
Recherche avancée
Médias (1)
-
Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (100)
-
MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (11840)
-
Can I know which byte range to read from a remote mp4 file for FFMpeg to decode a keyframe ?
12 octobre 2023, par db9117I need to decode a of keyframe of a video file (mp4, h264 encoded). I know the timestamp of the keyframe I want to extract/decode. I want to minimize amount of data being read in memory. For this, I need to know beforehand exactly the minimal byte range I would require that encompasses this keyframe. How do I know what is the minimal byte range in the whole mp4 byte stream I need to read in order to be able to decode the keyframe ?


I currently find the appropriate keyframe in the
index_entries
contained in the header. I get its byte position (pos
attribute) and timestamp (timestamp
attribute). I calculate the range as follows :

startBytes
: minimum of :

- 

- the
pos
of the keyframe - the
pos
of the nearest index entry in the audio stream happening at or before the keyframe's timestamp.






This way when it's decoding the frame, if it also needs the audio content for demuxing, it would have it.


endBytes
: maximum of :

- 

- the
pos
of the next frame in the video stream's index, after the keyframe - the
pos
of the next frame in the audio stream's index after the timestamp of the wished keyframe.






This way I know that I have everything up until the next frame in the index, which theoretically should be enough to decode the keyframe only.


I then read the appropriate byte range.


When I try to decode the frame, I run in a loop until I succeed :


- 

avcodec_read_frame
avcodec_send_packet
avcodec_receive_frame








I ignore
AVERROR(EAGAIN)
errors.

avcodec_receive_frame
fails multiple times with errorAVERROR(EAGAIN)
which I ignore, until it fails saying that the memory it wants to read isn't available (wants to read afterendBytes
). I explicitly tell it to fail if it wants to read more than it has already read.

Note : for other keyframes at other positions in other videos, it sometimes succeeds (probably because the range is big enough by chance), but it fails more often than not.


My question is : Why is the end of the range not enough to be able to decode only the one keyframe ? Is there any way to more precisely calculate the exact range in bytes I would need in order to decode a particular keyframe ?


- the
-
mp3 to wav with ffmpeg reduces quality and duration
25 novembre 2020, par NeretI took this mp3 file and converted it to wav with Audition and with this ffmpeg command :


ffmpeg -i "Casey Don’t You Fret - Dan Lebowitz.mp3" -c:a pcm_f32le "Casey Don’t You Fret - Dan Lebowitz_FFMPEG.wav"



After that I checked the statistics in Audition. The wav file which was generated with Audition has exactly the same statistics as the original mp3 file.




The duration of ffmpeg file has changed. Audio statistics became worse.
Why is this happening ? Can I fix it ?


I used
ffmpeg version 2020-11-22-git-0066bf4d1a-full_build-www.gyan.dev
on Windows.

UPDATE 1 :
I cut a few seconds of mp3 at the beginning and at the end :




FFMPEG added silence at the beginning and increased duration.


UPDATE 2 :
Look how ffmpeg changed the waveform of this
2.mp3
file in the middle :

ffmpeg -y -i 2.mp3 -c:a pcm_f32le 2_FFMPEG.wav





-
TCP Connection refused error when using FFMPEG for audio stream to HTTP on macOS
26 novembre 2020, par freddyI'm trying to stream my microphone input via HTTP using ffmpeg, so I can stream it in HTML. I run the following ffmpeg command :


ffmpeg -f avfoundation -i ":1" -c:a libmp3lame -f mp3 -r 30 http://localhost:809


It, however, crashes with the following error message :


ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
 built with Apple clang version 12.0.0 (clang-1200.0.32.27)
 configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --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-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
 libavutil 56. 51.100 / 56. 51.100
 libavcodec 58. 91.100 / 58. 91.100
 libavformat 58. 45.100 / 58. 45.100
 libavdevice 58. 10.100 / 58. 10.100
 libavfilter 7. 85.100 / 7. 85.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 7.100 / 5. 7.100
 libswresample 3. 7.100 / 3. 7.100
 libpostproc 55. 7.100 / 55. 7.100
Input #0, avfoundation, from ':1':
 Duration: N/A, start: 3445.340045, bitrate: 22579 kb/s
 Stream #0:0: Audio: pcm_f32le, 44100 Hz, hexadecagonal, flt, 22579 kb/s
[tcp @ 0x7fa46ec96600] Connection to tcp://localhost:8090 failed: Connection refused
http://localhost:8090: Connection refused



I've had success with streaming on that port using VLC, but it for some reason won't work using ffmpeg. Any ideas on how to fix this ?