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Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...)
Sur d’autres sites (7445)
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Trimming videos using FFMPEG + Python : Invalid Argument
26 juin 2019, par user2588534I am looking to use a short Python script to automate some video cutting for me based on a .txt file I created with timecodes. I use the following code for that :
#!/usr/bin/env python3
import subprocess
first = True
with open('VideoSlicepoints_Python.txt') as f:
for line in f.readlines():
if first:
first = False
else:
filename, VideoDing_Timecode, Video_R4Start, Video_R4End, Video_R24Start, Video_R24End, Video_R0plus15 = line.strip().split(';')
cmd = ['ffmpeg', '-i', filename, '-ss', Video_R4Start, '-to', Video_R4End, '-c', 'copy', '"S:/_RobotThesis/VideoRounds/' + filename[:-4] + '_r4' + filename[-4:] + '"']
print(" ".join(cmd))
subprocess.run(cmd, stderr=subprocess.STDOUT)This returns me with the following error when it is trying to loop over to videos :
[NULL @ 0000027cddf8a100] Unable to find a suitable output format for '"S:/_RobotThesis/VideoRounds/log1_front_r4.MTS"'
"S:/_RobotThesis/VideoRounds/log1_front_r4.MTS": Invalid argumentNow, I thought it was a formatting issue, but when I input the command directly (as is visible in the CMD prompt), it works just fine :
ffmpeg -i log1_front.MTS -ss 09:37.1 -to 11:37.1 -c copy "S:/_RobotThesis/VideoRounds/log1_front_r4.MTS"
For reference, this is an excerpt of the .txt file I am using (it is based on a csv file, that’s why it has headers and I excluded the header in the script) :
Filename;VideoDing_Timecode;Video_R4Start;Video_R4End;Video_R24Start;Video_R24End;Video_R0plus15
log1_front.MTS;04:44.0;09:37.1;11:37.1;28:00.3;30:00.3;19:44.0
log2_front.MTS;03:50.0;08:11.2;10:11.2;19:44.9;21:44.9;18:50.0
log3_front.MTS;04:10.1;08:32.4;10:32.4;16:49.2;18:49.2;19:10.1
log5_front.MTS;01:14.7;04:50.2;06:50.2;14:24.5;16:24.5;16:14.7Edit :
Here is the full output in the CMD, if needed :
ffmpeg -i log1_front.MTS -ss 09:37.1 -to 11:37.1 -c copy "S:/_RobotThesis/VideoRounds/log1_front_r4.MTS"
ffmpeg version N-94054-gdd357d76e5 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 8.3.1 (GCC) 20190414
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
libavutil 56. 28.100 / 56. 28.100
libavcodec 58. 53.100 / 58. 53.100
libavformat 58. 27.103 / 58. 27.103
libavdevice 58. 7.100 / 58. 7.100
libavfilter 7. 55.100 / 7. 55.100
libswscale 5. 4.101 / 5. 4.101
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
Input #0, mpegts, from 'log1_front.MTS':
Duration: 00:37:35.05, start: 1.440000, bitrate: 12560 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(top first), 1440x1080 [SAR 4:3 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc
Stream #0:1[0x101]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 256 kb/s
[NULL @ 000002d1f347a100] Unable to find a suitable output format for '"S:/_RobotThesis/VideoRounds/log1_front_r4.MTS"'
"S:/_RobotThesis/VideoRounds/log1_front_r4.MTS": Invalid argument -
How to run the ffmpeg command using node.js ?
12 juillet 2019, par Sachin ShahIn node, When I got the request from
/playMovie
from App, I need to broadcast the video.When I execute this command in terminal it works fine.
ffmpeg -re -i movie.mkv -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv rtmp://192.168.1.13/live/myStream
Now I’m going to setup this dynamic.
app.use('/playMovie', function (req, res) {
console.log("playMovie...");
let filePaht = 'movie.mkv';
let fileName = 'marvel-avengers';
let ffmpeg = spawn(`ffmpeg -re -i ${filePaht} -c:v libx264 -preset
superfast -tune zerolatency -c:a aac -ar 44100 -f flv
rtmp://192.168.1.13/live/${fileName}`);
ffmpeg.on('exit', (statusCode) => {
console.log("statusCode ::::::::::::::::::::::::::::::::: ",statusCode);
if (statusCode === 0) {
console.log('conversion successful')
}
})
ffmpeg
.stderr
.on('data', (err) => {
console.log('err:', new String(err))
})
});While run the app I got this error.
playMovie...
12/07/2019 15:27:17 31722 [ERROR] uncaughtException { Error: spawn ffmpeg -re -i movie.mkv -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv rtmp://192.168.1.13/live/marvel-avengers ENOENT
at _errnoException (util.js:1022:11)
at Process.ChildProcess._handle.onexit (internal/child_process.js:190:19)
at onErrorNT (internal/child_process.js:372:16)
at _combinedTickCallback (internal/process/next_tick.js:138:11)
at process._tickCallback (internal/process/next_tick.js:180:9)
code: 'ENOENT',
errno: 'ENOENT',
syscall: 'spawn ffmpeg -re -i movie.mkv -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv
rtmp://192.168.1.13/live/marvel-avengers',
path: 'ffmpeg -re -i movie.mkv -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv rtmp://192.168.1.13/live/marvel-avengers',
spawnargs: [] } -
How to convert aac to ogg opus keeping bit rate and sample rate unchanged
25 juin 2019, par DooviI’m trying to convert a .aac file to .opus but after inspecting with ffprobe I get different bit and sample rates.
While input file’s audio stream bit rate is 245995, the output file’s audio stream has no bit rate specified - "format" shows bit rate of 118788.
While input file’s audio stream sample rate is 44100, the output’s is 48000.
ffprobe -v error -show_format -show_streams input.aac
[STREAM]
index=0
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_time_base=1/44100
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
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/28224000
start_pts=N/A
start_time=N/A
duration_ts=106533390807
duration=3774.567418
bit_rate=245995
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
[/STREAM]
[FORMAT]
filename=input.aac
nb_streams=1
nb_programs=0
format_name=aac
format_long_name=raw ADTS AAC (Advanced Audio Coding)
start_time=N/A
duration=3774.567418
size=116065589
bit_rate=245995
probe_score=51
[/FORMAT]ffmpeg -nostdin -i input.aac -c:a libopus output.opus
ffmpeg version N-93449-g013f714 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
configuration: --prefix=/home/vagrant/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/vagrant/ffmpeg_build/include --extra-ldflags=-L/home/vagrant/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/vagrant/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
libavutil 56. 26.100 / 56. 26.100
libavcodec 58. 47.105 / 58. 47.105
libavformat 58. 26.101 / 58. 26.101
libavdevice 58. 7.100 / 58. 7.100
libavfilter 7. 48.100 / 7. 48.100
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
[aac @ 0x55d4b7e21d80] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'input.aac':
Duration: 01:02:54.57, bitrate: 245 kb/s
Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 245 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (aac (native) -> opus (libopus))
[libopus @ 0x55d4b7e3f8c0] No bit rate set. Defaulting to 96000 bps.
Output #0, opus, to 'output.opus':
Metadata:
encoder : Lavf58.26.101
Stream #0:0: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s
Metadata:
encoder : Lavc58.47.105 libopus
size= 52103kB time=00:59:53.21 bitrate= 118.8kbits/s speed=66.2x
video:0kB audio:51733kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.715930%ffprobe -v error -show_format -show_streams output.opus
[STREAM]
index=0
codec_name=opus
codec_long_name=Opus (Opus Interactive Audio Codec)
profile=unknown
codec_type=audio
codec_time_base=1/48000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
sample_rate=48000
channels=2
channel_layout=stereo
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/48000
start_pts=0
start_time=0.000000
duration_ts=172473677
duration=3593.201604
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:ENCODER=Lavc58.47.105 libopus
[/STREAM]
[FORMAT]
filename=output.opus
nb_streams=1
nb_programs=0
format_name=ogg
format_long_name=Ogg
start_time=0.000000
duration=3593.201604
size=53353867
bit_rate=118788
probe_score=100
[/FORMAT]How can I preserve the quality of the input file ? Am I missing something in the ffmpeg cmd ?