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• Anomalie #4328 : Depots inaccessibles

  7 mai 2019, par Armen Spidermian

  Bonjour,

  Je me suis aperçu de ce dysfonctionnement sous spip 3.2.1
  Le site a été mis en prod et j’ai cru un moment que le probleme venait
  de mon hébergeur
  J’ai donc rapatrié une copie du site sur une machine de dev
  ou nous envisagions de migrer mais le probleme existe toujours.
  Ah oui ma version de php est 5.6.38 chez notre hébergeur et 5.6.40 ,sur
  le serveur de dev.
  Je ne comprends d’autant mons cette eeereur que j’ai un autre spip (3.2.4)
  qui tourne sur cette machine et qui n’a pas ce probleme.
  Je comprends que vous n’ayez pu reproduire cette erreur et donc de la
  traiter
  Mais comme je l’ai dit dans mon ticket je n’ai pas la moindre idée d’ou
  cela peut venir et donc ou chercher,
  peut-être pouvez vous me soumettre quelques pistes ?

  D’avance Merci.

  redmine@spip.org a écrit :

  La demande #4328 <https://core.spip.net/issues/4328#change-14926> a
  été mise à jour par b b.

  • Statut changé de /Nouveau/ à /Fermé/
  • Resolution mis à /invalid/

  Cette question a plus sa place sur la liste spip-zone, sans plus
  d’infos ça ne peut pas être considéré comme une bug, d’autant plus
  qu’on ne reproduit pas le problème de notre côté. Je ferme le ticket,
  n’hésite pas à en ouvrir un autre ou a commenter ici une fois que tu
  as des infos exploitables à propos d’un bug.

  ---

  Anomalie #4328 : Depots inaccessibles
  <https://core.spip.net/issues/4328#change-14926>

  • Auteur : Armen Spidermian
  • Statut : Fermé
  • Priorité : Normal
  • Assigné à :
  • Catégorie :
  • Version cible :
  • Resolution : invalid
  • Navigateur :

  Spip refuse de se connecter au dépot.

  2019-05-06 13:50:01 78.230.118.204 (pid 7481) :Pri:ERREUR : Erreur
  connexion
  2019-05-06 13:50:02 78.230.118.204 (pid 7481) :Pri:ERREUR : Erreur
  connexion
  2019-05-06 13:50:02 78.230.118.204 (pid 7481) :Pri : !INFO : copie_locale
   : Echec recuperation https://plugins.spip.net/depots/principal.xml sur
  ../IMG/distant/xml/principalxml2fdc.xml status : 0

  Je n’arrive pas a comprendre d’ou ca vient...

  Fichiers ErreurSpip.png
  <https://core.spip.net/attachments/download/1083/ErreurSpip.png> (84,9 ko)

  ---

  Vous recevez ce mail car vous êtes impliqués sur ce projet.
  Pour changer les préférences d’envoi de mail, allez sur
  http://core.spip.net/my/account

• Why is audio stream extraction slow ? (compared to AviSynth)

  6 mai 2019, par LLL

  I want to extract the audio stream of an avi file as a wav file, it works but it is really slow ( 4-5fps) although I just want to copy the stream.

  Here is the type of stream I want to extract (ffprobe info) :
  Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

  Going through AviSynth does it about 100 times faster, but I would prefer a pure FFmpeg solution. Why such a speed difference ? It looks like FFmpeg is reading and processing through the whole file whereas AviSynth can just extract the data without reading it.

  Example :
  ffmpeg -i file.avi -vn -ac 2 -c:a copy audio.wav
  or
  ffmpeg -i file.avi -map 0:a -ac 2 -c:a copy audio.wav
  both work fine but take time.

  Using an AviSynth script as input :
  ffmpeg -i script.avs -map 0:a -ac 2 -c:a copy audio.wav
  with script.avs containing just :
  AviSource("file.avi")
  does the same but almost instantaneously !

  Any idea why AviSynth is so much faster and if there is a way to get the same speed in FFmpeg ?

  Edit : adding logs
  Using FFmpeg directly :

  E:\>ffmpeg -i "file.avi" -map 0:a -c:a copy -y -benchmark "output.wav"
  
  ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181201
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --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 --enable-libopenmpt
  
    libavutil      56. 25.100 / 56. 25.100
  
    libavcodec     58. 43.100 / 58. 43.100
  
    libavformat    58. 25.100 / 58. 25.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 47.100 /  7. 47.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  [avi @ 0000018d3c38a680] non-interleaved AVI
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, avi, from 'file.avi':
  
    Duration: 00:18:37.49, start: 0.000000, bitrate: 534682 kb/s
  
      Stream #0:0: Video: rawvideo, bgr24, 1280x720, 533183 kb/s, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Output #0, wav, to 'output.wav':
  
    Metadata:
  
      ISFT            : Lavf58.25.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=  192445kB time=00:18:37.12 bitrate=1411.2kbits/s speed=4.77x
  
  video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
  
  bench: utime=1.188s stime=50.766s rtime=234.254s
  
  bench: maxrss=17468kB

  Using AviSynth :

  E:\>ffmpeg -i "soundout.avs" -map 0:a -c:a copy -y -benchmark "output.wav"
  
  ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181201
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --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 --enable-libopenmpt
  
    libavutil      56. 25.100 / 56. 25.100
  
    libavcodec     58. 43.100 / 58. 43.100
  
    libavformat    58. 25.100 / 58. 25.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 47.100 /  7. 47.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, avisynth, from 'soundout.avs':
  
    Duration: 00:18:37.49, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (BGR[24] / 0x18524742), bgr24, 1280x720, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Output #0, wav, to 'output.wav':
  
    Metadata:
  
      ISFT            : Lavf58.25.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=  192445kB time=00:18:37.11 bitrate=1411.2kbits/s speed= 155x
  
  video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
  
  bench: utime=0.234s stime=1.047s rtime=7.236s
  
  bench: maxrss=23792kB

• How to correctly calculate which segments are ready to be downloaded using MPEG-DASH

  24 avril 2019, par igal k

  What i’m trying to do ?

  Write a simple MPEG-DASH client using the SegmentTemplate pattern to calculate which segments are ready to be downloaded for a live source.

  A picture taken using chrome’s debugging tools at a moment X showing an mpd request(8af651fd747.....mpd) and the actual segments fetched respectfully to that request.

  

  Given the following MPD

  <mpd availabilitystarttime="2019-04-24T06:43:32Z" maxsegmentduration="PT4.096S" minbuffertime="PT4.096S" minimumupdateperiod="PT15.835S" profiles="urn:mpeg:dash:profile:isoff-live:2011" publishtime="2019-04-24T11:14:01Z" suggestedpresentationdelay="PT11.878S" timeshiftbufferdepth="PT65.536S" type="dynamic" xmlns="urn:mpeg:dash:schema:mpd:2011">
  
    <location>https://content-aaps1.uplynk.com/channel/8af651fd7473474f86a05ffb0a1c8972.mpd?rmt=wv&amp;amp;cid=8af651fd7473474f86a05ffb0a1c8972&amp;amp;oid=600e5c27541344a1bf3818617ad712ce&amp;amp;prettydash=1&amp;amp;exp=1556091088&amp;amp;rn=4138683939&amp;amp;tc=1&amp;amp;ct=c&amp;amp;sig=5fb7f0c18f3f1d2ad4fdee53c02c1e1ed904bc5e8474f4ebf886d209ff7f21c9&amp;amp;pbs=05b6594bcf4b4728ac1094976a80194d</location>
  
    <period start="PT2826.240S">
  
      <adaptationset maxframerate="30" maxheight="720" maxwidth="1280" mimetype="video/mp4" segmentalignment="true" startwithsap="1">
  
        <representation bandwidth="2604473" codecs="avc1.64001e" framerate="30" height="360" scantype="progressive" width="640">
  
          <baseurl>https://x-default-stgec.uplynk.com/aapm/slices/8c1/600e5c27541344a1bf3818617ad712ce/8c1027496a964b049f1bd5895f8f0412/</baseurl>
  
          <segmenttemplate duration="368640" initialization="https://x-default-stgec.uplynk.com/aapm/slices/8c1/600e5c27541344a1bf3818617ad712ce/8c1027496a964b049f1bd5895f8f0412/$RepresentationID$_init.mp4?pbs=05b6594bcf4b4728ac1094976a80194d&amp;amp;_jt=l&amp;amp;chid=8af651fd7473474f86a05ffb0a1c8972" media="$RepresentationID$$Number%08d$.m4f?pbs=05b6594bcf4b4728ac1094976a80194d&amp;amp;_jt=l&amp;amp;chid=8af651fd7473474f86a05ffb0a1c8972" presentationtimeoffset="254361599" startnumber="690" timescale="90000"></segmenttemplate>
  
        </representation>
  
      </adaptationset>
  
    </period>
  
    <utctiming schemeiduri="urn:mpeg:dash:utc:http-iso:2014" value="https://content-aaps1.uplynk.com/misc/utcservertime"></utctiming>
  
  </mpd>

  I see that the next segment request should be #3955

  What i have tried so far

      period.end = 1556104456;
  
      period.start = 2826;
  
      availability_start_time = 1556088212;
  
      max_segment_duration = 4;
  
      time_shift_buffer_depth = 65

  So, first of all, i read DASH-IF-IOP 4.3 section 4.3.4.2 page #82 and implemented the following code :

     int k1 = 1;
  
  int period_duration = period.end - (period.start + data_.availability_start_time);
  
  int k2 = ceil((float)period_duration / (float)data_.max_segment_duration);
  
  double duration = ((float)representation.duration / (float)representation.timeScale);
  
  size_t live_edge = std::min(
  
      (int)floor((float)((data_.publish_time - data_.availability_start_time - period.start) / duration)), k2);
  
  
  
  size_t oldest = std::max(k1, (int)floor((float)((data_.publish_time - data_.availability_start_time - period.start -
  
                                                      data_.time_shift_buffer_depth) /
  
                                                  duration)));

  after calculating everything : k1=1, k2=3355, live_edge=3272 and oldest=3256

  Also tried using ffmpeg’s dashdec.c

  for min_segment :

  if (c->is_live && pls->fragment_duration)
  
      {
  
          num = pls->first_seq_no + (((get_current_time_in_sec() - c->availability_start_time) - c->time_shift_buffer_depth) * pls->fragment_timescale) / pls->fragment_duration;
  
      }

  for max_segment :

  num = pls->first_seq_no + (((get_current_time_in_sec() - c->availability_start_time)) * pls->fragment_timescale) / pls->fragment_duration;

  after a small modification :

  size_t pmax = (((data_.publish_time - data_.availability_start_time))) / duration;
  
  size_t pmin = ((data_.publish_time - data_.availability_start_time) - data_.time_shift_buffer_depth) / duration;

  pmin=3946 pmax=3961

  in the ffmpeg example, i had to manually remove the first_seq_no variable because it looked like i doubled added the SegmentTemplate@StartNumber.

  even after succeeding in this task, how do i exactly build the request list of Segment(NOW) ----> Segment(LIVE_EDGE)