbr, 25 tbn, 25 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (opus (native) -> mp3 (libmp3lame))
  
    Stream #1:0 -> #0:1 (copy)
  
  Output #0, mp3, to 'pipe:':
  
    Metadata:
  
      TPE1            : Spinnin' Records
  
      TIT2            : Tiësto & KSHMR feat. Vassy - Secrets (Official Music Video)
  
      TALB            : YouTube
  
      TSSE            : Lavf57.83.100
  
      Stream #0:0(eng): Audio: mp3 (libmp3lame), 48000 Hz, stereo, fltp (default)
  
      Metadata:
  
        encoder         : Lavc57.107.100 libmp3lame
  
      Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 320x180, q=2-31, 25 tbr, 25 tbn, 25 tbc
  
      Metadata:
  
        title           : Album Cover
  
        comment         : Cover (front)

• MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?

  23 janvier 2019, par AR5

  I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
  The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)

  Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.

  I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?

  I have already tried

  I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.

  I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.

  I also checked if both files were being served as 206 Partial Content and they both are indeed.

  I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv

  I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.

  I am really stuck on this issue, any help will be really appreciated.

  ---

  Edit

  I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :

  avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"

  Here is the command and log output from new server :

  ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
  
  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
  
  Input #0, mp3, from '/test/Track 01.mp3':
  
    Metadata:
  
      album           : Future Hndrxx Presents: The WIZRD
  
      artist          : Future
  
      genre           : Hip-Hop
  
      title           : Never Stop
  
      track           : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      date            : 2019
  
      encoder         : Lavf56.40.101
  
    Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60
  
      Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
  
  Please consider specifying a lower framerate, a different muxer or -vsync 2
  
  Output #0, mp3, to '/test/Track 01 (converted).mp3':
  
    Metadata:
  
      TALB            : Future Hndrxx Presents: The WIZRD
  
      TPE1            : Future
  
      TCON            : Hip-Hop
  
      TIT2            : Never Stop
  
      TRCK            : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      TDRC            : 2019
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc56.60.100 png
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (png (native) -> png (native))
  
    Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
  
  frame=    1 fps=0.1 q=-0.0 Lsize=    4788kB time=00:04:51.39 bitrate= 134.6kbits/s
  
  video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%

  Samples of MP3 files

  I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing

• Does stream seek order matter for ffmpeg av_seek_frame() ?

  2 janvier 2019, par necrosato

  I am attempting to seek both audio and video streams for an mp4 using the ffmpeg av_seek_frame method.

  I have encountered an issue when seeking that I have remedied by changing my seek order, but would like to make sure my fix is actually a fix and not some coincidental hack that works.

  I am attempting to seek both the audio and video stream to the first packet. For video, the first packet has a pts of 0. For audio, the first packet has a pts of -1024. The video stream has an index of 0 and the audio stream has an index of 1. This has all been verified using ffprobe on the media file to view the packets and streams.

  The following code does not work, it seeks both the audio and video streams to packets with pts of 0 :

  for (int i = format_context->nb_streams - 1; i >= 0; --i) {
  
      AVStream* stream = format_context->streams[i];
  
      av_seek_frame(format_context, i, stream->first_dts, flags);
  
  }

  But this properly seeks the video stream to pts 0 and audio stream to pts -1024 :

  for (int i = 0; i < format_context->nb_streams; ++i) {
  
      AVStream* stream = format_context->streams[i];
  
      av_seek_frame(format_context, i, stream->first_dts, flags);
  
  }

  Note that in the first example, audio is seeked before video, and in the second example video is seeked before audio.

  Does the order of the av_seek_frame calls actually matter, or is there a bug somewhere else in my code that this just so happens to cover up ?