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GetID3 - Bloc informations de fichiers
9 avril 2013, par
Mis à jour : Mai 2013
Langue : français
Type : Image
Autres articles (105)
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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 -
Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Contribute to a better visual interface
13 avril 2011MediaSPIP 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 (7727)
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Receive RTSP stream within docker container
12 août 2021, par WayneI am trying to decode a camera rtsp stream using ffmpeg_libs within a ubuntu docker container. The ffmpeg debug output seems to show that it successfully negotiates the rtsp-digest authentication (ie. RTSP/1.0 200 OK), and receives an SPS (nalu 7) and PPS (nalu 8), but nothing after that. It times out, retries, etc. That doesn’t really make sense to me.



The same code compiled and run locally (not in docker) works fully.



Also, if I decode a file, the code works fine both locally and in docker container. So, the basic ffmpeg_lib decode is working. The difficulty is with the stream interface running in docker.



Is there additional authentication through the docker interface, or maybe port access, or something ? I’m not much of a networking guy, so I’m really lost at this point.



The ffmpeg logs is below, and my docker run command is :



docker run -it --name VideoRx videorx:latest (also tried with -p 554)




Any help will be very much appreciated.

Thanks,
Wayne


avformat_version(): 3756900 Build: 3756900 Ident: Lavf57.83.100
avformat_open_input(): rtsp://admin:public_pwd@192.168.1.237
Probing rtsp score:100 size:0
[tcp @ 0x56263b430a20] No default whitelist set
[rtsp @ 0xaddr1] Sending:
OPTIONS rtsp://192.168.1.237:554 RTSP/1.0

... [snipped]
Initial authentication handshake (OPTIONS, DESCRIBE, SETUP).
All success, server replies: 'RTSP/1.0 200 OK'
....

[rtsp @ 0xaddr1] Sending:
PLAY rtsp://192.168.1.237:554/ RTSP/1.0
Range: npt=0.000-
CSeq: 5
User-Agent: Lavf57.83.100
Session: 420467284
Authorization: Digest username="admin", realm="IP Camera(C1003)", nonce="129b254c8da4e0ffb530f64f79938bcd", uri="rtsp://192.168.1.237:554/", response="82c6c0f1fadea3739846866e8e50e855"

--
[rtsp @ 0xaddr1] line='RTSP/1.0 200 OK' 
[rtsp @ 0xaddr1] line='CSeq: 5'
[rtsp @ 0xaddr1] line='Session: 420467284'
[rtsp @ 0xaddr1] line='RTP-Info: url=rtsp://192.168.1.237:554/trackID=1;seq=43938;rtptime=4022155312'
[rtsp @ 0xaddr1] line='Date: Thu, Aug 02 2018 15:53:00 GMT'
[rtsp @ 0xaddr1] line=''
avformat_open_input(): Success erc: 0
avformat_find_stream_info()
[h264 @ 0xaddr2] nal_unit_type: 7, nal_ref_idc: 3
[h264 @ 0xaddr2] nal_unit_type: 8, nal_ref_idc: 3
[rtsp @ 0xaddr1] UDP timeout, retrying with TCP 
[rtsp @ 0xaddr1] ...
... Stalls waiting for additional packets



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Duration of short ogg files (Telegram Voice messages) not correct when loaded into Python
4 août 2018, par KrommeI’m trying to read voice messages, sent by Telegram, using Python but for short voice clips (< 10 seconds), it doesn’t work. It shortens the duration for some reason. It looks like it has something to do with
OGG codec
, but I’m not really sure.See here’s my code, the voice clip is about six seconds, however
pydub
reads my 6 second voiceclip as 0.06 seconds.import telegram
from pydub import AudioSegment
AudioSegment.ffmpeg = "./dependencies/ffmpeg-20180802-c9118d4-win64-static/bin/ffmpeg"
AudioSegment.converter = "./dependencies/ffmpeg-20180802-c9118d4-win64-static/bin/ffmpeg"
bot = telegram.Bot(token=token)
f = bot.get_file(file_id)
f.download('output/voiceclips/{}.ogg'.format(file_id))
myaudio = AudioSegment.from_ogg("output/voiceclips/{}.ogg".format(file_id))
print('ID: {}, which is {} seconds'.format(file_id, myaudio.duration_seconds))
>>> ID: ______, which is 0.06 secondsWhen I open the file in
VLC-player
, it also states that is has 0 seconds. When I try to convert it to WAV-files using FFmpeg it reads the ogg file as 6 seconds, but writes it as 0.05-second WAV file.ffmpeg -i infile.ogg outfile.wav
ffmpeg version N-91549-gc9118d4d64 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 7.3.1 (GCC) 20180722
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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
libavutil 56. 18.102 / 56. 18.102
libavcodec 58. 22.100 / 58. 22.100
libavformat 58. 17.101 / 58. 17.101
libavdevice 58. 4.101 / 58. 4.101
libavfilter 7. 26.100 / 7. 26.100
libswscale 5. 2.100 / 5. 2.100
libswresample 3. 2.100 / 3. 2.100
libpostproc 55. 2.100 / 55. 2.100
[ogg @ 0000020dd375ad40] 727 bytes of comment header remain
Input #0, ogg, from 'infile.ogg':
Duration: 00:00:06.03, start: 0.000000, bitrate: 20 kb/s
Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'outfile.wav':
Metadata:
ISFT : Lavf58.17.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
Metadata:
encoder : Lavc58.22.100 pcm_s16le
size= 6kB time=00:00:00.05 bitrate= 873.0kbits/s speed=4.12x
video:0kB audio:6kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.354167%For larger files it does the work !
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ffmpeg with "-pattern_type glob" and variable in bash script
20 juin 2019, par KlausPeterI’m trying to let ffmpeg make a video of all pictures in a directory using the -pattern_type glob switch and "/foo/bar/*.jpg". This works well, if I execute the command manually für just one directory. For example :
ffmpeg -framerate 35 -pattern_type glob -i '/root/webcam_test/2018-07-21/*.jpg' -vf scale=1280:-1 -c -c:v libx264 -pix_fmt yuv420p /root/clips/out01_cut.mp4
However, if I do it in a bash script and set the path via a variable, according to ffmpegs output, the variable gets substituted correctly, but ffmpeg states that
’/root/webcam_test/2018-07-21/*.jpg’ : No such file or directory
The part of the script looks like this :
for D in `find /root/webcam_test/ -type d`
do
[...]
cmd="ffmpeg -framerate 35 -pattern_type glob -i '$D/*.jpg' -vf scale=1280:-1 -c -c:v libx264 -pix_fm t yuv420p /root/clips/$d_cut.mp4"
echo $cmd
[...]
doneDoes anyone know how to make ffmpeg do its wildcard interpretation even if the path is constructed by a script and not just try to plainly use the given path ?
Best regards and thanks in advance