
Recherche avancée
Médias (1)
-
Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (108)
-
Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
-
Ajouter des informations spécifiques aux utilisateurs et autres modifications de comportement liées aux auteurs
12 avril 2011, parLa manière la plus simple d’ajouter des informations aux auteurs est d’installer le plugin Inscription3. Il permet également de modifier certains comportements liés aux utilisateurs (référez-vous à sa documentation pour plus d’informations).
Il est également possible d’ajouter des champs aux auteurs en installant les plugins champs extras 2 et Interface pour champs extras. -
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
Sur d’autres sites (15686)
-
Trim off N bytes from audio file using SoX / FFmpeg etc, on Windows ?
17 novembre 2020, par Rinaldo JonathanMy team accidentally on purpose clicked NO when Audacity asked to save the recording. So I left with bunch of *.au files, after using recovery program.


Some of them did have header and still open-able with audacity itself (example : this one), and some other are just complete nonsense, sometimes having the header filled with text from random javascript or HTML code (like this one). Probably hard disk half overwritten with browser cache ? I don't know. And at this point, I almost don't care.


The audacity is on default settings, with sample rate 44100Hz. I can open a-113.au using audacity, from standard open files. I also was able to achieve open files using "Open RAW files" from Audacity, using this settings :




so I assume header takes 12384 bytes.


Now, how do I trim 12384 bytes from the file when opened as RAW, with either SoX or ffmpeg ? because if I open it as RAW with 0 offset (default settings), it will add the header as a noise.


Current ffmpeg command I used :
ffmpeg.exe -f f32le -ar 44.1k -ac 1 -i source destination

Current sox command I used :sox -t raw --endian little --rate 44100 -b 1 -b 32 --encoding floating-point %%A "converted/%%~nxA.wav"

Both still have header as a noise in the converted files.

-
ffmpeg : xstack doesn't work when inputs are scaled to certain dimensions
4 juin 2020, par dfriend21I'm using ffmpeg to create a mosaic of videos using the
xstack
filter. The input videos may come in varying dimensions, so I'm using thescale
filter to scale them beforehand, and I'm using theforce_original_aspect_ratio
option and then thepad
filter to keep the original aspect ratios of each video and add black bars to the sides to make each video have the correct dimensions.


I have a command that's working - however, it's inconsistent. For some dimensions it works, while for others it doesn't.



I'm using the
fluent-ffmpeg
Node.js module to callffmpeg
from Node.js. To do this, I'm passing an array of strings to thecomplexFilter()
function.


The following strings for the complex filter works :



"[0:v]scale=400:250:force_original_aspect_ratio=decrease,pad=400:250:(ow-iw)/2:(oh-ih)/2 [s0]"
"[1:v]scale=400:250:force_original_aspect_ratio=decrease,pad=400:250:(ow-iw)/2:(oh-ih)/2 [s1]"
"[s0][s1]xstack=inputs=2:fill='black':layout=0_0|w0_0[v]"
"[0:a][1:a]amix=inputs=2[a]"




However, if I change the output dimensions of each video to be 400:225 instead of 400:250 it fails.



"[0:v]scale=400:225:force_original_aspect_ratio=decrease,pad=400:225:(ow-iw)/2:(oh-ih)/2 [s0]"
"[1:v]scale=400:225:force_original_aspect_ratio=decrease,pad=400:225:(ow-iw)/2:(oh-ih)/2 [s1]"
"[s0][s1]xstack=inputs=2:fill='black':layout=0_0|w0_0[v]"
"[0:a][1:a]amix=inputs=2[a]"




The following error is given :



An error occurred: ffmpeg exited with code 1: Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #1:1
Conversion failed!




If it's relevant, the first video has dimensions of 1280x720 while the second video has dimensions of 320x240.



Anyone know why one set of dimensions works while the other doesn't ?



EDIT : Here is the full ffmpeg log for when it fails :



ffmpeg version git-2020-05-13-b12b053 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200513
 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-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 45.100 / 56. 45.100
 libavcodec 58. 84.100 / 58. 84.100
 libavformat 58. 43.100 / 58. 43.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 80.100 / 7. 80.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/user_name/Desktop/vids/vid1.mp4':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: mp41isom
 creation_time : 2020-05-21T15:52:20.000000Z
 Duration: 00:00:10.76, start: 0.000000, bitrate: 8385 kb/s
 Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuvj420p(pc), 1280x720 [SAR 1:1 DAR 16:9], 8237 kb/s, 29.99 fps, 30 tbr, 30k tbn, 60 tbc (default)
 Metadata:
 creation_time : 2020-05-21T15:52:20.000000Z
 handler_name : VideoHandler
 encoder : AVC Coding
 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 165 kb/s (default)
 Metadata:
 creation_time : 2020-05-21T15:52:20.000000Z
 handler_name : SoundHandler
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/user_name/Desktop/vids/vid2.mp4':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: mp41isom
 creation_time : 2020-05-21T15:54:37.000000Z
 Duration: 00:00:11.01, start: 0.000000, bitrate: 836 kb/s
 Stream #1:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuvj420p(pc), 320x240 [SAR 1:1 DAR 4:3], 669 kb/s, 29.88 fps, 30 tbr, 30k tbn, 60 tbc (default)
 Metadata:
 creation_time : 2020-05-21T15:54:37.000000Z
 handler_name : VideoHandler
 encoder : AVC Coding
 Stream #1:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 163 kb/s (default)
 Metadata:
 creation_time : 2020-05-21T15:54:37.000000Z
 handler_name : SoundHandler
Stream mapping:
 Stream #0:0 (h264) -> scale
 Stream #0:1 (aac) -> amix:input0
 Stream #1:0 (h264) -> scale
 Stream #1:1 (aac) -> amix:input1
 xstack -> Stream #0:0 (libx264)
 amix -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
[swscaler @ 000001343fefc200] deprecated pixel format used, make sure you did set range correctly
[Parsed_pad_1 @ 000001343f8dc3c0] Padded dimensions cannot be smaller than input dimensions.
[Parsed_pad_1 @ 000001343f8dc3c0] Failed to configure input pad on Parsed_pad_1
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #1:1
Conversion failed!

Done in 0.66s.



-
FFmpeg merge two files from different sources with different bitrates
2 mars 2023, par FamousWollufI have created an intro video with Adobe Premiere Pro and I want to add the intro video to my downloaded Mixer stream videos before uploading them to YouTube, without reencoding the complete video through Adobe Premiere because that takes a long time because the videos are more then an hour.



I tried it with FFmpeg
ffmpeg -safe 0 -f concat -i list.txt -c copy output.mp4
.
The intro is normal but the video part from Mixer is running way to fast.

I think it has something todo with the bitrate but I can't figure it out.


Output of intro file :



Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: mp42mp41
 creation_time : 2020-05-12T11:12:11.000000Z
 Duration: 00:00:06.06, start: 0.000000, bitrate: 2408 kb/s
 Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 2086 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
 Metadata:
 creation_time : 2020-05-12T11:12:11.000000Z
 handler_name : ?Mainconcept Video Media Handler
 encoder : AVC Coding
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 317 kb/s (default)
 Metadata:
 creation_time : 2020-05-12T11:12:11.000000Z
 handler_name : #Mainconcept MP4 Sound Media Handler




Output of Mixer file :



Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.12.100
 Duration: 01:02:00.58, start: 0.000000, bitrate: 1937 kb/s
 Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720, 1800 kb/s, 29.41 fps, 29.97 tbr, 16k tbn, 32k tbc (default)
 Metadata:
 handler_name : VideoHandler
 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
 Metadata:
 handler_name : SoundHandler




Output of the merged file :



Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.43.100
 Duration: 00:57:03.84, start: 0.000000, bitrate: 2109 kb/s
 Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 3377 kb/s, 55.14 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
 Metadata:
 handler_name : ?Mainconcept Video Media Handler
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 140 kb/s (default)
 Metadata:
 handler_name : #Mainconcept MP4 Sound Media Handler




If I need to reencode something my preference would be the intro because it's only 6 seconds.