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Médias (91)
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999,999
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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The Slip - Artworks
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
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Demon seed (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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The four of us are dying (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Corona radiata (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Lights in the sky (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
Autres articles (41)
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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 (6591)
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avcodec/movtextdec : Fix immediately adjacent styles
17 octobre 2020, par Andreas Rheinhardtavcodec/movtextdec : Fix immediately adjacent styles
The checks for whether a style should be opened/closed at the current
character position are as follows : A variable entry contained the index
of the currently active or potentially next active style. If the current
character position coincided with the start of style[entry], the style
was activated ; this was followed by a check whether the current
character position coincided with the end of style[entry] ; if so, the
style was deactivated and entry incremented. Afterwards the char was
processed.The order of the checks leads to problems in case the endChar of style A
coincides with the startChar of the next style (say B) : Style B was never
opened. When we are at said common position, the currently active style
is A and so the start pos check does not succeed ; but the end pos check
does and it closes the currently active style A and increments entry.
At the next iteration of the loop, the current character position is
bigger than the start position of style B (which is style[entry]) and
therefore the style is not activated.The solution is of course to first check for whether a style needs to be
closed (and increment entry if it does) before checking whether the next
style needs to be opened.Reviewed-by : Philip Langdale <philipl@overt.org>
Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com> -
tools/target_dec_fuzzer : Adjust VQA threshold
18 juillet 2020, par Michael Niedermayertools/target_dec_fuzzer : Adjust VQA threshold
Fixes : Timeout (169sec -> 9sec)
Fixes : 23745/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VQA_fuzzer-5638172179693568Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by : Michael Niedermayer <michael@niedermayer.cc> -
FFMPEG :Adjusting PTS and DTS of encoded packets with real timestamps information
19 août 2020, par jackey balwaniI am trying to record a video of activities in desktop using muxing in FFMPEG Library.
I referred following link for my implementation.
https://ffmpeg.org/doxygen/trunk/muxing_8c_source.html


I observe that the encoded video plays too fast. Below is the code of encoding the frame and then writing it to output file.


static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
 AVStream *st, AVFrame *frame)
{
 int ret;
 
 // send the frame to the encoder
 ret = avcodec_send_frame(c, frame);
 if (ret < 0) {
 fprintf(stderr, "Error sending a frame to the encoder: %s\n",av_err2str(ret));
 exit(1);
 }
 while (ret >= 0) {
 AVPacket pkt = { 0 };

 ret = avcodec_receive_packet(c, &pkt);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
 break;
 else if (ret < 0) {
 fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
 exit(1);
 }

 /* rescale output packet timestamp values from codec to stream timebase(from 25 to 90000 this case) */
 av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
 pkt.stream_index = st->index;

 /* Write the compressed frame to the media file. */
 log_packet(fmt_ctx, &pkt);
 ret = av_interleaved_write_frame(fmt_ctx, &pkt);
 av_packet_unref(&pkt);
 if (ret < 0) {
 fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
 exit(1);
 }
 }
 
 return ret == AVERROR_EOF ? 1 : 0;
 }



-> I am rescaling output packet timestamp values from codec to stream timebase (25 -> 90,000) before writing.
-> Since encoded video does plays very fast, so I have added temporary fix to multiply stream timebase by some factor just before av_packet_rescale_ts (in my case it is 5 -> 5*st->time_base.den which becomes 4,50,000) so that output video duration increases. This may not be the accurate solution as this would not give the real time shift between the frames.


/** rescale output packet timestamp values from codec to stream timebase **/
st->time_base.den = 5*st->time_base.den;
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
/** Write the compressed frame to the media file. **/
return av_interleaved_write_frame(fmt_ctx, pkt);



-> I want that output video duration should be similar to the input. I have heard of things like giving real timestamps to the PTS and DTS of frame and packets and encoding at variable or unknown frame rate.
Can we use these , If yes then How to implement ?
any suggestions or pseudo code of the approach would really help.


Thanks