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• Video files recorded in Google Chrome have stuttering audio

  4 juin 2018, par maxpaj

  Background

  I’m developing a platform where users can record videos of themselves or their screen and send them as video messages to customers / clients.

  I have limited users to only using my application with Google Chrome and I’m using the MediaRecorder API to record the video data from the users screen or webcamera. The codecs that are used for recording are VP8/OPUS (WEBM container).

  I need the videos to run in as many browsers as possible, so I’m using a 3rd party service to transcode videos from whatever format I’m getting from the users to a H.265/AAC MP4 container (caniuse MPEG-4/H.264).

  Issue

  Lately I’ve seen that some videos recorded on Mac OSX machines have the video and audio out of sync or that the video and audio stutters, depending on which player I’m using. I call these video files corrupt, for lack of a better word. Playing a corrupt file in Google Chrome renders smooth playing audio. Playing the video in VLC on my Windows machine renders stuttering audio.

  When I run the corrupt video files through the transcoding service I get video files with stuttering audio, no matter which player I’m using.

  This is an unwanted result and pretty much unacceptable since the audio needs to be smooth in order for the recipient of a video to not be bothered with the quality.

  Debugging

  According to the transcoding service support, this happens because of their mechanisms that try to sync up the audio and video from the corrupt file :

  Inspecting our encoding logs, I’ve noticed the following kind of
  warnings :

  [2018-05-16 14:08:38.009] [pcm_s16le @ 0x1d608c0] pcm_encode_frame :
  filling in for 5856 missing samples (122 ms) before pts 40800 to
  correct sync ! [2018-05-16 14:08:38.009] [pcm_s16le @ 0x1d608c0]
  pcm_encode_frame : dropping 2880 samples (60 ms) at pts 43392 to help
  correct sync to -3168 samples (-66 ms) !

  The problem here comes from the way that the audio in the original
  source file is encoded.

  -

  you should ensure that the audio is not out of sync (audio timestamps
  are correct) in your source file before submitting the job

  Running a corrupt file through ffmpeg on my own machine, re-encoding with the same codecs, produces the same kind of stuttering video. The logs produce an alarming amount of errors. Here is a sample of the log output :

  [libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 194.8kbits/s dup=0 drop=5 speed=0.31x
  
  [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15434, current: 15394; changing to 15434. This may result in incorrect timestamps in the output file.
  
  [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15434, current: 15414; changing to 15434. This may result in incorrect timestamps in the output file.
  
  [libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 193.3kbits/s dup=0 drop=5 speed=0.309x
  
  [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15539, current: 15499; changing to 15539. This may result in incorrect timestamps in the output file.
  
  [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15539, current: 15519; changing to 15539. This may result in incorrect timestamps in the output file.
  
  [libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 192.0kbits/s dup=0 drop=5 speed=0.308x
  
  [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15667, current: 15627; changing to 15667. This may result in incorrect timestamps in the output file.
  
  [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15667, current: 15647; changing to 15667. This may result in incorrect timestamps in the output file.
  
  [libopus @ 0000029938e24d80] Queue input is backward in time

  I tried running the same inputs through another transcoding service and those outputs worked a lot better - video was still stuttering but the audio played smoothly, which is more important to the use case of my application.

  To my knowledge, this have so far only occurred for users on Mac OSX machines.

  Questions

  1. Is there anything I can do to have the files work better ? Or is this entirely a consequence of how encoding of video and audio in Google Chrome works ?

  2. One step in the right direction would be to just be able to detect when the video is corrupt. How can I do that ?

• ffmpeg rtsp stream to YouTube livestream not doing anything

  29 mai 2018, par felixosth

  I’m using C# to initiate ffmpeg. I’ve a onvif bridge server for a CCTV VMS and I’m building a application to enable the user to livestream any CCTV camera to YouTube.

  The rtsp stream to the camera looks like this :

  rtsp ://onvif:bridge@localhost:554/live/xxxxx-xxxguidtocameraxxx-xxxxx

  I’m new to ffmpeg and I’m using this arg :

  -f lavfi -i anullsrc -rtsp_transport udp -i camerastreamurl -tune zerolatency -vcodec libx264 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp ://x.rtmp.youtube.com/live2/streamkey -loglevel debug

  It seems like ffmpeg isn’t finding the camera stream, only the fake audio one. It just freezes.

  This is the result of the debug log :

  ffmpeg version N-91172-gebf85d3190 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 18.102 / 56. 18.102
  
    libavcodec     58. 19.104 / 58. 19.104
  
    libavformat    58. 17.100 / 58. 17.100
  
    libavdevice    58.  4.100 / 58.  4.100
  
    libavfilter     7. 24.100 /  7. 24.100
  
    libswscale      5.  2.100 /  5.  2.100
  
    libswresample   3.  2.100 /  3.  2.100
  
    libpostproc    55.  2.100 / 55.  2.100
  
  Splitting the commandline.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'lavfi'.
  
  Reading option '-i' ... matched as input url with argument 'anullsrc'.
  
  Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'udp'.
  
  Reading option '-i' ... matched as input url with argument 'rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b'.
  
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  
  Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
  
  Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument '+'.
  
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
  
  Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
  
  Reading option '-strict' ...Routing option strict to both codec and muxer layer
  
   matched as AVOption 'strict' with argument 'experimental'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  
  Reading option 'rtmp://x.rtmp.youtube.com/live2/xxxxxxxx' ... matched as output url.
  
  Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
  
  Finished splitting the commandline.
  
  Parsing a group of options: global .
  
  Applying option loglevel (set logging level) with argument debug.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input url anullsrc.
  
  Applying option f (force format) with argument lavfi.
  
  Successfully parsed a group of options.
  
  Opening an input file: anullsrc.
  
  detected 8 logical cores
  
  [AVFilterGraph @ 0000027a34bad7c0] query_formats: 2 queried, 3 merged, 0 already done, 0 delayed
  
  [Parsed_anullsrc_0 @ 0000027a34badb80] sample_rate:44100 channel_layout:'stereo' nb_samples:1024
  
  [lavfi @ 0000027a34babc80] All info found
  
  Input #0, lavfi, from 'anullsrc':
  
    Duration: N/A, start: 0.000000, bitrate: 705 kb/s
  
      Stream #0:0, 1, 1/44100: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: input url rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
  
  Successfully parsed a group of options.
  
  Opening an input file: rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
  
  [tcp @ 0000027a34bb5980] No default whitelist set

  Edit :
  I’m getting this result with minimal args :
  

• ffmpeg - Writing thumbnail to Azure Blob Storage

  20 juin 2018, par Heinrich

  Is is possible to write directly to Azure using ffmpeg ? I am using ffmpeg to generate thumbnails for videos and I need to save them to an Azure Blob storage using a pre generated SAS url, is this possible ?
  When I run ffmpeg it seems to run happily, without any trace of errors as far as I can tell, but the file is never written\create in the storage. I know the target url works as I can upload a file to exact same url using postman.

  Below is the ffmpeg command that I am trying to get to work :

  .\ffmpeg.exe -i {srcurl} -loglevel 48 -frames:v 1 -y -method PUT -chunked_post 0 -headers "x-ms-blob-type: BlockBlob" -f apng {targeturl}

  I took the urls out as they are to long but the are both in the format as below :

  https://{host}.blob.core.windows.net/{pathtofile}/{filename}?sp=r&st=2018-06-03T23:07:25Z&se=2018-06-04T07:07:25Z&spr=https&sv=2017-11-09&sig={signature}

  Now I know that it works to some degree because if I write the file locally everything is happy. ffmpeg is able to get the video from the source url, read the file and create a thumbnail locally.

  The thing that confuses me more however is that if I specify a rubbish url path, one where I know it shouldn’t have permission to write to, it still does not complain at all.

  It is able to resolve the host name because changing the host to some non-existent one it will cause the "Failed to resolve hostname" error.