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Autres articles (72)

  • Soumettre améliorations et plugins supplémentaires

    10 avril 2011

    Si vous avez développé une nouvelle extension permettant d’ajouter une ou plusieurs fonctionnalités utiles à MediaSPIP, faites le nous savoir et son intégration dans la distribution officielle sera envisagée.
    Vous pouvez utiliser la liste de discussion de développement afin de le faire savoir ou demander de l’aide quant à la réalisation de ce plugin. MediaSPIP étant basé sur SPIP, il est également possible d’utiliser le liste de discussion SPIP-zone de SPIP pour (...)

  • Configurer la prise en compte des langues

    15 novembre 2010, par

    Accéder à la configuration et ajouter des langues prises en compte
    Afin de configurer la prise en compte de nouvelles langues, il est nécessaire de se rendre dans la partie "Administrer" du site.
    De là, dans le menu de navigation, vous pouvez accéder à une partie "Gestion des langues" permettant d’activer la prise en compte de nouvelles langues.
    Chaque nouvelle langue ajoutée reste désactivable tant qu’aucun objet n’est créé dans cette langue. Dans ce cas, elle devient grisée dans la configuration et (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (8164)

  • ffmpeg YUV to RGB distored color and position

    15 novembre 2012, par user1542140

    Sorry that I still cannot post images for my question since low reputation.

    I use the ffmpeg function to convert the decoded frame, from YUV to RGB24, but the color and resulted image is distorted seriously. Following is my code snip, the frame width and height is (176, 144)

    len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
    
    if (got_picture) {
    
            //...
    

    
    AVFrame *pFrameRGB = avcodec_alloc_frame();
    
    // Determine required buffer size and allocate buffer
    
    int numBytes=avpicture_get_size(PIX_FMT_RGB24, c->width, c->height);
    
    uint8_t *buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
    
    avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24, c->width, c->height);
    

    
    struct SwsContext *img_convert_ctx = sws_getContext(c->width, c->height, PIX_FMT_YUV420P, c->width, c->height, PIX_FMT_BGR24, SWS_BICUBIC, NULL, NULL, NULL);
    
    sws_scale(img_convert_ctx, picture->data, picture->linesize, 0, picture->height, pFrameRGB->data, pFrameRGB->linesize);
    
    sws_freeContext(img_convert_ctx);
    
    // Save the frame to disk
    
    if(++frame<=5)
    
        SaveFrame(pFrameRGB, c->width, c->height, frame);

  • keepalive type and frequency in ffmpeg [on hold]

    19 novembre 2013, par Jack Simth

    My company has a bunch of IP cameras that we distribute - specifically Grandstream - and the manufacturer has changed their firmware. The normal keepalive that ffmpeg uses for the rtsp streams ( either ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL) ; or ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL) ; both in in libavformat/rtspdec.c) is no longer working, for two reasons :

    1) The new Grandstream firmware is now checking for a receiver report to determine whether or not the program reading the stream is live, not just anything.

    2) The new Grandstream firmware requires that the receiver report to keep the connection alive happen at least once every 25 seconds, and on the audio stream it is currently only happening about every 30 seconds or so (video is getting it every 7 seconds or so).

    So after about a minute with ffmpeg connected, the camera stops sending the audio stream, the audio stream on ffmpeg reads end-of-file, and then ffmpeg shuts everything down.

    As I can't change the firmware, I'm trying to dig through the ffmpeg code to make it send the appropriate receiver report for the keep alive... but I am getting nowhere. I've added a little snippet of code into the receiver reports so I know when they're running when I call ffmpeg on debug, but... well, it's not going well.

    Test command :
    ffmpeg -loglevel debug -i rtsp ://admin:admin@192.168.4.3:554/0 -acodec libmp3lame -ar 22050 -vcodec copy -y -f flv /dev/null &> test.txt

    Test output :

    `[root@localhost ffmpeg]# cat test.txt
    
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Aug 21 2013 14:24:28 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
    
  configuration: --datadir=/usr/share/ffmpeg --bindir=/usr/local/bin --libdir=/usr/local/lib --incdir=/usr/local/include --shlibdir=/usr/lib --mandir=/usr/share/man --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-libx264 --enable-gpl --enable-version3 --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-x11grab --enable-librtmp --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-static --enable-libgsm --enable-libxvid --enable-libvpx --enable-libvorbis --enable-libvo-aacenc --enable-libmp3lame
    
  libavutil      52. 38.100 / 52. 38.100
    
  libavcodec     55. 18.102 / 55. 18.102
    
  libavformat    55. 12.100 / 55. 12.100
    
  libavdevice    55.  3.100 / 55.  3.100
    
  libavfilter     3. 79.101 /  3. 79.101
    
  libswscale      2.  3.100 /  2.  3.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  3.100 / 52.  3.100
    
Splitting the commandline.
    
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
    
Reading option '-i' ... matched as input file with argument 'rtsp://admin:admin@192.168.4.3:554/0'.
    
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'libmp3lame'.
    
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '22050'.
    
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'.
    
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
    
Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
    
Reading option '/dev/null' ... matched as output file.
    
Finished splitting the commandline.
    
Parsing a group of options: global .
    
Applying option loglevel (set logging level) with argument debug.
    
Applying option y (overwrite output files) with argument 1.
    
Successfully parsed a group of options.
    
Parsing a group of options: input file rtsp://admin:admin@192.168.4.3:554/0.
    
Successfully parsed a group of options.
    
Opening an input file: rtsp://admin:admin@192.168.4.3:554/0.
    
[rtsp @ 0x9d9ccc0] SDP:
    
v=0
    
o=StreamingServer 3331435948 1116907222000 IN IP4 192.168.4.3
    
s=h264.mp4
    
c=IN IP4 0.0.0.0
    
t=0 0
    
a=control:*
    
m=video 0 RTP/AVP 96
    
a=control:trackID=0
    
a=rtpmap:96 H264/90000
    
a=fmtp:96 packetization-mode=1; sprop-parameter-sets=Z0LgHtoCgPRA,aM4wpIA=
    
m=audio 0 RTP/AVP 0
    
a=control:trackID=1
    
a=rtpmap:0 PCMU/8000
    
a=ptime:20
    
m=application 0 RTP/AVP 107
    
a=control:trackID=2
    
a=rtpmap:107 vnd.onvif.metadata/90000
    

    

    
[rtsp @ 0x9d9ccc0] video codec set to: h264
    
[NULL @ 0x9d9f400] RTP Packetization Mode: 1
    
[NULL @ 0x9d9f400] Extradata set to 0x9d9f900 (size: 22)!
    
[rtsp @ 0x9d9ccc0] audio codec set to: pcm_mulaw
    
[rtsp @ 0x9d9ccc0] audio samplerate set to: 8000
    
[rtsp @ 0x9d9ccc0] audio channels set to: 1
    
[rtsp @ 0x9d9ccc0] hello state=0
    
[h264 @ 0x9d9f400] Current profile doesn't provide more RBSP data in PPS, skipping
    
    Last message repeated 1 times
    
[rtsp @ 0x9d9ccc0] All info found
    
Guessed Channel Layout for  Input Stream #0.1 : mono
    
Input #0, rtsp, from 'rtsp://admin:admin@192.168.4.3:554/0':
    
  Metadata:
    
    title           : h264.mp4
    
  Duration: N/A, start: 0.000000, bitrate: 64 kb/s
    
    Stream #0:0, 28, 1/90000: Video: h264 (Constrained Baseline), yuv420p, 640x480, 1/180000, 10 tbr, 90k tbn, 180k tbc
    
    Stream #0:1, 156, 1/8000: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
    
Successfully opened the file.
    
Parsing a group of options: output file /dev/null.
    
Applying option acodec (force audio codec ('copy' to copy stream)) with argument libmp3lame.
    
Applying option ar (set audio sampling rate (in Hz)) with argument 22050.
    
Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy.
    
Applying option f (force format) with argument flv.
    
Successfully parsed a group of options.
    
Opening an output file: /dev/null.
    
Successfully opened the file.
    
detected 2 logical cores
    
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'time_base' to value '1/8000'
    
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'sample_rate' to value '8000'
    
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'sample_fmt' to value 's16'
    
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'channel_layout' to value '0x4'
    
[graph 0 input from stream 0:1 @ 0x9f15380] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4
    
[audio format for output stream 0:1 @ 0x9efa7c0] Setting 'sample_fmts' to value 's32p|fltp|s16p'
    
[audio format for output stream 0:1 @ 0x9efa7c0] Setting 'sample_rates' to value '22050'
    
[audio format for output stream 0:1 @ 0x9efa7c0] Setting 'channel_layouts' to value '0x4|0x3'
    
[audio format for output stream 0:1 @ 0x9efa7c0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
    
[AVFilterGraph @ 0x9f15980] query_formats: 4 queried, 9 merged, 3 already done, 0 delayed
    
[auto-inserted resampler 0 @ 0x9dfada0] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s16p r:22050Hz
    
Output #0, flv, to '/dev/null':
    
  Metadata:
    
    title           : h264.mp4
    
    encoder         : Lavf55.12.100
    
    Stream #0:0, 0, 1/1000: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 640x480, 1/90000, q=2-31, 1k tbn, 90k tbc
    
    Stream #0:1, 0, 1/1000: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 22050 Hz, mono, s16p
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
  Stream #0:1 -> #0:1 (pcm_mulaw -> libmp3lame)
    
Press [q] to stop, [?] for help
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 135.4kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 134.4kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 135.0kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 135.5kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 136.9kbits/s
    
Queue input is backward in time=     233kB time=00:00:13.69 bitrate= 139.4kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 136.3kbits/s
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 13926; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 13952; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 13979; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14005; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14031; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14057; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14083; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14109; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14135; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14161; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14188; changing to 14239. This may result in incorrect timestamps in the output file.
    
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14214; changing to 14239. This may result in incorrect timestamps in the output file.
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.5kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 142.0kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 142.5kbits/s
    
Receiver Report delay: 469789, gettime: -1527669086, last_recep: 322446, timebase: -1534837492
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.5kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.7kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.1kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 140.6kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 140.7kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.9kbits/s
    
Receiver Report delay: 132993, gettime: -1516538925, last_recep: 322446, timebase: -1518568234
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.6kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.6kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.7kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.4kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 140.0kbits/s
    
Receiver Report delay: 897727, gettime: -1504870331, last_recep: 322446, timebase: -1518568552
    
[NULL @ 0x9d9f400] Current profile doesn't provide more RBSP data in PPS, skipping
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.4kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.1kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.0kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.0kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 138.6kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 138.5kbits/s
    
Current profile doesn't provide more RBSP data in PPS, skippingrate= 138.2kbits/s
    
EOF on sink link output stream 0:1:default.time=00:00:58.40 bitrate= 139.6kbits/s
    
No more output streams to write to, finishing.
    
[libmp3lame @ 0x9dfa580] Trying to remove 344 more samples than there are in the queue
    
frame=  589 fps= 11 q=-1.0 Lsize=    1003kB time=00:00:58.85 bitrate= 139.5kbits/s
    
video:724kB audio:231kB subtitle:0 global headers:0kB muxing overhead 4.955356%
    
2959 frames successfully decoded, 0 decoding errors
    
[AVIOContext @ 0x9e021c0] Statistics: 3 seeks, 2860 writeouts
    
[root@localhost ffmpeg]#

  • How to crop last N seconds from a video

    10 décembre 2013, par digitalfootmark

    Is there any way to crop the last N seconds from a video ?

    I know there is an option for start time and duration, but neither of these are usable in this use case. Video can have any possible length, so the duration cannot be fixed value.

    Maybe I can determine the length first with ffmpeg, and calculate the duration in my own code ?

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