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  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

  • L’espace de configuration de MediaSPIP

    29 novembre 2010, par

    L’espace de configuration de MediaSPIP est réservé aux administrateurs. Un lien de menu "administrer" est généralement affiché en haut de la page [1].
    Il permet de configurer finement votre site.
    La navigation de cet espace de configuration est divisé en trois parties : la configuration générale du site qui permet notamment de modifier : les informations principales concernant le site (...)

  • Contribute to a better visual interface

    13 avril 2011

    MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
    Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.

Sur d’autres sites (9302)

  • ffmpeg itsoffset doesnt apply offset to mp4

    26 juillet 2019, par brux

    The following command mixes an mp4 and an mp3 together, keeping the audio from the mp4.

    ffmpeg -i video.mp4 -i audio.mp3 -map 0:v -c:v copy -filter_complex '[0:a][1:a]amix[aout]' -map '[aout]' -shortest out.mp4

    The command works as expected.

    Now I want to offset the mp4 file (both the audio and video stream of the mp4) so that there is a delay of 500ms at the start of the mp4, here is my command :

    ffmpeg -itsoffset 00:00:00.500 -i video.mp4 -i audio.mp3 -map 0:v -c:v copy -filter_complex '[0:a][1:a]amix[aout]' -map '[aout]' -shortest out.mp4

    This doesnt work as expected, the output doesnt have the expected delay of 500ms at the start of the mp4 streams. It appears the output is just the same as the first command I ran.

    The version of ffmpeg I am using is :

    ffmpeg version n4.0-39-gda39990

    Here are the files I’m using :

    video.mp4

    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/video.mp4':
         Metadata:
           major_brand     : mp42
           minor_version   : 0
           compatible_brands: isommp42
           creation_time   : 2019-07-26T03:28:49.000000Z
           com.android.version: 8.0.0
         Duration: 00:00:07.64, start: 0.000000, bitrate: 20534 kb/s
           Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuvj420p(pc, smpte170m), 1920x1080, 20966 kb/s, SAR 1:1 DAR 16:9, 29.70 fps, 29.75 tbr, 90k tbn, 180k tbc (default)
           Metadata:
             rotate          : 270
             creation_time   : 2019-07-26T03:28:49.000000Z
             handler_name    : VideoHandle
           Side data:
             displaymatrix: rotation of 90.00 degrees
           Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 96 kb/s (default)
           Metadata:
             creation_time   : 2019-07-26T03:28:49.000000Z
             handler_name    : SoundHandle

    audio.mp3

    Input #0, mp3, from '/storage/emulated/0/audio.mp3':
         Metadata:
           album           : Sunflower (Spider-Man: Into the Spider-Verse) - Single
           composer        : Louis Bell, Carter Lang, Austin Richard Post, Billy Walsh & Khalif Malik Ibin Shaman Brown
           genre           : Hip-Hop/Rap
           copyright       : This Compilation ℗ 2018 Republic Records, a division of UMG Recordings, Inc.
           title           : Sunflower (Spider-Man: Into the Spider-Verse)
           artist          : Post Malone & Swae Lee
           album_artist    : Post Malone & Swae Lee
           track           : 01/01
           TYER            : 2018-10-18T07:00:00Z
         Duration: 00:02:38.07, start: 0.000000, bitrate: 325 kb/s
           Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
           Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 1400x1400 [SAR 300:300 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
           Metadata:
             comment         : Cover (front)
  • ffmpeg - How to convert massive amounts of files in parallel ?

    15 juillet 2019, par Forivin

    I need to convert about 1.5TiB or audio files which are in either flac or wav format. They need to be converted into mp3 files, keeping important meta data and the cover art etc. and the bitrate needs to be 320k.

    This alone is easy :

    ffmpeg -i "$flacFile" -ab 320k -map_metadata 0 -id3v2_version 3 -vsync 2 "$mp3File" < /dev/null

    But the problem is making it faster. The command from above only uses 12.5% of the CPU. I’d much rather use like 80%. So I played around with the threads flag, but it doesn’t make it faster or slower :

    ffmpeg -i "$flacFile" -ab 320k -map_metadata 0 -id3v2_version 3 -vsync 2 -threads 4 "$mp3File" < /dev/null

    But it only utilizes my CPU by 13%. I think it only uses one thread. My CPU has 8 physical cores btw (+ hyperthreading).

    So my idea now is to somehow have multiple instances of ffmpeg running at the same time, but I have no clue how to do that properly.

    This is my current script to take all flac/wav files from one directory (recursively) and convert them to mp3 files in a new directory with the exact same structure :

    #!/bin/bash

    SOURCE_DIR="/home/fedora/audiodata_flac"
    TARGET_DIR="/home/fedora/audiodata_mp3"

    echo "FLAC/WAV files will be read from '$SOURCE_DIR' and MP3 files will be written to '$TARGET_DIR'!"
    read -p "Are you sure? (y/N)" -n 1 -r
    echo    # (optional) move to a new line
    if [[ $REPLY =~ ^[Yy]$ ]] ; then # Continue if user enters "y"

       # Find all flac/wav files in the given SOURCE_DIR and iterate over them:
       find "${SOURCE_DIR}" -type f \( -iname "*.flac" -or -iname "*.wav" \) -print0 | while IFS= read -r -d '' flacFile; do
           if [[ "$(basename "${flacFile}")" != ._* ]] ; then # Skip files starting with "._"
               tmpVar="${flacFile%.*}.mp3"
               mp3File="${tmpVar/$SOURCE_DIR/$TARGET_DIR}"
               mp3FilePath=$(dirname "${mp3File}")
               mkdir -p "${mp3FilePath}"
               if [ ! -f "$mp3File" ]; then # If the mp3 file doesn't exist already
                   echo "Input: $flacFile"
                   echo "Output: $mp3File"
                   ffmpeg -i "$flacFile" -ab 320k -map_metadata 0 -id3v2_version 3 -vsync 2 "$mp3File" < /dev/null
               fi
           fi
       done
    fi

    I mean I guess I could append an & to the ffmpeg command, but that would cause throusands of ffmpeg instances to run at the same time, which is too much.

  • ffplay playback h264 encoded camera preview blur

    11 juillet 2019, par doufu

    I use the ffplay tool provided by ffmpeg to play an h264 encoded camera. When the camera is still, the interface will be very clear, but when I cover the camera for a while to remove the occlusion or move the camera quickly, the interface will be blurred, resulting in a mosaic-like picture, but the interface will become clear after a few seconds.
    Why is this happening ?
    Am I missing some options to run ffplay ?
    PS:Using AMCap will not have the above problem.
    The command opening my camera I used :

    ffplay.exe -f dshow -i video="camera name"

    Preview screencast video is uploaded to youtube

    The following is the printout information of ffplay:
    enter image description here