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• FFMPEG Errors : "max_analyze_duration" "buffer underflow" "packet too large" What to do ?

  8 février 2013, par Leon5x

  I have an error while just copying a video with ffmpeg.
  I use the command :

  ffmpeg -i leon.mpg -vcodec copy -acodec copy leon2.mpg

  The errors :

  [mpeg @ 00000000020ebd20] max_analyze_duration 5000000 reached at 5004000 microseconds
  
  [mpeg @ 00000000042f4020] buffer underflow i=0 bufi=11286 size=14824
  
  [mpeg @ 00000000042f4020] packet too large, ignoring buffer limits to mux it

  What do I have to set that the buffer errors don't occur anymore ?

  Here is a picture of what ffmpeg gives out first. After that the red errors repeat really many times. There you also can see what the film's codec and so on is.
  See this picture :

  I use the ffmpeg Version git-5ce023b (2013-01-15) - Win64 - static build from Zeranoe.
  I tried some things but nothing happened. I searched a while but found no solution.

• Error while extracting subtitle from mkv or m2ts to srt with FFmpeg

  14 février 2013, par user2071701

  I need to extract subtitles from different video files to .srt format (to use it in html5 video).
  I tried a lot of variant i found with google. But every time i get this error :

  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  I think, this error means that ffmpeg can decode source subtitle but cant encode it to .srt format. All codecs a enabled (i compiled the later ffmpeg version from git a few times with a different configuration).

  Here is the output :

      # /usr/local/bin/ffmpeg -i /var/video/sources/Balbesy1.m2ts -an -vn  -copyinkf -scodec srt -f srt -y  sub.srt
  
  ffmpeg version N-49947-g9f16cb9 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Feb 14 2013 14:26:10 with gcc 4.4.5 (Debian 4.4.5-8)
  
    configuration: --enable-encoder=dvdsub --enable-decoder=dvdsub --enable-decoder=pgssub --enable-encoder=srt --enable-decoder=srt --enable-encoder=srt --enable-decoder=srt
  
    libavutil      52. 17.101 / 52. 17.101
  
    libavcodec     54. 91.103 / 54. 91.103
  
    libavformat    54. 63.100 / 54. 63.100
  
    libavdevice    54.  3.103 / 54.  3.103
  
    libavfilter     3. 37.101 /  3. 37.101
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
  [mpegts @ 0x2719040] Stream #5: not enough frames to estimate rate; consider increasing probesize
  
  [mpegts @ 0x2719040] Could not find codec parameters for stream 5 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [NULL @ 0x271fec0] start time is not set in estimate_timings_from_pts
  
  Input #0, mpegts, from '/var/video/sources/Balbesy1.m2ts':
  
    Duration: 01:18:11.89, start: 599.958300, bitrate: 37378 kb/s
  
    Program 1 
  
      Stream #0:0[0x1011]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc
  
      Stream #0:1[0x1100]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 0x0086), 48000 Hz, 5.1(side), fltp, 768 kb/s
  
      Stream #0:2[0x1101]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 0x0086), 48000 Hz, 5.1(side), fltp, 768 kb/s
  
      Stream #0:3[0x1102]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Stream #0:4[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s
  
      Stream #0:5[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)
  
  Output #0, srt, to 'sub.srt':
  
      Stream #0:0: Subtitle: srt
  
  Stream mapping:
  
    Stream #0:5 -> #0:0 (pgssub -> srt)
  
  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  sorry for my english

• How can I encode and segment audio files without having gaps (or audio pops) between segments when I reconstruct it ?

  16 mai 2013, par fenduru

  I'm working on a web application that requires streaming and synchronization of multiple audio files. For this, I am using the Web Audio API over HTML5 audio tags because of the importance of timing audio.

  Currently, I'm using FFMPEG's segmentation feature to encode and segment the audio files into smaller chunks. The reason I am segmenting them is so I can start streaming from the middle of the file instead of starting from the beginning (otherwise I would've just split the files using UNIX split, as shown here. The problem is that when I string the audio segments back together, I get an audio pop between segments.

  If I encode the segments using a PCM encoding (pcm_s24le) in a .wav file, the playback is seamless, which leads me to believe that the encoder is padding either the beginning or the end of the file. Since I will be dealing with many different audio files, using .wav would require far too much bandwidth.

  I'm looking to one of the following solutions to the problem :

  • How can I segment encoded audio files seamlessly,
  • How can I force an encoder to NOT pad audio frames using ffmpeg (or another utility), or
  • What is a better way to stream audio (starting at an arbitrary track time) without using an audio tag ?

  System Information

  • Custom node.js server
  • Upon upload of an audio file, node.js pipes the data into ffmpeg's encoder
  • Need to use HTML5 Web Audio API supported encoding
  • Server sends audio chunks 1 at a time through a WebSockets socket

  Thanks in advance. I've tried to be as clear as possible but if you need clarification I'd be more than willing to provide it.