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  • FFmpeg "Non-monotonous DTS in output stream" error when processing video from Safari's MediaRecorder

    17 juillet 2024, par Hackermon

    I'm recording a video stream in Safari with MediaRecorder, then sending it to a remote server which then uses ffmpeg to reencode the video. When reencoding with FFmpeg, I get a lot of warnings and the final video is broken, frame are glitching and out of sync but the audio sounds fine.

    


    Here's my MediaRecorder script :

    


    const camera = await navigator.mediaDevices.getUserMedia({ audio: true, video: true });
const recorder = new MediaRecorder(camera, {
        mimeType: 'video/mp4', // Safari only supports MP4
        bitsPerSecond: 1_000_000,
});

recorder.ondataavailable = async ({ data: blob }) => {
        // open contents in new tab
        var fileURL = URL.createObjectURL(file);
         window.open(fileURL, '_blank');
};

recorder.start();
setTimeout(() => recorder.stop(), 5000);


    


    I download the video blob from Safari and use this command to reencode it :

    


    ffmpeg -i ./blob.mp4 -preset ultrafast -strict -2 -threads 10 -c copy ./output.mp4


    


    Logs :

    


    ffmpeg version 4.2.7-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
  configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x559826616000] DTS 29 < 313 out of order
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './chunk1.mp4':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2024-07-17T14:30:47.000000Z
  Duration: 00:00:01.00, start: 0.000000, bitrate: 3937 kb/s
    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(progressive), 640x480 [SAR 1:1 DAR 4:3], 6218 kb/s, 33.36 fps, 600 tbr, 600 tbn, 1200 tbc (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 420 kb/s (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Audio
File './safari3.mp4' already exists. Overwrite ? [y/N] Output #0, mp4, to './safari3.mp4':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    encoder         : Lavf58.29.100
    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(progressive), 640x480 [SAR 1:1 DAR 4:3], q=2-31, 6218 kb/s, 33.36 fps, 600 tbr, 19200 tbn, 600 tbc (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 420 kb/s (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Audio
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10016, current: 928; changing to 10017. This may result in incorrect timestamps in the output file.
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10017, current: 1568; changing to 10018. This may result in incorrect timestamps in the output file.
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10018, current: 2208; changing to 10019. This may result in incorrect timestamps in the output file.
...x100
frame=  126 fps=0.0 q=-1.0 Lsize=     479kB time=00:00:00.97 bitrate=4026.2kbits/s speed= 130x    
video:425kB audio:51kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.652696%


    


    Not sure what's happening or how to fix it. This issue only happens in Safari, videos from Chrome are perfectly fine.

    


    I've tried various flags :

    


    -fflags +igndts
-bsf:a aac_adtstoasc
-c:v libvpx-vp9 -c:a libopus
etc


    


    None of them seem to fix the issue.

    


  • What is the best way to split videos into equally sized parts using ffmpeg ? [closed]

    18 juin 2024, par GBPU

    I have tried to split an mp4 file into smaller parts of equal time length like this ffmpeg -i ../data/2024-06-02_12-34-51.mp4 -c copy -map 0 -segment_time 00:00:05 -f segment v1_%03d.mp4. However, this produced videos of highly variables size, some 25x larger than others. I assume this was due to inconsistent framerate during recording.

    


    Next, I tried a script that would split based and limit each part to a specific size :

    


    #!/bin/sh
# Short script to split videos by filesize using ffmpeg by LukeLR

if [ $# -ne 3 ]; then
    echo 'Illegal number of parameters. Needs 3 parameters:'
    echo 'Usage:'
    echo './split-video.sh FILE SIZELIMIT "FFMPEG_ARGS'
    echo 
    echo 'Parameters:'
    echo '    - FILE:        Name of the video file to split'
    echo '    - SIZELIMIT:   Maximum file size of each part (in bytes)'
    echo '    - FFMPEG_ARGS: Additional arguments to pass to each ffmpeg-call'
    echo '                   (video format and quality options etc.)'
    exit 1
fi

FILE="../data/$1"
SIZELIMIT="$2"
FFMPEG_ARGS="$3"

# Duration of the source video
DURATION=$(ffprobe -i "$FILE" -show_entries format=duration -v quiet -of default=noprint_wrappers=1:nokey=1|cut -d. -f -2)

# Duration that has been encoded so far
CURDURATION=0

# Filename of the source video (without extension)
BASENAME="${FILE%.*}"

# Extension for the video parts
#EXTENSION="${FILE##*.}"
EXTENSION="mp4"

# Number of the current video part
i=1

# Filename of the next video part
NEXTFILENAME="$BASENAME-$i.$EXTENSION"

echo "Duration of source video: $DURATION"

# Until the duration of all partial videos has reached the duration of the source video
#while [[ $CUR_DURATION -lt $DURATION ]]; do
while [[ $(bc <<< "$CURDURATION < $DURATION") -eq 1 ]]; do
    # Encode next part
    echo ffmpeg -i "$FILE" -ss "$CURDURATION" -fs "$SIZELIMIT" $FFMPEG_ARGS "$NEXTFILENAME"
    ffmpeg -ss "$CURDURATION" -i "$FILE" -fs "$SIZELIMIT" $FFMPEG_ARGS "$NEXTFILENAME"

    # Duration of the new part
    NEWDURATION=$(ffprobe -i "$NEXTFILENAME" -show_entries format=duration -v quiet -of default=noprint_wrappers=1:nokey=1|cut -d. -f -2)

    # Total duration encoded so far
    echo $CURDURATION
    CURDURATION=$(bc <<< "$CURDURATION + $NEWDURATION")
    echo $CURDURATION

    i=$((i + 1))

    echo "Duration of $NEXTFILENAME: $NEWDURATION"
    echo "Part No. $i starts at $CURDURATION"
    echo "Current Duration: $CURDURATION"

    NEXTFILENAME="$BASENAME-$i.$EXTENSION"
done


    


    I call the script like this : bash split-video.sh 2024-06-02_12-34-51.mp4 10000000 "-c copy"
Unfortunately, this has an issue where some of the sub videos are extremely short and have wildly inconsistent numbers of frames in them (some with nearly 400, others with 1), despite being similar sizes. I am guessing this has something to do with inconsistent framerate and keyframes or something ?

    


    I am curious what the best way to split a video into equally sized parts, and ideally with similar numbers of frames, is using ffmpeg.

    


  • How to convert rtmp hevc video stream to srt av1 endpoint with ffmpeg ?

    20 juin 2024, par Lulík

    i want use ffmpeg to listen rtmp stream and send to srt endpoint.

    


    Flow : smartphone (camera) -> laptop (ffmpeg script) -> desktop (obs studio)

    


    ffmpeg script show warning message and in obs stuido i can see any video only audio.

    


    Thank you in advance.

    


    Console output while running script (error in the end is bcs i stoped sending data from phone) :

    


    ffmpeg version git-2024-06-20-8d6014d Copyright (c) 2000-2024 the FFmpeg developers
  built with gcc 12 (Debian 12.2.0-14)
  configuration: --enable-libsvtav1 --enable-libsrt
  libavutil      59. 24.100 / 59. 24.100
  libavcodec     61.  8.100 / 61.  8.100
  libavformat    61.  3.104 / 61.  3.104
  libavdevice    61.  2.100 / 61.  2.100
  libavfilter    10.  2.102 / 10.  2.102
  libswscale      8.  2.100 /  8.  2.100
  libswresample   5.  2.100 /  5.  2.100
Input #0, flv, from 'rtmp://192.168.0.194/s/streamKey':
  Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
  Stream #0:0: Video: hevc (Main), yuv420p(tv, smpte170m/bt470bg/smpte170m), 1080x1920, 10240 kb/s, 30 fps, 120 tbr, 1k tbn
  Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 131 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (hevc (native) -> av1 (libsvtav1))
  Stream #0:1 -> #0:1 (aac (native) -> mp2 (native))
Press [q] to stop, [?] for help
Svt[info]: -------------------------------------------
Svt[info]: SVT [version]:   SVT-AV1 Encoder Lib 595a874
Svt[info]: SVT [build]  :   GCC 12.2.0   64 bit
Svt[info]: LIB Build date: Jun 20 2024 14:25:08
Svt[info]: -------------------------------------------
Svt[info]: Number of logical cores available: 12
Svt[info]: Number of PPCS 76
Svt[info]: [asm level on system : up to avx2]
Svt[info]: [asm level selected : up to avx2]
Svt[info]: -------------------------------------------
Svt[info]: SVT [config]: main profile   tier (auto) level (auto)
Svt[info]: SVT [config]: width / height / fps numerator / fps denominator       : 1080 / 1920 / 120 / 1
Svt[info]: SVT [config]: bit-depth / color format                   : 8 / YUV420
Svt[info]: SVT [config]: preset / tune / pred struct                    : 10 / PSNR / random access
Svt[info]: SVT [config]: gop size / mini-gop size / key-frame type          : 641 / 16 / key frame
Svt[info]: SVT [config]: BRC mode / rate factor                     : CRF / 35 
Svt[info]: SVT [config]: AQ mode / variance boost                   : 2 / 0
Svt[info]: -------------------------------------------
Svt[warn]: Failed to set thread priority: Invalid argument
Output #0, mpegts, to 'srt://192.168.0.167:9998?mode=caller':
  Metadata:
    encoder         : Lavf61.3.104
  Stream #0:0: Video: av1, yuv420p(tv, smpte170m/bt470bg/smpte170m, progressive), 1080x1920, q=2-31, 120 fps, 90k tbn
      Metadata:
        encoder         : Lavc61.8.100 libsvtav1
  Stream #0:1: Audio: mp2, 44100 Hz, stereo, s16, 384 kb/s
      Metadata:
        encoder         : Lavc61.8.100 mp2
[mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading.
[in#0/flv @ 0x55ec9219cc40] Error during demuxing: Input/output error1990.7kbits/s speed=0.967x    
[out#0/mpegts @ 0x55ec922247c0] video:4431KiB audio:1138KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 6.374870%
frame=  723 fps= 31 q=35.0 Lsize=    5923KiB time=00:00:24.12 bitrate=2011.3kbits/s speed=1.04x


    


    I send video stream from mobile app over rtmp encoded with hevc to my laptop where running script ffmpeg -f flv -listen 1 -i rtmp://192.168.0.194/s/streamKey -c:v libsvtav1 -f mpegts srt://192.168.0.167:9998?mode=caller. On the desktop i have obs with media source input srt://192.168.0.167:9998?mode=listener.

    


    When i run ffmpeg script without video codec option (-c:v libsvtav1) its working fine and in obs i can see video from my phone camera. With the option i can not see video only audio.
I clearly dont understand warning message : [mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading..
Do I need specify codec (av1) in obs media source or my ffmpeg script is wrong ?