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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
Autres articles (67)
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Gestion générale des documents
13 mai 2011, parMédiaSPIP ne modifie jamais le document original mis en ligne.
Pour chaque document mis en ligne il effectue deux opérations successives : la création d’une version supplémentaire qui peut être facilement consultée en ligne tout en laissant l’original téléchargeable dans le cas où le document original ne peut être lu dans un navigateur Internet ; la récupération des métadonnées du document original pour illustrer textuellement le fichier ;
Les tableaux ci-dessous expliquent ce que peut faire MédiaSPIP (...) -
Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
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 (...)
Sur d’autres sites (4875)
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avformat/asfdec_o : Don't segfault with lots of attached pics
12 novembre 2020, par Andreas Rheinhardtavformat/asfdec_o : Don't segfault with lots of attached pics
The ASF file format has a limit of 127 streams and the "asf_o" demuxer
(the ASF demuxer from Libav) has an array of pointers for a structure
called ASFStream that is allocated on demand for every stream. Attached
pictures are not streams in the sense of the ASF specification, yet the
demuxer created an ASFStream for them ; and in one codepath it also
forgot to check whether the array of ASFStreams is already full. The
result is a write beyond the end of the array and a segfault lateron.Fixing this is easy : Don't create ASFStreams for attached picture
streams.(Other results of the current state of affairs are unnecessary allocations
(of ASFStreams structures), the misparsing of valid files (there might not
be enough ASFStreams left for the valid streams if attached pictures take
up too many) ; furthermore, the ASFStreams created for attached pictures all
have the stream number 0, an invalid stream number (the valid range is
1-127). This means that invalid data (packets for a stream with stream
number 0) won't get rejected lateron.)Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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movenc : Allow writing a DASH sidx atom at the start of files
21 octobre 2014, par Martin Storsjömovenc : Allow writing a DASH sidx atom at the start of files
This is mapped to the faststart flag (which in this case
perhaps should be called "shift and write index at the
start of the file"), which for fragmented files will
write a sidx index at the start.When segmenting DASH into files, there’s usually one sidx
at the start of each segment (although it’s not clear to me
whether that actually is necessary). When storing all of it
in one file, the MPD doesn’t necessarily need to describe
the individual segments, but the offsets of the fragments can be
fetched from one large sidx atom at the start of the file. This
allows creating files for the DASH ISO BMFF on-demand profile.Signed-off-by : Martin Storsjö <martin@martin.st>
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How to adjust mpeg 2 ts start time with ffmpeg ?
29 juin 2015, par Maxim KornienkoI’m writing simple HLS (Http Live Streaming) java server to live cast (really live, not on demand) screenshow + voice. I constantly get chunks of image frames and audio samples as input to my service and produce mpeg 2 ts files + m3u8 playlist web page as output. The workflow is the following :
- Collect (buffer) source video frames and audio for certain period of time
- Convert series of video frames to h.264 encoded video file
- Convert audio samples to mp3 audio file
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Merge them to
.ts
file with ffmpeg commandffmpeg -i audio.mp3 -i video.mp4 -f mpegts -c:a copy -c:v copy -vprofile main -level:v 4.0 -vbsf h264_mp4toannexb -flags -global_header segment.ts
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Publish several
.ts
files on m3u8 playlist.
The problem is resulting playlist interrupts after first segment is played. VLC logs following error :
freetype error: Breaking unbreakable line
ts error: libdvbpsi (PSI decoder): TS discontinuity (received 0, expected 4) for PID 17
ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 0
ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 4096
core error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 1000 ms)
core error: ES_OUT_RESET_PCR called
core error: Could not convert timestamp 185529572000
ts error: libdvbpsi (PSI decoder): TS discontinuity (received 0, expected 4) for PID 17
ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 0
ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 4096
core error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 8653 ms ignored)
core error: Could not get display date for timestamp 0
core error: Could not convert timestamp 185538017000
core error: Could not convert timestamp 185538267000
core error: Could not convert timestamp 185539295977
...I guess the reason is that start time of segments do not belong to one stream, but it’s impossible to concat and resegment (with
ffmepg -f segment
) whole stream once new chunk is added. Tried adding#EXT-X-DISCONTINUITY
tag to playlist as suggested here but it didn’t help. When Iffprobe
them I get :Input #0, mpegts, from '26.ts':
Duration: 00:00:10.02, start: 1.876978, bitrate: 105 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 640x640, 4 fps, 4 tbr, 90k tbn, 8 tbc
Stream #0:1[0x101]: Audio: mp3 ([3][0][0][0] / 0x0003), 48000 Hz, mono, s16p, 64 kb/sWhere start value in line
Duration: 00:00:10.02, start: 1.876978, bitrate: 105 kb/s
is more or less equal for all segments.
When I check segments from available proven-to-work playlists (like http://vevoplaylist-live.hls.adaptive.level3.net/vevo/ch1/appleman.m3u8) they all have diffrenet start values for each segment, for example :Input #0, mpegts, from 'segm150518140104572-424570.ts':
Duration: 00:00:06.17, start: 65884.808689, bitrate: 479 kb/s
Program 257
Stream #0:0[0x20]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 320x180 [SAR 1:1 DAR 16:9], 30 fps, 29.97 tbr, 90k tbn, 60 tbc
Stream #0:1[0x21]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 115 kb/s
Stream #0:2[0x22]: Data: timed_id3 (ID3 / 0x20334449)and the next after it
Input #0, mpegts, from 'segm150518140104572-424571.ts':
Duration: 00:00:06.22, start: 65890.814689, bitrate: 468 kb/s
Program 257
Stream #0:0[0x20]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 320x180 [SAR 1:1 DAR 16:9], 30 fps, 29.97 tbr, 90k tbn, 60 tbc
Stream #0:1[0x21]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 124 kb/s
Stream #0:2[0x22]: Data: timed_id3 (ID3 / 0x20334449)differ in the way that start time of
segm150518140104572-424571.ts
is equal to start time + duration ofsegm150518140104572-424570.ts
.How could this start value be adjusted with
ffmpeg
? Or maybe my whole aproach is wrong ? Unfortunately I couldn’t find on the internet working example of live (not on demand) video service implemented with ffmepg.