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• webM files shows green and purple effects on mobile

  11 octobre 2015, par Naveen Gamage

  I have converted several GIFs to webM files using ffmpeg on my Ubuntu 14.04 server.

  Heres the code I used for conversation.

  ffmpeg -i your_gif.gif -c:v libvpx -crf 12 -b:v 500K output.webm

  source https://gist.github.com/ndarville/10010916

  The problem is converted webM files shows perfectly fine on PCs but on my mobile it shows with green and purple shadows.

  PC

  

  Mobile

  

  I tried changing -crf and -b:v values to their max but nothing happens.

  webM file : http://d1pnsuxwa0it39.cloudfront.net/uploads/comments/webm/4673555.webm

  edit :

  also I can see webM files on some other sites fine. I think this has to do something with the way I convert files.

  edit :

  I have tried another code I found on stackoverflow but still the same.

  ffmpeg -f gif -i infile.gif outfile.mp4

  EDIT :

  If anyone think this has something to do with the way I installed FFMPEG, I followed the steps on FFMPEG official docs.

  https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu

  EDIT :

  Input file :

  http://d1pnsuxwa0it39.cloudfront.net/test/1.gif

  Output file :

  http://d1pnsuxwa0it39.cloudfront.net/test/output.webm

  FFMPEG CLI output

   /home/naveencg/bin/ffmpeg -i 1.gif -c:v libvpx -crf 12 -b:v 500K output.webm
  
  ffmpeg version 2.5.git Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Dec 31 2014 14:37:15 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --prefix=/home/naveencg/ffmpeg_build --extra-cflags=-I/home/naveencg/ffmpeg_build/include --extra-ldflags=-L/home/naveencg/ffmpeg_build/lib --bindir=/home/naveencg/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 19.100 / 56. 19.100
  
    libavformat    56. 16.102 / 56. 16.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  6.100 /  5.  6.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, gif, from '1.gif':
  
    Duration: N/A, bitrate: N/A
  
      Stream #0:0: Video: gif, bgra, 350x169, 25 fps, 25 tbr, 100 tbn, 100 tbc
  
  [libvpx @ 0x1e2bf60] v1.3.0
  
  Output #0, webm, to 'output.webm':
  
    Metadata:
  
      encoder         : Lavf56.16.102
  
      Stream #0:0: Video: vp8 (libvpx), yuva420p, 350x169, q=-1--1, 500 kb/s, 25 fps, 1k tbn, 25 tbc
  
      Metadata:
  
        encoder         : Lavc56.19.100 libvpx
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (gif (native) -> vp8 (libvpx))
  
  Press [q] to stop, [?] for help
  
  frame=   21 fps=0.0 q=0.0 size=      58kB time=00:00:00.84 bitrate= 569.7kbits/sframe=   44 fps= 41 q=0.0 size=     110kB time=00:00:01.76 bitrate= 512.4kbits/sframe=   62 fps= 39 q=0.0 size=     153kB time=00:00:02.48 bitrate= 505.9kbits/sframe=   84 fps= 40 q=0.0 size=     210kB time=00:00:03.36 bitrate= 510.8kbits/sframe=   88 fps= 41 q=0.0 Lsize=     218kB time=00:00:03.52 bitrate= 508.3kbits/s
  
  video:216kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.971527%

• Is FFmpegAudioDecoder supposed to reinitialize upon append of new init segment

  14 novembre 2023, par martin

  I am attempting to switch audio tracks but when switching the FFmpegAudioDecoder never reinitializes like it does with video tracks of differing resolutions. I am not certain if this is the intended behavior of FFmpegAudioDecoder and would love to learn more about the expected behavior.

  


  When switching audio tracks I end up calling the following operations :

  


  if sourceBuffer.getIsUpdate() {sourceBuffer.abort()}
sourceBuffer.remove(0-videoDuration)
initSegmentDataStream = fetch init segment of new audio representation
sourceBuffer.appendBuffer(initSegmentDataStream)


  


  These are the Media tab messages from initial video load

  


  ChunkDemuxer
Selected FFmpegAudioDecoder for audio decoding, config: codec: aac, profile: unknown, bytes_per_channel: 2, channel_layout: STEREO, channels: 2, samples_per_second: 48000, sample_format: Signed 16-bit, bytes_per_frame: 4, seek_preroll: 0us, codec_delay: 0, has extra data: false, encryption scheme: Unencrypted, discard decoder delay: false, target_output_channel_layout: STEREO, target_output_sample_format: Unknown sample format, has aac extra data: true
Cannot select DecryptingVideoDecoder for video decoding
Cannot select VDAVideoDecoder for video decoding
Cannot select VpxVideoDecoder for video decoding
Selected Dav1dVideoDecoder for video decoding, config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1280,720], visible rect: [0,0,1280,720], natural size: [1280,720], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}
Dropping audio frame (DTS 0us PTS -105375us,-62709us) that is outside append window [0us,9223372036854775807us).
Dropping audio frame (DTS 42666us PTS -62708us,-20042us) that is outside append window [0us,9223372036854775807us).
Truncating audio buffer which overlaps append window start. PTS -20041us frame_end_timestamp 22625us append_window_start 0us
Effective playback rate changed from 0 to 1


  


  For comparison this is what I get when appending the init segment of a different video resolution / track

  


  video decoder config changed midstream, new config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1920,1080], visible rect: [0,0,1920,1080], natural size: [1920,1080], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}



  


  Chrome version : Version 119.0.6045.123 (Official Build)

  


  When appending the new init segment of an audio track I was expecting the FFmpegAudioDecoder to be reinitialized like the Dav1dVideoDecoder does for video tracks

  


• ffmpeg : Overlay audios with different lengths using amix and apad

  23 novembre 2017, par Valdir

  I’m trying to overlay two audio files with the following command :

  ffmpeg -y -i mp3/test1.mp3 -i mp3/test2.mp3 -filter_complex "[0:0][1:0] amix=inputs=2:duration=longest" -c:a libmp3lame merged.mp3

  But it fails if the length of the audios are different :

  ffmpeg -y -i live.mp3 -i mp3/test1.mp3 -filter_complex "[0:0][1:0] amix=inputs=2:duration=longest" -c:a libmp3lame merged2.mp3
  
  ffmpeg version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --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-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    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 @ 0x22cf760] Format mp3 detected only with low score of 1, misdetection possible!
  
  [mp3 @ 0x22cf760] Could not find codec parameters for stream 0 (Audio: mp3, 0 channels, s16p): unspecified frame size
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  live.mp3: could not find codec parameters
  
  Input #0, mp3, from 'live.mp3':
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Audio: mp3, 0 channels, s16p
  
  [mp3 @ 0x22d10a0] Skipping 0 bytes of junk at 18034.
  
  Input #1, mp3, from 'mp3/test1.mp3':
  
    Metadata:
  
      title           : Nothing Else Matters [Official Music Video]
  
      artist          : Metallica
  
    Duration: 00:06:25.80, start: 0.025057, bitrate: 192 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc56.14
  
  [abuffer @ 0x22fa2c0] Value inf for parameter 'time_base' out of range [0 - 2.14748e+09]
  
      Last message repeated 3 times
  
  [abuffer @ 0x22fa2c0] Error setting option time_base to value 1/0.
  
  [graph 0 input from stream 0:0 @ 0x22fa3e0] Error applying options to the filter.
  
  Error configuring complex filters.
  
  Numerical result out of range

  How do I add the apad filter so that silence is added to the shortest audio ?