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• Remux and segment only parts of a video file without difference in output

  20 juin 2013, par Christian P.

  I have a working program built on top of libav (alternatively ffmpeg - expertise is either is useful here).

  It takes an mp4 video, encoded with h264 video / AAC audio, and remuxes it to MPEG TS and segments it into X second chunks. It is analogous to the following ffmpeg command :

  ffmpeg -y -i video.mp4 -c:a copy -bsf:a aac_adtstoasc -c:v copy -bsf:v h264_mp4toannexb -flags -global_header -map 0 -f segment -segment_time 10 -segment_list playlist.m3u8 -segment_format mpegts chunk_%03d.ts

  The reason I am not using the command-line, is that I wish to generate only a subset of the segments. So if a video results in 10 segments of between 8 and 12 seconds (the segments are never exactly the desired length due to keyframes), I might wish to generate segments 3-7 at a later time.

  The complete code for my program can be found here.

  I use av_read_frame to read every frame from the source file, remux (including a bitfilter process) and write to output file. Once the duration since the last output becomes close/greater than the desired segment length, I flush the output file, close it, open the next segment and continue.

  I have tried altering the code to do an av_seek_frame to the end of the first segment and start from there (I also attempted to start at the end of the 2nd and 3rd segment). The new segments are the same length (in seconds and pts), but have a different size than the comparable segments from the full runthrough (within a few kilobytes) - the starting segment (whether it's the 2nd, 3rd or other) also shows as having 2 packets LESS than the comparable segment from previously.

  I assumed that av_seek_frame would give me an exact match as if I had manually done a loop with av_read_frame up to that frame, but it seems like it's not the case.

  What I wish for :

  • A way to "fast-forward" in the file to a specific (not approximate) point in the file.
  • To write from that point forward and have the output be completely identical to the output a full run provides (same size, same length, same exact bytes).

• how to configure ffserver to save the incoming feed in a different file every 30 mins or so

  29 mai 2014, par Muhammad Ali

  I want to have a backlog of my camera video in files of 30 mins duration. What i’ve read so far from the internet about ffserver has allowed me to connect all my cameras to ffserver using ffmpeg and receive the video on another system using VLC or ffplay .

  Now what I want is to store the camera video in a different file preferably named as time-stamp. And keep streaming the live video.

  Once there is a list of 30 mins files. I’d like to have a playlist which can be opened in VLC player of files that have been saved and can be played.

  sort of like a remote media player with video sources coming from ffserver with playlist and 30 mins duration video clips.

• Generating 64kbps audio-only mpegts for HTTP Live segmenter to meet 64kbps audio only requirement

  14 juin 2013, par Pobre

  I am trying to convert our mp4 files into mpeg-ts and segment it into .ts files for my iphone app to play. I am using Carson McDonalds's HTTP-Live-Video-Stream-Segmenter-and-Distributor to do that.

  I got his stuff complied and working correctly. I am currently trying to meet Apple's requirement where I need to provide a baseline 64 kbps audio only stream to my m3u8 playlist.
  Carson doesn't seem to have a profile for that.

  I need to be able to generate 64kbps audio-only stream from mp4, and turn that into mpeg-ts for the segmenter into ts. I am trying to find the right ffmpeg command that will validate without problem using Apple's mediastreamvalidator.

  So far I modified an existing encoding profile to try to achieve 64kbps total :

  ffmpeg -er 4 -i %s -f mpegts -acodec libmp3lame -ar 22050 -ab 32k -s 240x180 -vcodec libx264 -b 16k -flags +loop+mv4 -cmp 256 -partitions +parti4x4+partp8x8+partb8x8 -subq 7 -trellis 1 -refs 5 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 64k -maxrate 16k -bufsize 16k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 4:3 -r 10 -g 30 -async 2 - | %s %s %s %s %s

  but then when I try to validate it using mediastreamvalidator, it gives error after few ts :

  Playlist Validation : OK

  Segments :

  sample_cell_4x3_64k-00001.ts :

  WARNING : Media segment exceeds target duration of 10.00 seconds by 1.30 seconds (segment duration is 11.30 seconds)

  sample_cell_4x3_64k-00002.ts :

  WARNING : Media segment exceeds target duration of 10.00 seconds by 1.40 seconds (segment duration is 11.40 seconds)

  ....
  ....

  sample_cell_4x3_64k-00006.ts :

  ERROR : (-1) Unknown video codec : 1836069494 (program 0, track 0)
  ERROR : (-1) failed to parse segment as either an MPEG-2 TS or an ES

  sample_cell_4x3_64k-00007.ts :

  ERROR : (-1) Unknown video codec : 1836069494 (program 0, track 0)
  ERROR : (-1) failed to parse segment as either an MPEG-2 TS or an ES

  ....
  ....
  Average segment duration : 10.26 seconds
  Average segment bitrate : 376797.92 bps
  Average segment structural overhead : 349242.17 bps (92.69 %)

  Is there someway I can generate this correctly with just audio which totals 64kbps and turn it into mpeg-ts ready to be segmented and validated correctly ?

  Am I approaching the problem right ?