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• Demux H264 from (already recorded) raw RTSP stream on HDD

  4 janvier 2017, par Pascal Zurek

  I once had to record a video from an IP camera (Sony SRG 300SE) in a hurry. I did so by launching

  gst-launch-1.0 rtspsrc location="rtsp://192.168.0.100/video1" ! filesink location="recording.raw"

  which ran flawlessly.
  Now, I would like to demux the audio and video (aac and h264) from that file. I now know that it would have been much smarter to do that right when receiving the stream, but well, I didn’t.

  I already tried

  1. gst-launch-1.0 rtspsrc location="recording.raw" ! ..., which results in "No valid RTSP URL was provided" (GStreamer 1.10.2) ;
  2. ffmpeg -i recording.raw -c:v copy -c:a copy recording.mp4, which only leads to "Invalid data found when processing input" (ffmpeg 3.2.2)
  3. gst-launch-1.0 filesrc location="recording.raw" ! application/x-rtp ! rtph264depay ! h264parse ! matroskamux ! filesink location=out.mkv, leading to
     WARNING: from element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: Could not decode stream.
Additional debug info:
gstrtpbasedepayload.c(503): gst_rtp_base_depayload_handle_buffer (): /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:
Received invalid RTP payload, dropping

  Is there any way I get at least the video out of that stream ?

  Update :
  4. cat recording.raw | ffmpeg -i - -c:v copy -c:a copy recording.mp4 leads to pipe:: Invalid data found when processing input in ffmpeg 3.2.2, and to the following in an older build :

  [aac @ 0x2d4b3a0] Format aac detected only with low score of 1, misdetection possible!
  
  [aac @ 0x2d4c180] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.                                                                                                                                            
  
  [aac @ 0x2d4c180] Sample rate index in program config element does not match the sample rate index configured by the container.                                                         
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Prediction is not allowed in AAC-LC.
  
  [aac @ 0x2d4c180] channel element 3.10 is not allocated
  
  [aac @ 0x2d4c180] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 0x2d4c180] channel element 1.6 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.5 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.8 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.12 is not allocated
  
  [aac @ 0x2d4c180] channel element 3.2 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.10 is not allocated
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Prediction is not allowed in AAC-LC.
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] TNS filter order 15 is greater than maximum 12.
  
  [aac @ 0x2d4c180] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 0x2d4c180] Inconsistent channel configuration.
  
  [aac @ 0x2d4c180] get_buffer() failed
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Prediction is not allowed in AAC-LC.
  
  [aac @ 0x2d4c180] channel element 1.5 is not allocated
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Pulse tool not allowed in eight short sequence.
  
  [aac @ 0x2d4c180] Number of bands (30) exceeds limit (21).
  
  [aac @ 0x2d4c180] Number of bands (16) exceeds limit (14).
  
  [aac @ 0x2d4c180] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 0x2d4c180] Sample rate index in program config element does not match the sample rate index configured by the container.

  ...thus creating a AAC only file. As far as I can see, the AAC is silence (which could have be the case, though).

  Thanks in advance,
  Pascal

• ffmpeg - when scaling, how to keep shapes of people's heads

  19 janvier 2017, par Dave

  Ok, I’m quite familiar with FFMPEG utility in general, and have used it for
  years to cut short snippets from videos, etc. But it’s only in the last
  month that and I decided to learn to use it to transcode with video-filters
  etc. (Before that, I was using other tools such as ’Handbrake’ and ’FreeMake’
  and VLC, etc.)

  For my ffmpeg transcodes, my target output resolution will always be constant, from one transcode run to the next. But the resolution and display aspect-ratio of the input file, from one transcode run to the next, will vary...could be almost any values.
  The input files will never already have black-bars when displayed.

  So, the relevant portion [ i.e. the video-filter(s) part) of my cmd line ] presently is as follows :

  ffmpeg ... -vf "scale=720:406,setsar=1,pad=720:506:0:40:Black" ...

  Also note : I do NOT use the "-aspect" option in the cmd-line. (Maybe I’ll
  need to (???) to solve my issue, but I’m unsure about how that interacts
  with scaling.)

  ( EDIT : Oh, I happen to have chosen that resolution value of 720x406, for
  the image-area (i.e. inside the top/bottom black bars) because it
  has an aspect ratio of 16:9 (Of course, 16:9 ratio is common these days. )

  My cmd always executes cleanly and produces an output file (a WebM, tho I doubt
  that container types and/or vcodec choices matter at all to scaling algorithm issues).

  So the issue/problem that I’m trying to solve is how to prevent any stretching
  in either direction. In other words, a round soccer ball in the input file
  must yield a round ball in the output file ! (NOT oval-shaped in either axis).

  ( Edit #2 : Oh, I forgot to mention that I’m not have the same amount of stretching from one ffmpeg output file to the next. Sometimes there is
  no stretch in my output file, and with some other input file, the
  people are too tall in the output, and some other output file will have
  people are too wide. I’m assuming
  there is some single cmd that will always work for each randomly sized
  input file, WITHOUT having to resort to examining meta-data of each
  input and then having to adjust portions of the needed ffmpeg cmd.
  I assume this because I have used a tool called "FreeMake" that needs
  no such adjustment. When you do a ’scale’ with that program, it asks
  you to choose one of four adjustment-algorithms labeled "original"
  "stretched", "zoom..." and "auto". If I recall correctly, it was the
  "auto" choice that prevented any stretching.)

  The goal of that last filter (i.e. the "pad=720:506:0:40:Black" phrase) is to
  add a black bar of 40 pixels to the top and 60 pixels to the bottom.
  (That filter IS producing the black-bands, as desired. I mention it,
  because I’m unsure whether it could be having any effect on the altered
  shape of the ’round soccer ball’). If the "pad" filter IS part of the
  issue, then maybe I’ll need to make multiple ffmpeg cmds to achieve
  my overall goal (!?!?). [I’d LIKE to be able to do everything in just
  one ffmpeg cmd, as shown.]

  OK ?

  So are there any image-processing and ffmpeg gurus out there that
  know how to fix my problem ?

  TIA...

  Dave

• FFmpeg errors when adding a new song to queue in Discord bot

  5 novembre 2023, par Feras

  I am facing issues with my Discord bot, which is designed to play music from YouTube and Spotify. The bot uses pytube and youtube-dl to download the audio and then plays it using Discord's FFmpegPCMAudio. The bot is able to play a song correctly, but errors occur as soon as I try to add a new song to the queue. Here are some of the error messages I'm receiving :

  


  [aac @ 0000017d8fe84ac0] channel element 0.10 is not allocated
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0000017d8fe84ac0] channel element 3.12 is not allocated
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0000017d8fe84ac0] channel element 2.12 is not allocated
...
[aac @ 0000017d8fe84ac0] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0000017d8fe84ac0] Number of scalefactor bands in group (56) exceeds limit (47).
Error while decoding stream #0:0: Invalid data found when processing input


  


  Here's how I'm using FFmpeg in my code :

  


  ctx.voice_client.play(discord.FFmpegPCMAudio(executable="ffmpeg", source=new_song['file_path'], options="-err_detect ignore_err -strict -2 -loglevel warning"), after=lambda e: play_next(ctx))


  


  The first song plays without any issues, but as soon as I add a new song to the queue, these errors appear.

  


  How can I resolve these errors ?

  


  I have tried adjusting FFmpeg options and ensuring the downloaded files are intact, but the errors persist.