Newest 'ffmpeg' Questions - Stack Overflow

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  • Mapping Video streaming URL with SPRING MVC

    31 mai 2017, par Valeriane

    I'm developping a site for video streamings.

    My application server works with spring MVC (front is managed by angular).

    I have another server, that provides stream flux (uses ffmpeg).

    Today I have direct URL to my video server, something like that : http://myvideos_server.com/videos/video.m3u8

    this URL is detectable client side....

    Is it any technic do not provide my videos streaming URL directly ? Is it possible mapping videos stream URLs with Spring MVC ?

  • converting video with avconv while capturing

    31 mai 2017, par Oliver

    I am successfully capturing a video stream with that device: Easycap DC60

    But I need to run the following commands in serial:

    First that:

    sudo killall -9 somagic-both

    sudo killall -9 somagic-capture

    sudo somagic-init

    sudo somagic-both -n 1>.video 2>.audio

    [CTRL+C]

    Then that:

    avconv \

    -f rawvideo -pix_fmt uyvy422 -r 30 -s:v 720x480 -i .video \

    -f s16le -sample_rate 24000 -ac 2 -i .audio -strict experimental \

    -vcodec mpeg4 -vtag xvid -qscale:v 7 \

    -vf yadif -s:v 720x540 \

    video.avi

    The problem is that when I try to run them together in a row (with "&" at the end of "sudo somagic-both -n 1>.video 2>.audio") I get that error:

    ffmpeg version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2000-2017 the FFmpeg developers
      built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
      configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --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-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
      libavutil      54. 31.100 / 54. 31.100
      libavcodec     56. 60.100 / 56. 60.100
      libavformat    56. 40.101 / 56. 40.101
      libavdevice    56.  4.100 / 56.  4.100
      libavfilter     5. 40.101 /  5. 40.101
      libavresample   2.  1.  0 /  2.  1.  0
      libswscale      3.  1.101 /  3.  1.101
      libswresample   1.  2.101 /  1.  2.101
      libpostproc    53.  3.100 / 53.  3.100
    Input #0, rawvideo, from '.video':
      Duration: N/A, bitrate: 165888 kb/s
        Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x480, 165888 kb/s, 30 tbr, 30 tbn, 30 tbc
    [s16le @ 0x24d7480] Estimating duration from bitrate, this may be inaccurate
    Guessed Channel Layout for  Input Stream #1.0 : stereo
    Input #1, s16le, from '.audio':
      Duration: 00:00:00.71, bitrate: 767 kb/s
        Stream #1:0: Audio: pcm_s16le, 24000 Hz, 2 channels, s16, 768 kb/s
    File 'video.avi' already exists. Overwrite ? [y/N] y
    Output #0, avi, to 'video.avi':
      Metadata:
        ISFT            : Lavf56.40.101
        Stream #0:0: Video: mpeg4 (xvid / 0x64697678), yuv420p, 720x540, q=2-31, 200 kb/s, 30 fps, 30 tbn, 30 tbc
        Metadata:
          encoder         : Lavc56.60.100 mpeg4
        Stream #0:1: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 24000 Hz, stereo, s16p
        Metadata:
          encoder         : Lavc56.60.100 libmp3lame
    Stream mapping:
      Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg4 (native))
      Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    Multiple frames in a packet from stream 0
    [pcm_s16le @ 0x24f33c0] Invalid PCM packet, data has size 2 but at least a size of 4 was expected
    Error while decoding stream #1:0: Invalid data found when processing input
    frame=    0 fps=0.0 q=0.0 Lsize=      38kB time=00:00:03.19 bitrate=  98.0kbits/s    
    video:0kB audio:25kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 50.802467%
    

    I think it happens because it is converting more quickly than it is capturing. I tried to use the solution given here without success.

    Anyone ever faced that situation before?

  • How to show segment lengths of hls file with ffmpeg

    30 mai 2017, par endorphins

    I'm trying to use ffmpeg to retrieve the segment duration of every segment for every stream of an HLS file. It looks like I can do this if I know all of the names of the individual files in advance and call ffprobe on each one. This seems really tedious and I won't always know the names of all of the individual files, only the main .m3u8. Is there any way I can get all of this information with just one ffmpeg command?

    Additionally, is the duration retrieved by ffprobe on a .aac file accurate? I get the message that it's "Estimating duration from bitrate, this may be inaccurate".

  • ffmpeg copy command in MLT Multimedia Framework

    30 mai 2017, par Jmv Jmv

    Regards,

    What is the equivalent of this ffmpeg command

    ffmpeg -i in.mp4 -c copy out.mp4
    

    in MLT Multimedia Framework MLT?

    Thank very much!

  • iOS - Convert Audio Format (opus to mp3)

    30 mai 2017, par Android0077

    Recently I started to develop application that work with .opus file (Audio Format).

    I am working with external SDK that can processor a mp3/wav file, unfortunately my local file is a .opus file and I need to convert it to mp3/wav format in order to process the file.

    I read and research a lot around the network to find a solution, I found the FFmpegWrapper library that can convert two type of Audio Format but when I try to convert .opus to .mp3/ , I get this error: opus codec not supported in WAVE format I do not know what can be done, I'll be happy to help.

    Any information about how to convert .Opus format to any other format will be appreciated.

    Thanks