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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (5571)
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"Error opening input" error when running ffmpeg via shell script
15 octobre 2024, par Roni YanivI'm on a macbook running sonoma 14.7.


I have the following ffmpeg command which runs just fine by itself in the same folder of the files I need to modify :


ffmpeg -i "water-stream-90fps.mp4" -vf "drawtext=text='Frame\: %{frame_num}': start_number=1: x=(w-text_w): y=(h-text_h):fontsize=(h*10/100):fontcolor=yellow,drawtext=text='%{pts\:hms}':x=0:y=0:fontsize=(h*10/100):fontcolor=yellow" "water-stream-90fps-with-markers.mp4"


However, when I try to use it in a shell script by running
./add_markers.sh
in the same folder, I get an error.

This is the
add_markers.sh
script :

#!/bin/bash

# Replace '/path/to/your/folder' with the actual path to your folder
folder=$(pwd)

# Replace 'ffmpeg -i input.mp4 -vf scale=640:480 output.mp4' with your desired FFmpeg command
ffmpeg_command="ffmpeg -i input.mp4 -vf \"drawtext=text='Frame\: %{frame_num}': start_number=1: x=(w-text_w): y=(h-text_h):fontsize=(h*10/100):fontcolor=yellow,drawtext=text='%{pts\:hms}':x=0:y=0:fontsize=(h*10/100):fontcolor=yellow\" output.mp4"

echo "$ffmpeg_command"

# Loop through all files in the folder and execute the FFmpeg command

for file in "$folder"/*; do
 filename=$(basename "${file%.*}") #get only the first part of the file name without path or extension

 # Replace 'input.mp4' with the current filename and 'output.mp4' with the new filename. 
 # Added double quotes to manage filenames with spaces.
 new_command="${ffmpeg_command//input.mp4/\"$filename.mp4\"}"
 new_command="${new_command//output.mp4/\"$filename-with-markers.mp4\"}"

 echo "Running $new_command"

 $new_command

done



This is the error I get (for one of the files, same error for all of them) :


Running ffmpeg -i "water-stream-90fps.mp4" -vf "drawtext=text='Frame\: %{frame_num}': start_number=1: x=(w-text_w): y=(h-text_h):fontsize=(h*10/100):fontcolor=yellow,drawtext=text='%{pts\:hms}':x=0:y=0:fontsize=(h*10/100):fontcolor=yellow" "water-stream-90fps-with-markers.mp4"
ffmpeg version 7.0 Copyright (c) 2000-2024 the FFmpeg developers
 built with Apple clang version 15.0.0 (clang-1500.3.9.4)
 configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/7.0 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopenvino --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon
 libavutil 59. 8.100 / 59. 8.100
 libavcodec 61. 3.100 / 61. 3.100
 libavformat 61. 1.100 / 61. 1.100
 libavdevice 61. 1.100 / 61. 1.100
 libavfilter 10. 1.100 / 10. 1.100
 libswscale 8. 1.100 / 8. 1.100
 libswresample 5. 1.100 / 5. 1.100
 libpostproc 58. 1.100 / 58. 1.100
[in#0 @ 0x6000015bc300] Error opening input: No such file or directory
Error opening input file "water-stream-90fps.mp4".
Error opening input files: No such file or directory



I tried all kinds of things but my brain is starting to swell. Any guidance/help would be appreciated.


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Alexa developer console gives me "The audio is not of a supported MPEG version" Error
20 mars, par Boban BoBo BanjevicI am trying to make an Alexa skill, and I am using audio mp3 files stored is S3, lambda function has access to my DynamoDb and ultimately S3. But I have an issue with my audio files. I keep on getting "The audio is not of a supported MPEG version" when I test alexa skill. I tried multiple
ffmpeg -i
file conversions.

When I type
ffprobe -v quiet -print_format json -show_format -show_streams "Piano.mf.A1.audio10.mp3"
this is what i have for my file down below :


Input #0, mp3, from 'Piano.mf.A1.audio10.mp3':
 Metadata:
 encoder : Lavf61.9.107
 Duration: 00:00:07.00, start: 0.025057, bitrate: 129 kb/s
 Stream #0:0: Audio: mp3 (mp3float), 44100 Hz, stereo, fltp, 128 kb/s
 Metadata:
 encoder : Lavc61.33
PS C:\Users\boban\OneDrive\Desktop\newBob2> ffprobe -v quiet -print_format json -show_format -show_streams "Piano.mf.A1.audio10.mp3"
{
 "streams": [
 {
 "index": 0,
 "codec_name": "mp3",
 "codec_long_name": "MP3 (MPEG audio layer 3)",
 "codec_type": "audio",
 "codec_tag_string": "[0][0][0][0]",
 "codec_tag": "0x0000",
 "sample_fmt": "fltp",
 "sample_rate": "44100",
 "channels": 2,
 "channel_layout": "stereo",
 "bits_per_sample": 0,
 "initial_padding": 0,
 "r_frame_rate": "0/0",
 "avg_frame_rate": "0/0",
 "time_base": "1/14112000",
 "start_pts": 353600,
 "start_time": "0.025057",
 "duration_ts": 98795520,
 "duration": "7.000816",
 "bit_rate": "128000",
 "disposition": {
 "default": 0,
 "dub": 0,
 "original": 0,
 "comment": 0,
 "lyrics": 0,
 "karaoke": 0,
 "forced": 0,
 "hearing_impaired": 0,
 "visual_impaired": 0,
 "clean_effects": 0,
 "attached_pic": 0,
 "timed_thumbnails": 0,
 "non_diegetic": 0,
 "captions": 0,
 "descriptions": 0,
 "metadata": 0,
 "dependent": 0,
 "still_image": 0,
 "multilayer": 0
 },
 "tags": {
 "encoder": "Lavc61.33"
 }
 }
 ],
 "format": {
 "filename": "Piano.mf.A1.audio10.mp3",
 "nb_streams": 1,
 "nb_programs": 0,
 "nb_stream_groups": 0,
 "format_name": "mp3",
 "format_long_name": "MP2/3 (MPEG audio layer 2/3)",
 "start_time": "0.025057",
 "duration": "7.000816",
 "size": "112892",
 "bit_rate": "129004",
 "probe_score": 51,
 "tags": {
 "encoder": "Lavf61.9.107"
 }
 }
}



It should be everything that is required, am I doing something wrong, this is the only thing stopping me from finishing the skill. Asking ChatGPT said it could be because of the mp3float, and after many conversions it still doesn't work.


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Windows phone 8 neon inline assembly ffmpeg [duplicate]
5 février 2013, par user2036635Possible Duplicate :
windows phone8 wp8 arm neon assemblyI am about to transplant a project just like ffmpeg onto wp8(ARM). Unfortunately, most part of the project was written by arm neon inline assembly code (NEON inline assembly) with AT&T format..
Any suggestion about tools or methods to quickly doing that ? Or what should I do for that ?
I asked this question before but closed for the reason which is not constructive but I do not think so because what I mentioned is INLINE assembly with AT&T.
Any suggestion would be appreciated.