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• h264 : skip chroma edges at the picture boundary while deblocking 4:4:4

  20 janvier 2014, par Janne Grunau

  h264 : skip chroma edges at the picture boundary while deblocking 4:4:4
  This handles macroblock edges for the chroma components in the same way
  
  as for the luma compoment for 4:4:4 streams. The Spec explicitly states
  
  that the deblocking filter is not applied to edges at the boundary of
  
  the picture.
  Signed-off-by : Janne Grunau <janne-libav@jannau.net>
  

  • [DBH] libavcodec/h264.c

• avcodec/vp8 : Fix wrong vpx_rac_is_end() check

  16 novembre 2022, par Hirokazu Honda

  avcodec/vp8 : Fix wrong vpx_rac_is_end() check
  The check of vpx_rac_is_end check(s) are added originally from
  
  1afd246960202917e244c844c534e9c1e3c323f5. It causes a regression
  
  of some vp8 stream. b6b9ac5698c8f911841b469af77199153278c55c fixes
  
  the regression by a sort of band-aid way. This fixes the wrongness
  
  of the original commit. vpx_rac_is_end() should be called against
  
  the bool decoder for the vp8 headr context, not one for each
  
  coefficient. Reference is vp8_dixie_tokens_process_row() in token.c
  
  in spec 20.16.
  Fixes : Ticket 8069
  
  Fixes : regression of 1afd246960202917e244c844c534e9c1e3c323f5.
  
  Fixes : b6b9ac5698c8f911841b469af77199153278c55c
  Co-authored-by : Ronald S. Bultje <rsbultje@gmail.com>
  
  Signed-off-by : Hirokazu Honda <hiroh@chromium.org>
  
  Signed-off-by : Ronald S. Bultje <rsbultje@gmail.com>
  

  • [DH] libavcodec/vp8.c

• 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