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• ffmpeg command to copy raw data into MP4 hangs and never finishes

  28 mai 2019, par James Adams

  I am running an ffmpeg command to copy raw H.265 data into an MP4 file and the command never completes. The file isn’t large, just 10 seconds worth of data. Here’s the command I’m running :

  $ ffmpeg -rtsp_transport tcp -i rtsp://user:pass@71.185.124.195:554/c1/b1558830329/e1558830339/replay/ -vcodec copy -y test_clip.mp4

  I then get output like this :

  ffmpeg version 3.4.6-0ubuntu0.18.04.1 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Guessed Channel Layout for Input Stream #0.1 : mono
  
  Input #0, rtsp, from 'rtsp://user:pass@71.85.104.195:554/c1/b1558830329/e1558830339/replay/':
  
    Metadata:
  
      title           : ONVIF RTSP Server
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709, progressive), 1920x1080, 30 fps, 30 tbr, 90k tbn, 60 tbc
  
      Stream #0:1: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:1 -> #0:1 (pcm_mulaw (native) -> aac (native))
  
  Press [q] to stop, [?] for help
  
  [aac @ 0x55b71ce31900] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
  
  Output #0, mp4, to 'test_clip.mp4':
  
    Metadata:
  
      title           : ONVIF RTSP Server
  
      encoder         : Lavf57.83.100
  
      Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 30 fps, 30 tbr, 90k tbn, 90k tbc
  
      Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 8000 Hz, mono, fltp, 48 kb/s
  
      Metadata:
  
        encoder         : Lavc57.107.100 aac
  
  [mp4 @ 0x55b71ce17e00] Non-monotonous DTS in output stream 0:0; previous: 18000, current: 3000; changing to 18001. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x55b71ce17e00] Non-monotonous DTS in output stream 0:0; previous: 18001, current: 6000; changing to 18002. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x55b71ce17e00] Non-monotonous DTS in output stream 0:0; previous: 18002, current: 9000; changing to 18003. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x55b71ce17e00] Non-monotonous DTS in output stream 0:0; previous: 18003, current: 12000; changing to 18004. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x55b71ce17e00] Non-monotonous DTS in output stream 0:0; previous: 18004, current: 15000; changing to 18005. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x55b71ce17e00] Non-monotonous DTS in output stream 0:0; previous: 18005, current: 18000; changing to 18006. This may result in incorrect timestamps in the output file.
  
  frame=   44 fps=0.0 q=-1.0 size=     256kB time=00:00:01.43 bitrate=1463.4kbits/frame=   60 fps= 57 q=-1.0 size=     512kB time=00:00:01.96 bitrate=2132.9kbits/frame=   76 fps= 48 q=-1.0 size=     768kB time=00:00:02.50 bitrate=2516.7kbits/frame=   92 fps= 44 q=-1.0 size=    1024kB time=00:00:03.03 bitrate=2765.6kbits/frame=  108 fps= 41 q=-1.0 size=    1024kB time=00:00:03.56 bitrate=2352.0kbits/[NULL @ 0x55b71cdfa540] SEI type 5 size 2208 truncated at 1944
  
  frame=  123 fps= 39 q=-1.0 size=    1280kB time=00:00:04.06 bitrate=2578.6kbits/frame=  139 fps= 38 q=-1.0 size=    1536kB time=00:00:04.60 bitrate=2735.5kbits/frame=  155 fps= 37 q=-1.0 size=    1536kB time=00:00:05.13 bitrate=2451.3kbits/frame=  171 fps= 36 q=-1.0 size=    1792kB time=00:00:05.66 bitrate=2590.7kbits/frame=  187 fps= 36 q=-1.0 size=    2048kB time=00:00:06.20 bitrate=2706.1kbits/frame=  203 fps= 35 q=-1.0 size=    2304kB time=00:00:06.73 bitrate=2803.2kbits/frame=  219 fps= 35 q=-1.0 size=    2304kB time=00:00:07.26 bitrate=2597.4kbits/frame=  235 fps= 34 q=-1.0 size=    2560kB time=00:00:07.80 bitrate=2688.7kbits/frame=  246 fps= 33 q=-1.0 size=    2560kB time=00:00:08.16 bitrate=2568.0kbits/frame=  267 fps= 34 q=-1.0 size=    3072kB time=00:00:08.86 bitrate=2838.3kbits/frame=  282 fps= 34 q=-1.0 size=    3072kB time=00:00:09.36 bitrate=2686.8kbits/frame=  298 fps= 33 q=-1.0 size=    3328kB time=00:00:09.90 bitrate=2753.9kbits/frame=  314 fps= 33 q=-1.0 size=    3328kB time=00:00:10.43 bitrate=2613.1kbits/^Cspeed=1.11x    

  The command never completes and I need to kill it using Ctrl-C.

  I have also tried adding the options -nostdin -loglevel error and appending this to the end of the command : > /dev/null 2>&1 < /dev/null but to no avail.

  I am testing the above since it mimics the actual code I’m developing which utilizes a Python package that wraps calls to ffmpeg. The Python code below works well on a laptop but hangs on EC2 (both are Ubuntu 18.04) :

  import argparse
  
  import datetime
  
  import ffmpeg
  
  
  
  
  
  # ------------------------------------------------------------------------------
  
  if __name__ == "__main__":
  
  
  
      # USAGE
  
      # $ python collect_and_store.py --rtsp rtsp://user:pass1@71.85.125.110:554 \
  
      #       --duration 30 --count 10
  
  
  
      # construct the argument parser and parse the arguments
  
      args_parser = argparse.ArgumentParser()
  
      args_parser.add_argument("--rtsp",
  
                               required=True,
  
                               type=str,
  
                               help="RTSP URL for video stream")
  
      args_parser.add_argument("--duration",
  
                               required=True,
  
                               type=int,
  
                               help="duration of saved clips (in seconds)")
  
      args_parser.add_argument("--count",
  
                               required=True,
  
                               type=int,
  
                               help="number of clips to save")
  
      args = vars(args_parser.parse_args())
  
  
  
      # sanity check for some of the arguments
  
      if not args["rtsp"].lower().startswith("rtsp://"):
  
          raise ValueError("Invalid input URL -- only RTSP supported")
  
  
  
      seconds_per_clip = args["duration"]
  
      start = int(datetime.datetime.now().strftime("%s"))
  
      end = start + seconds_per_clip
  
      number_of_files_to_collect = args["count"]
  
  
  
      while number_of_files_to_collect > 0:
  
  
  
          # build URL with start and end times
  
          # NOTE URL is for Uniview RTSP, add options for other camera types
  
          url = args["rtsp"] + f"/c1/b{start}/e{end}/replay/"
  
  
  
          # file where we'll write clip data
  
          temp_file = f"clip_b{start}_e{end}.mp4"
  
  
  
          # create the equivalent of the ffmpeg command:
  
          # $ ffmpeg -i  -vcodec copy -y -rtsp_transport tcp 
  
          stream = ffmpeg.input(url)
  
          stream = ffmpeg.output(stream, temp_file,
  
                                 **{"codec:v": "copy",
  
                                    "rtsp_transport": "tcp",
  
                                    "y": None
  
                                    }
  
                                 )
  
          ffmpeg.run(stream)
  
  
  
          print(f"\n\nMP4 file created: {temp_file}")
  
  
  
          number_of_files_to_collect -= 1
  
          start = end + 1
  
          end = start + seconds_per_clip

• Adding ffmpeg OMX codec to Genymotion Android 4.4.2 emulator

  22 avril 2016, par photon

  Basic Question :

  Is there a way to add a new audio codec to the Genymotion Android emulator, short of downloading the entire Android source, learning how to build it, and creating my own version of Android ?

  ---

  Context :

  I have written a java Android app that acts as an audio renderer, as well as being a DLNA/OpenHome server and client. Think "BubbleUpnp" without video. My primary development platform is Win8.1. The program started as an ActiveState "pure-perl" DLNA MediaServer on Windows, which I then ported to Ubuntu, which I got working under Android a few years ago. It was pretty funky ... all UI being presented thru an HTTP server/jquery/jquery-ui, served from an Ubuntu shell running under Android (a trick in itself), serving up HTML pages to Chrome running on the same (Android) device. Besides being "funky" it had a major drawback that it required a valid IP address to work ... as I could not figure out how to get ubuntu to have a local loopback device for a 127.0.0.01 localhost I use the app as a "car stereo" on my boat (which is my home), which is often not hooked up to the internet.

  I had a hard time getting started in Android app development because the speed of the Android emulators in Eclipse was horrid, and the ADB drivers did not work from Win8 for the longest time.

  Then one day, about a year ago, I ran into Genymotion (kudos to the authors), and all of a sudden I had a workable Android development environment, so I added a Java implementation of the DLNA server, which then grew into a renderer also, using Android’s MediaPlayer class, and, adding the ability to act as a DLNA control point, and more recently also added OpenHome servers and renderers to it.

  In a separate effort, I created a build environment for this program called fpCalc, based on ffMpeg, on a variety of platforms, including Win, Linux, and Android x86, arm, and arm7 devices (bitbucket.org/phorton1/) and did an extensive series of tests to determine the validity, and longevity of fpcalc fingerprints, discovering that the fpCalc fingerprint changed based on the version of ffmpeg it was built against, a separate topic to be sure, but in the process, learned at least a bit about how to build ffmpeg as well as Android shared libraries, JNI interfaces, etc.

  So now the Android-Java version of the program has advanced past the old perl version, and I am debating whether I want to continue to try to build the perl version (and or add an wxPerl UI) to it.

  One issue that has arisen, for me, is that the Genymotion emulator does not support WMA decoding ... as Android dropped support for WMA due to licensing issues, etc, a ways back in time ... yet my music library has significant numbers of tunes in WMA files, and I don’t want to "convert" them, my carefully thought-out philosophy is that my program does not modify the contents, or tags, or anything in the original media files that I have accumulated, or will receive in the future, rather treating them as "artifacts" worth preserving "as is". No conversion is going to make a file "better" than it was, and I wish to preserve ALL of the original sources for ALL of my music going forward.

  So, I’m thinking, gee, I can build FFMPEG on 7 different platforms, and I see all these references to "OMX FFMPEG Codec Support for Android" on the net, so I’m thinking, "All I need to do is create the OMX Component and somehow get it into Genymotion".

  I have studied up OMX, OpenMaxIL, seen Michael Chen’s posts, seen the stack overflow questions

  How to make ffmpeg codec componet as OMX component

  and

  Android : How to integrate a decoder to multimedia framework

  and Cedric Fung’s page https://vec.io/posts/use-android-hardware-decoder-with-omxcodec-in-ndk, and Michael Chen’s repository at https://github.com/omxcodec , as well as virtually every other page on the net that mentions any combination of libstagefright, OMX, Genymotion, and FFMPEG.

  (this page would not let me put more than 2 links as i don’t have a "10" reputation, or I would have listed some of the sources I have seen) ..

  My Linux development environment is a Ubuntu12.04 vbox running on my win machine. I have downloaded and run the Android-x86 iso as a vbox, and IT contains the ffmpeg codecs, but unfortunately, it neither supports a wifi interface, nor the vbox "guest additions", so it has a really funky mouse. I tried for about 3 days to address those two issues, but in the end do not feel it is usable for my puproses, and I really like the way genymotion "feels", particularly the moust support, so I’d like to keep genymotion as my "windows android" virtual device under which I may run my program, deprecate and stop using my old perl source,

  except genymotion does not support WMA files ...

  ---

  Several side notes :

  (a) There is no good way to write a single sourced application in Java that runs natively in Windows, AND as an Android app.

  (b) I don’t want to reboot my Windows machine to a "real" Android device just to play my music files. The machine has to stay in Windows as I use it for other things as well.

  (c) I am writing this as my machine is in the 36th hour of downloading the entire ASOP source code base to a partition in my Ubuntu vbox while I am sitting in a hotel room on a not-so-good internet connection in Panama City, Panama, before I return to my boat in remote Bocas Del Toro Panama, where the internet connection is even worse.

  (d) I did get WMA decoding to work in my app by calling my FFMPEG executable from Java (converting it to either WAV/PCM or AAC), but, because of limitations in Android’s MediaPlayer, it does not work well, particularly for remotely hosted WMA files ... MediaPlayer insists on having the whole file present before it starts to play, which can take several seconds or longer, and I am hoping that by getting a ’real’ WMA codec underneath MediaPlayer, that problem will just disappear ....

  ---

  So, I’m trying to figure this whole mess out. There are a lot of tantalizing clues, and suggestions, but what I have found, or at least what I am starting to believe, is that if I want to add a simple WMA audio decoding codec to Android (Genymotion), not only do I have to download, basically, the ENTIRE ASOP Android source tree, and learn a new set of tools (repo, etc), but I have to (be able to) rebuild, from scratch, the entire Android system, esp. libstagefright.so in such a way as to be COMPLETELY compatible with the existing one in GenyMotion, while at the same time adding ffmpeg codecs ala Michael Chen’s page.

  And I’m just asking, is it, could it really be that difficult ?

  ---

  Anyways, this makes me crazy. Is there no way to just build a new component, or at worst a new OMX core, and add it to Genymotion, WITHOUT building all of Android, and preferably, based only on the OMX h files ? Or do I REALLY have to replace the existing libstagefright.so, which means, basically, rebuilding all of Android ...

  p.s. I thought it would be nice to get this figured out, build it, and then post the installable new FFMPEG codecs someplace for other people to use, so that they don’t also grow warts on their ears and have steam shooting out of their eyeballs, while they get old trying to figure it out ....

• How to fix : Frame rate differs from source after encoding certain mp4 videos to wmv

  25 juin 2019, par hpenhp

  I’m having trouble with a usually working ffmpeg script for encoding mp4 and more to wmv. The goal is to keep it like the original, except for the resolution (smaller) and the format (wmv).

  When it usually works for a lot of videos, on some of them it doesn’t... and i would like to understand and avoid that.

  I tried several things, forcing the framerate for the output, using -vsync 2 option, -vsync 0 also.. noone of that fixed the issue.

  An other thing, worth adding, is that the fps value from media info is good (24 fps) on the output like the input, but through the M$ windows explorer, it says 30 frames per sec, and when you play the video you see that the text that was matching the input doesn’t match the output any more (because of the fps mismatching).

  Below the script i use :

  ffmpeg -threads auto -i input.mp4 -s 480*320 -b:v 1000k -vcodec msmpeg4 -acodec wmav2 -f asf output.wmv

  Here is the ffmpeg log :

  ffmpeg version 3.4.1-1~16.04.york0 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.5) 20160609
  
    configuration: --prefix=/usr --extra-version='1~16.04.york0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    WARNING: library configuration mismatch
  
    avcodec     configuration: --prefix=/usr --extra-version='1~16.04.york0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared --enable-version3 --disable-doc --disable-programs --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './test_2k-24__hq_960x540.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: mp42mp41isomavc1
  
      creation_time   : 2019-05-27T14:37:50.000000Z
  
    Duration: 01:28:52.80, start: 0.000000, bitrate: 1232 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, smpte170m), 960x540, 1091 kb/s, 24 fps, 24 tbr, 24 tbn, 48 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-05-27T14:37:50.000000Z
  
        handler_name    : L-SMASH Video Handler
  
        encoder         : AVC Coding
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 137 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-05-27T14:37:50.000000Z
  
        handler_name    : L-SMASH Audio Handler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> msmpeg4v3 (msmpeg4))
  
    Stream #0:1 -> #0:1 (aac (native) -> wmav2 (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, asf, to '/nas/copiedetravail/test_2k-24__hq_960x540.mp4.wmv':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: mp42mp41isomavc1
  
      WM/EncodingSettings: Lavf57.83.100
  
      Stream #0:0(und): Video: msmpeg4v3 (msmpeg4) (MP43 / 0x3334504D), yuv420p(progressive), 480x320, q=2-31, 1000 kb/s, 24 fps, 1k tbn, 24 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-05-27T14:37:50.000000Z
  
        handler_name    : L-SMASH Video Handler
  
        encoder         : Lavc57.107.100 msmpeg4
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/1000000 buffer size: 0 vbv_delay: -1
  
      Stream #0:1(und): Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, stereo, fltp, 128 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-05-27T14:37:50.000000Z
  
        handler_name    : L-SMASH Audio Handler
  
        encoder         : Lavc57.107.100 wmav2
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 31 times
  
  frame=  228 fps=0.0 q=2.0 size=     291kB time=00:00:09.45 bitrate= 252.4kbits/s speed=18.9x    
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 1 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
      Last message repeated 1 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
      Last message repeated 1 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
      Last message repeated 5 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 2 times
  
  frame=  449 fps=449 q=2.0 size=     663kB time=00:00:18.98 bitrate= 286.2kbits/s speed=  19x    
  
  frame=  629 fps=418 q=1.6 size=    1307kB time=00:00:26.16 bitrate= 409.2kbits/s speed=17.4x    
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 30 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 7 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 2 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 34 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 1 times
  
  frame=  821 fps=410 q=2.0 size=    1960kB time=00:00:34.26 bitrate= 468.7kbits/s speed=17.1x    
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 71 times
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 2 dct coefficients to -127..127
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 82 times
  
  frame= 1026 fps=409 q=2.0 size=    2695kB time=00:00:42.96 bitrate= 513.8kbits/s speed=17.1x    
  
  [msmpeg4 @ 0x5648e407cb60] warning, clipping 1 dct coefficients to -127..127
  
      Last message repeated 102 times
  
  frame= 1202 fps=400 q=1.6 size=    3754kB time=00:00:50.13 bitrate= 613.4kbits/s speed=16.7x  
  
  ......
  
  frame=127987 fps=301 q=2.0 Lsize=  753842kB time=01:28:52.77 bitrate=1158.0kbits/s speed=12.6x    
  
  video:650704kB audio:83244kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.710554%

  EDIT :
  My workaround was to encode to MPEG, doing this, i’m able to read my file and the frame rate is same as source. If someone knows why it couldn’t keep the original framerate in wmv, i’m still interested to know.